| 00:00.32 | Trionnis | anyone around that can help with a sip authentication problem? sip.conf and the sip debugs are here: http://pastebin.ca/691032 |
| 00:01.37 | Trionnis | I have autocreatepeer=yes and allowguest=yes in the top of sip.conf, however it still won't let the other system send a call out through asterisk |
| 00:04.13 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:07.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:07.31 | drwelby | For Zoiper as an IZX client to asterisk, what should dtmfmode in iax.conf be set to? |
| 00:07.59 | drwelby | I have it =auto, but I can't seem to send DTMF to Asterisk |
| 00:08.34 | JT | IZX? |
| 00:08.56 | drwelby | iz iax for hizzoes |
| 00:08.58 | Trionnis | Inter Zoiper Exchange? |
| 00:09.01 | Trionnis | hehe :) |
| 00:09.06 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:11.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:14.14 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 00:15.05 | drwelby | Damn, iy must be monday if my chat client crashes right when I ask a question |
| 00:15.17 | drwelby | and I can't spell "it" |
| 00:15.32 | *** join/#asterisk AirCoder (n=Aircoder@ppp-71-133-4-40.dsl.irvnca.pacbell.net) |
| 00:16.10 | AirCoder | any one using inphonex with asterisk have a quick q... |
| 00:18.42 | threat | hello, I am still having DMTF problems, I cannot use the key pad on automated menus |
| 00:19.24 | threat | I use g729 codec, my phone provider used a payload type of 96 which I have added to a file in the asterisk source code and recompiled |
| 00:19.30 | threat | is there anything else I need to do to get it working? |
| 00:20.17 | threat | I am using dtmfmode=rfc2833 |
| 00:21.13 | *** join/#asterisk Poehali (n=actionma@74.93.5.186) |
| 00:21.26 | Poehali | hey TK I followed ur guide |
| 00:21.37 | Poehali | here's the problem though |
| 00:21.53 | Poehali | I don't know for sure that asterisk is seeing SPA3102 because there's no way I can test it |
| 00:22.23 | threat | I have added in [96] = {0, AST_RTP_DTMF} |
| 00:22.27 | *** join/#asterisk thermalwetland (n=Matt@pele.comtelhi.com) |
| 00:23.16 | thermalwetland | anyone try to apply this patch to 1.4.9 - http://bugs.digium.com/view.php?id=4903 |
| 00:23.23 | thermalwetland | it allows SIP over TCP |
| 00:23.27 | threat | Any ideas? |
| 00:25.40 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 00:25.52 | kuku5 | Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 75.57.175 << What does that mean ? |
| 00:25.57 | AirCoder | poehali you tring to see if an adapter is comunicating with asterisk? |
| 00:26.17 | _ShrikE | kuku5: didnt happen to get that from an audiocodes did you? |
| 00:26.38 | Trionnis | audiocodes? |
| 00:26.41 | Trionnis | *growl* |
| 00:27.09 | _ShrikE | its a brand of voip gateway. I have seen that before but only with them. |
| 00:27.15 | _ShrikE | me too. |
| 00:27.16 | kuku5 | _ShrikE: what do you mean ? |
| 00:27.34 | threat | What is the asterisk mailing list? I am not getting any answers here |
| 00:27.38 | Trionnis | like saying "oh, yeah we support TBCT transfers!" |
| 00:27.44 | Trionnis | then finding out that they don't |
| 00:27.58 | Trionnis | after 2 months of fighting with our pri providers about it not working... |
| 00:28.02 | JT | threat: google.com |
| 00:28.09 | threat | JT: I hate you |
| 00:28.14 | outtolunc | 2 months <G> |
| 00:28.25 | JT | threat: ? |
| 00:28.43 | threat | JT: You have always been far from helpful |
| 00:28.53 | JT | "asterisk mailing list" > second result |
| 00:28.57 | JT | highly helpful |
| 00:29.00 | Trionnis | welcome to IRC |
| 00:29.03 | JT | you must just be highly lazy |
| 00:29.08 | AirCoder | lol |
| 00:29.12 | outtolunc | http://lists.digium.com/mailman/listinfo |
| 00:29.12 | Trionnis | RTFM then come back and ask :) |
| 00:29.20 | Poehali | lol |
| 00:29.22 | Poehali | I knew it |
| 00:29.23 | Trionnis | (yes, I'm being sarcastic) |
| 00:29.33 | outtolunc | the second link on google:asterisk mailing lists |
| 00:29.34 | JT | lists.digium.com is not hard to guess either :) |
| 00:29.54 | JT | outtolunc: some people are just tools who want to double click on links in their irc client |
| 00:29.57 | JT | unfortunately |
| 00:30.02 | outtolunc | nods |
| 00:30.02 | AirCoder | any one farmiliar with the inphonex network i got a intresting question. |
| 00:30.04 | threat | What the hell, this is an asterisk channel and nobody knows the mailing list without going to google? |
| 00:30.14 | AirCoder | and i researched the question for a good 4 hours. |
| 00:30.14 | *** join/#asterisk Barmal (n=info@c-24-30-126-164.hsd1.ga.comcast.net) |
| 00:30.17 | puzzled | threat: I do: lists.digium.com |
| 00:30.23 | Trionnis | outtolunc: yes, 2 months, after no less than 10 requests to their support people wanting help with the debugs in AC |
| 00:30.28 | JT | threat: yes, it's because we don't remember moronic information that can be googled like http://lists.digium.com/mailman/listinfo |
| 00:30.33 | threat | puzzled: yay |
| 00:30.38 | outtolunc | some of us have been subscribed for going on 5ish years |
| 00:30.42 | Trionnis | AudioCodes is a very crappy company when it comes to supporting their overpriced hardware |
| 00:30.52 | outtolunc | forgive us for not remembering offhand <G> |
| 00:30.53 | JT | it's the sort of url you need to go to VERY RARELY |
| 00:30.56 | Poehali | anyone know how I can navigate to https://asterisk/static/config/setup/install.html from https://asterisk? |
| 00:31.00 | threat | JT: perhaps this channel needs a bot to hold this type of info |
| 00:31.09 | JT | ~google |
| 00:31.10 | jbot | well, google is a search engine found at http://www.google.com/ |
| 00:31.13 | threat | or maybe it should be put into the topic |
| 00:31.15 | Trionnis | pwnt |
| 00:31.16 | Trionnis | lol |
| 00:31.20 | puzzled | hahaha |
| 00:31.24 | Trionnis | threat, you just got served |
| 00:31.27 | Trionnis | big time |
| 00:31.33 | threat | JT: so add in mailing list now |
| 00:31.40 | JT | ~lists |
| 00:31.41 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-5e92c359db2dfa8c) |
| 00:31.42 | JT | ~list |
| 00:31.43 | jbot | one warez list being sent |
| 00:31.46 | JT | heh |
| 00:31.49 | puzzled | lol |
| 00:31.50 | threat | heh |
| 00:31.56 | Trionnis | ~mailing list |
| 00:31.59 | watchy | super warez |
| 00:32.01 | JT | jbot: lists is at http://lists.digium.com/mailman/listinfo |
| 00:32.05 | threat | JT: and you should use another bot for your warez related activities :) |
| 00:32.28 | Barmal | where can be the problem that asterisk answers incomming call but after about 10sec call hangs up? sip debug show that invite is still keeps going after answered call, and I can talk for those 10secs... what can be wrong? |
| 00:32.51 | threat | JT: thank you, you have now redeamed your self |
| 00:33.21 | JT | excellent |
| 00:33.24 | Trionnis | anyone ever messed around with VoiceGenie? |
| 00:33.33 | Trionnis | and managed to get it to play well with * ? |
| 00:33.51 | threat | Trionnis: nope :( |
| 00:34.12 | threat | Trionnis: have you ever got DTMF working at payload type 96? |
| 00:34.31 | Trionnis | I think VG is probably the only company I've ever worked with that's *worse* at support than AudioCodes |
| 00:34.39 | Trionnis | nope, can't say I have |
| 00:34.47 | Barmal | where should I look for the problem? call comes from voiceeclipse |
| 00:35.03 | _ShrikE | audiocodes is an absolute nightmare |
| 00:35.14 | _ShrikE | imo |
| 00:35.18 | Trionnis | yup |
| 00:35.48 | Trionnis | "here, try this new firmware" |
| 00:36.10 | Trionnis | "oops, we destroyed your $4000 Mediant 3000... we'll ship you an RMA from Israel in about a month" |
| 00:36.16 | Trionnis | ... |
| 00:36.47 | Trionnis | I'll be so happy to get rid of those damned things and go native sip |
| 00:36.57 | _ShrikE | exactly |
| 00:37.14 | Trionnis | just have to get Level 3 to stop trying to bend us over on the monthly commitment |
| 00:37.33 | Trionnis | $10k a month is a bit of a chunk to commit to for sip termination |
| 00:37.36 | Trionnis | heh |
| 00:37.44 | _ShrikE | lord I guess. |
| 00:38.08 | Trionnis | well, we're not far from it right now |
| 00:38.16 | Trionnis | doing about 750k min a month |
| 00:38.20 | Trionnis | but still... |
| 00:38.36 | _ShrikE | 10k is still a whole lot for 750k |
| 00:38.39 | Trionnis | yea |
| 00:38.54 | Trionnis | but they're the only "big" players right now that are willing to give us a direct handoff |
| 00:38.55 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 00:39.18 | Trionnis | Qwest, Gblx, XO, and such won't even talk to me until I hit at least 2 mil min a month |
| 00:39.36 | Trionnis | and none of the other smaller guys will do lata/ocn rates |
| 00:40.03 | Trionnis | I'm not paying 2c a minute when I can get sub 1c through Level3 |
| 00:40.12 | _ShrikE | right |
| 00:40.34 | Trionnis | hell, I've got 1.8 on PSTN pri right now, it's just that the loop is killing me |
| 00:40.44 | Trionnis | farking AT&T... $8k/mo for a ds3 loop, ugh |
| 00:41.44 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 00:42.36 | AirCoder | any one farmiliar with the inphonex network i got a intresting question. |
| 00:42.51 | Trionnis | can't say I am |
| 00:44.02 | dlynes | Anyone know what might be causing the error, 'chan_sip.c: Remote host can't match request BYE to call 'blahblahblah@ip.add.re.ss'. Giving up.? |
| 00:44.24 | dlynes | Another thing manifesting itself on this system is that trying to transfer a call results in a dropped call on all ends |
| 00:44.31 | dlynes | This is all on asterisk 1.4.11 |
| 00:47.42 | Barmal | what does mean udp cheksum offload? |
| 00:49.04 | wothinn | Barmal: Means your network card checks to make sure the UDP packet is well-formed and your operating system doesn't need to make your CPU do it. |
| 00:50.29 | Barmal | wothinn: what needs to be changed on asterisk config files? |
| 00:51.26 | *** join/#asterisk heartones (n=heartone@196.202.118.23) |
| 00:51.52 | heartones | hi any one awake |
| 00:52.05 | Barmal | tshark showes every packet comming from me with that error... Can it be the reason why the call is beeing dropped? |
| 00:52.07 | Trionnis | nope |
| 00:52.13 | wothinn | I'm far from an asterisk expet, Barmal, but I don't think Asterisk can turn it on and off... it's purely a network card driver thing. |
| 00:52.14 | Trionnis | I'm sleep-typing |
| 00:52.37 | wothinn | Well, in that case, I'd try replacing your NIC. |
| 00:53.36 | Barmal | no it works fine with other sip provider not with voiceeclipse... and voiceeclipse works fine connected to other asterisk server.... :( |
| 00:55.40 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:56.16 | riddlebox | can someone give me a link to the O'reilly book? |
| 00:56.40 | riddlebox | !book |
| 00:57.37 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 00:57.46 | riddlebox | ~book |
| 00:57.47 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 00:58.59 | Poehali | that book is so reader unfriendly though |
| 00:59.10 | Poehali | there needs to be a asterisk for dummies version |
| 00:59.15 | kiscokid | I managed to read it |
| 00:59.31 | Poehali | how? |
| 00:59.46 | kiscokid | If you follow through the examples you'll get a working * system |
| 01:00.28 | Poehali | I need SPA3102 specific examples but they don't seem to have it |
| 01:00.39 | kiscokid | If you can't read that book maybe you should consider hiring a consultant or buying the * appliance |
| 01:03.03 | Poehali | you are hired! |
| 01:04.36 | kiscokid | you could look at this http://www.voip-info.org/wiki/index.php?page=Linksys-Cisco+3102 |
| 01:06.16 | kiscokid | and here http://forum.voxilla.com/linksys-sipura-voip-support-forum/ |
| 01:07.18 | Poehali | all I need to know is if sipura is communicating with asterisk |
| 01:07.21 | Trionnis | can someone give some advice on allowing an unauthenticated outbound call through asterisk v 1.4.11 ? |
| 01:07.23 | Poehali | is there a command to check that? |
| 01:07.33 | Trionnis | I have allowguest set to yes, but it doesn't seem to be working |
| 01:07.40 | kiscokid | Poehali: try sip show peers |
| 01:07.56 | kuku5 | Do I need a stun or ser server to work out nat issues |
| 01:08.24 | Trionnis | stun server can't hurt anything really, might as well enable it |
| 01:09.34 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-0cbc44b0b19372e8) |
| 01:09.42 | Poehali | kiscokid: I get "1 sip peers [monitored: 0 online, 0 offline unmonitored: 0 1 offline |
| 01:09.59 | Poehali | this is because I added the lines from TK's guide |
| 01:10.10 | Poehali | but I don't think it's really seeing the device |
| 01:10.29 | kiscokid | maybe its not registering |
| 01:10.50 | Poehali | is there a registration thing I need to do on the sipura settings? |
| 01:11.32 | *** join/#asterisk pruonckk (n=mike@200.212.179.130) |
| 01:11.44 | kiscokid | yeah, you have to tell it the ip address of the * server as well as put in the name and password for the sip.conf entry |
| 01:11.58 | *** join/#asterisk vn (n=nostalge@modemcable083.222-80-70.mc.videotron.ca) |
| 01:12.22 | vn | hi, is it true that its better to have a static WAN IP when having VoIP? |
| 01:12.23 | kiscokid | just like any sip device |
| 01:12.27 | pruonckk | somebody here using hylafax iaxmodem and asterisk, im trying to do this work without successs |
| 01:12.55 | Poehali | kiscokid: which section would it be under? I see voice/info, voice/system, voice/user 1, voice/ PSTN user |
| 01:13.08 | Trionnis | user 1 |
| 01:13.16 | pruonckk | how can i redirect a incoming call to iaxmodem device ttyIAX ? |
| 01:13.20 | Trionnis | most likely |
| 01:14.20 | kiscokid | Poehali: probably voice/system but I never configured one of those |
| 01:14.39 | Poehali | Trionnis: under user one it has: call forward settings, selective call forward settings, dial settings, service settings |
| 01:14.49 | Poehali | none of them are for inputting IP address |
| 01:14.58 | *** join/#asterisk powerkill (n=powerkil@84.205.154.247) |
| 01:15.14 | kiscokid | Poehali: what is under service settings? |
| 01:15.18 | powerkill | hi |
| 01:15.23 | powerkill | does someones use a quadgsm from voismart ? |
| 01:15.33 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 01:16.06 | Poehali | kiscokid: cw setting yes/no, block ANC setting no, CID setting, dist ring setting, yes, message waiting ... |
| 01:16.26 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 01:16.37 | kiscokid | Poehali: do you have the manual for that device? |
| 01:17.34 | Poehali | kiscokid: yes but it doesn't describe asterisk |
| 01:17.44 | jdg | pruonckk: dial(IAX2/iaxmodem) |
| 01:17.57 | pruonckk | i have try this without success |
| 01:18.05 | pruonckk | i cant understund what is wrong |
| 01:18.09 | kiscokid | it should describe connecting to a sip server |
| 01:19.15 | pruonckk | i dont have any log to help me, iax dont show anything, hylafax dont show anything |
| 01:19.22 | pruonckk | i dont know |
| 01:19.48 | jdg | pruonckk: Does "iax2 show peers" show your iaxmodem ? |
| 01:19.57 | pruonckk | yes |
| 01:20.12 | Poehali | kiscokid: is asterisk considered a sip server? |
| 01:20.17 | pruonckk | jdg, i can send fax |
| 01:20.21 | pruonckk | but i can receive |
| 01:20.30 | pruonckk | (cant receive) |
| 01:20.33 | kiscokid | Poehali: yes |
| 01:20.36 | hmmhesays | can anyone recommend an inexpensive up that that can be monitored so it will shut down properly when battery is low? |
| 01:21.06 | Poehali | kiscokid: manual has "connecting to voice gateway" |
| 01:21.34 | jdg | pruonckk: what does * console show for incoming calls ? |
| 01:21.57 | kiscokid | Poehli: looking at the manual |
| 01:22.07 | pruonckk | jdg, wait a second, i will configure again |
| 01:23.11 | Poehali | voice gateway=sip server? |
| 01:23.24 | pruonckk | jdg, Executing Dial("IAX2/pabx-sp-2", "IAX2/iaxmodem") |
| 01:23.38 | pruonckk | but saty calling, no answer from iaxmodem |
| 01:24.00 | pruonckk | (*saty -> stay) |
| 01:24.11 | *** part/#asterisk vn (n=nostalge@modemcable083.222-80-70.mc.videotron.ca) |
| 01:25.23 | jdg | does "faxstat -s" show ttyIAX running and idle ? |
| 01:25.45 | pruonckk | all running and idle |
| 01:25.56 | pruonckk | ( i have configured 3 IAX devices ) |
| 01:26.26 | jdg | fine |
| 01:27.09 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 01:27.32 | TJNII | So if I have queue(queuename,n) how long is the timeout before going to the next step in the dialplan? |
| 01:29.10 | TJNII | Is it (timeout * retry) from queue.conf? |
| 01:29.29 | pruonckk | jdg, anyother idea ? |
| 01:29.36 | jdg | pruonckk: sorry, no more idea |
| 01:29.40 | pruonckk | hehe |
| 01:29.48 | pruonckk | jdg, thanks man |
| 01:31.32 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-108ef2dd7a6d499e) |
| 01:32.46 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 01:34.24 | *** join/#asterisk strav (n=sdfsdf@modemcable078.64-56-74.mc.videotron.ca) |
| 01:34.27 | strav | he |
| 01:35.51 | jdg | pruonckk: faxgetty running ? |
| 01:36.17 | pruonckk | yes |
| 01:36.27 | pruonckk | /usr/local/sbin/faxgetty ttyIAX2 |
| 01:36.31 | pruonckk | one for each device |
| 01:37.12 | pruonckk | jdg, i need go now, i will try more tomorow |
| 01:37.19 | pruonckk | thanks for your help |
| 01:37.26 | pruonckk | good night for all |
| 01:37.29 | jdg | ok ! |
| 01:39.17 | strav | I came few a little while ago about a problem I'm having with my current configuration. It seems asterisk dosen't receive the input when it's waiting for an extension (waitexten cmd). Here is the relevant part of my extension.conf: http://pastebin.ca/691135 . If anyone cares... |
| 01:39.52 | strav | (note, I tested this setup yesterday and it was working fine) |
| 01:41.02 | strav | (yes and in my extension copy/paste, I ommited the [start] block title.) |
| 01:42.19 | hmmhesays | you guys have any suggestions for telephone line surge protection? |
| 01:43.27 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:45.52 | *** join/#asterisk Egonis (n=root@70.54.211.179) |
| 01:46.14 | strav | ... Yes it is a noob question. waitexten is most straightforward to use. Still, as my server answer and does process the command, I really can't know why entering an extension number has no effect (while in debugging mode of course)... |
| 01:46.29 | Egonis | I have compiled zaptel and asterisk (1.4.11) and chan_zap.so isn't being automatically compiled. How do I force it to compile? Zaptel is loaded and found all FXS channels |
| 01:47.52 | strav | egonis: don't you have options to your .configure? |
| 01:48.43 | Egonis | strav: it has 'XXX' over it, so I cannot enable or disable it |
| 01:50.57 | strav | anyone? could changing my config to autofallthrough=yes help my actual problem? |
| 01:52.02 | jdg | strav: what does * exactly ? |
| 01:53.08 | strav | when pressing star, you'll loop the current block (with goto command), then having the menu told and executed again. |
| 01:54.06 | jdg | Sorry, I mean what does asterisk do when call comes in menu_principal ? |
| 01:54.06 | Poehali | anyone here configured SPA3102 before? |
| 01:55.41 | strav | jdg, waits for an extension. Then given the right extension, it conditionally calls another block of instructions. If the user press *, the block is looped and if the user waits too long, it hangs up. (timeout). |
| 01:56.53 | strav | jdg: w8 there's perhaps something wrong in what I pasted due to the changes I've been trying. |
| 01:57.38 | jdg | strav: I think I understand what you want to do, but what does actually happen. You say it doesn't work |
| 01:59.08 | strav | jdg: here is the whole thing as it is used to be: http://pastebin.ca/691155 |
| 01:59.40 | strav | jdg: now, my problem actually is that asterisk does not process the extension when given one (at the waitexten cmd) |
| 02:02.12 | CCFL_Man2 | i can't get this western electric dial working |
| 02:02.43 | TJNII | Which adapter? |
| 02:02.49 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 02:03.53 | strav | jdg: here is the exact output I get from the command line client (debug and verbose are normal): http://pastebin.ca/691161 |
| 02:05.26 | jdg | So it doesn't read your digits ? Could it be a DTMF configuration problem on the SIP channel ? |
| 02:06.13 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:07.21 | strav | perhaps. Though, if it's related, I didn't made any changes to my sip.conf since last time this config worked. |
| 02:13.54 | strav | jdg, if it's relevant, my account's dtmfmode in sip.conf is set to auto... |
| 02:14.17 | *** join/#asterisk thermalwetland (n=Matt@pele.comtelhi.com) |
| 02:15.40 | *** join/#asterisk hacim (n=micah@debian/developer/micah) |
| 02:15.49 | hacim | what did 'set verbose' get turned into? |
| 02:16.02 | JT | core set verbose, like a lot of stuff |
| 02:16.13 | jdg | I believe dtmfmode can only be set to: inband, rfc2833 or info |
| 02:16.33 | hacim | so: core set verbose 4 |
| 02:16.53 | kiscokid | Poehali: can you see the Proxy and Registration items in your SPA3102 menus? |
| 02:17.47 | hacim | hmm, I did 'core set verbose 4' and now 'help' doesn't work |
| 02:18.43 | Poehali | kiscokid: yes |
| 02:19.28 | strav | jdg, I'll try those with a barebone setup just to test, thanks |
| 02:19.44 | kiscokid | Poehli: that's were you fill in the info about your server and sip peer |
| 02:19.48 | *** join/#asterisk shido6 (n=shido6@74-130-227-15.dhcp.insightbb.com) |
| 02:20.28 | kiscokid | Poehli: for example, set Proxy and Outbound Proxy to the ip addr of your * server |
| 02:20.38 | Poehali | so sip peer is [user]? |
| 02:20.50 | kiscokid | yes |
| 02:21.04 | kiscokid | also authid |
| 02:21.16 | kiscokid | password is secret |
| 02:21.37 | kiscokid | Display name is whatever you want |
| 02:21.49 | hacim | i just upgraded from 1.2 to SVN and ported my config files, but I am not able to register via sip now |
| 02:21.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:22.10 | hacim | and even with 'core set verbose 10' I'm not seeing anything |
| 02:22.11 | JT | which SVN? |
| 02:23.00 | catch23 | Anyone here run asterisk under xen 3.1? |
| 02:23.30 | Poehali | kiscokid: okay I did that, sip show peers still show offline though |
| 02:23.38 | *** join/#asterisk pepo-- (n=pepOSX@190.72.151.54) |
| 02:23.57 | kiscokid | you may have to reboot the spa to get it to register |
| 02:24.27 | kiscokid | you should look at the * console to see if it registers |
| 02:25.22 | Poehali | sip show peers still show offline |
| 02:25.54 | Poehali | is there any other way I can check? |
| 02:26.04 | *** join/#asterisk melbert (n=mmelbert@pppoe83.vdsl.aspStation.net) |
| 02:26.06 | kiscokid | try making a call |
| 02:26.33 | Poehali | that won't work |
| 02:26.39 | Poehali | it always gives me busy signals |
| 02:27.18 | melbert | I am sure that this has been asked before but what does everyone do about redundancy? |
| 02:27.39 | thermalwetland | Anyone try to apply a patch for this bug? http://bugs.digium.com/view.php?id=4903 |
| 02:27.45 | thermalwetland | It allows SIP over TCP |
| 02:28.24 | melbert | more specifically what does everyone do for redundancy of your upstream provider? |
| 02:29.12 | melbert | we have been having lots of trouble with Verizon and we need a "backup" if there is such a thing |
| 02:29.13 | kiscokid | Poehali: guess I am stumped |
| 02:29.32 | strav | jdg, I tried all the dtmfmodes you suggested, digits aren't read so far. |
| 02:29.54 | kiscokid | Poehali: this article looks interesting: http://weblog.infoworld.com/venezia/archives/009482.html |
| 02:30.02 | JT | melbert: how are calls delivered from the upstream? |
| 02:30.21 | Poehali | kiscokid: me too |
| 02:30.50 | melbert | 6 PRIi's that have 3 toll free number in a hunt group |
| 02:31.21 | JT | see if they can have lines connected to more than one CO? |
| 02:31.40 | JT | but if the problems are upstream of that, that's really somerthing you need to solve with them |
| 02:32.44 | jdg | how is dtmf sent from the client ? |
| 02:32.56 | melbert | our last problem actually went beyond our local loop....We wanted incoming calls from Canda to be allowed on all of our numbers and they clobbered one of our toll free numbers in the process |
| 02:33.25 | melbert | it was down for about 7 hours before they resolved the issue |
| 02:33.28 | strav | jdg: I don't know. I gotta tell, this is the second time I play with asterisk. |
| 02:34.15 | JT | melbert: yeah asterisk can't fix that. |
| 02:34.27 | melbert | JT is there anyway to "failover" a particular 800 number to an alternate provider? |
| 02:34.35 | JT | i doubt it |
| 02:34.41 | JT | unless you ask your provider to do so |
| 02:34.51 | JT | if you ask them to forward it during a failure |
| 02:34.55 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 02:35.04 | melbert | I know asterisk cant fix it...I was just looking for some guidence to see if some is doing something like that |
| 02:35.09 | jdg | strav: well, you need to match client and asterisk settings. Also inband dtmf only works with alaw or ulaw |
| 02:36.01 | melbert | JT so it should be possible for them to forward the numbers somewhere else while the issue is being worked on? |
| 02:36.09 | JT | definitely |
| 02:36.11 | JT | easy |
| 02:36.21 | JT | (as long as you speak to the right people) |
| 02:36.34 | strav | jdg: I mostly used the sip settings offered by the client. As for the codecs, I'm only using ulaw right now as other may not fully work. |
| 02:37.10 | strav | hmmm. May I ask again to "public attention", is there any reason why the entered digits aren't read by the waitexten? |
| 02:37.54 | melbert | JT HA...that is a good one when you deal with Verizon |
| 02:38.12 | melbert | there is no "right" person at Verizon |
| 02:38.55 | JT | melbert: i found it very hard to speak to any human from verizon sales australia |
| 02:39.03 | JT | they don't seem very customer friendly :P |
| 02:40.20 | melbert | same thing here in the US |
| 02:41.03 | melbert | hmmm...so if we got a backup provider we could limp along until Verizon straightened themselves out.....hmmm |
| 02:41.29 | JT | yeah, it's a piece of piss for a telco to redirect a number |
| 02:42.23 | melbert | yeah...that it would be easy to add incoming calls from Canada as well...but they managed to bugger that up |
| 02:42.56 | *** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 02:49.31 | WilliamK | hey melbert, I know Verizon has a product called Disaster Routing as well as ATT does the same thing as well.... another option you could utilize is a provider that allows you to change where your 800# forwards to via a web interface |
| 02:50.00 | melbert | yeah |
| 02:50.29 | melbert | I guess we would need two providers giving us pri's / t1s and failover to the other when it goes down |
| 02:50.43 | WilliamK | I know a provider that's charging 2.9c/min for the 800# and has a web interface if you would like |
| 02:50.48 | WilliamK | no relation to me |
| 02:51.03 | melbert | do they provide t1's or pri's? |
| 02:51.24 | melbert | or is it a voip connection over the the internet? |
| 02:51.25 | WilliamK | for what area? |
| 02:51.31 | melbert | pittsburgh |
| 02:51.53 | n00dle | Ack! ztcfg-dude is giving me "ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)". I've already cleaned and recompiled... what next? |
| 02:51.55 | WilliamK | they actually do Long Distance, and 800# stuff nationwide, however they only do local access in select areas |
| 02:52.09 | melbert | who is it? |
| 02:52.19 | WilliamK | they resell Sprint and Global Crossing LD |
| 02:52.23 | WilliamK | Pioneer Telephone |
| 02:52.37 | melbert | ok |
| 02:52.54 | WilliamK | prepaid over the web, and you can get 800#s forwarded to any # you like |
| 02:53.38 | *** join/#asterisk [TK]D-Fender (n=joe_blow@64.235.216.2) |
| 02:54.10 | melbert | yeah...we are renegotiating our contract after 8 monthes with Verizon since we have had 2 outages over 6 hours apiece. |
| 02:54.31 | WilliamK | ouch, that hurts |
| 02:55.51 | melbert | yeah |
| 02:56.02 | WilliamK | I'm looking forward to the day already that I can get trunks to the tandems VS relying on the CLEC to carry all the traffic |
| 02:56.05 | hacim | I've got ztdummy module loaded, but asterisk isn't compiling meetme |
| 02:58.16 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@015-799-396.area5.spcsdns.net) |
| 03:00.45 | melbert | it would not be possible to have 1 800 be shared by two different providers in a round robin/load balancing config would it? |
| 03:02.06 | melbert | that way we could have two different providers and if one went down we would still have one .... just at half capacity |
| 03:02.27 | melbert | I think that is pipe dream....but worth an ask |
| 03:03.14 | [TK]D-Fender | melbert: You're afraid of your PROVIDER going down? thats a new paranoia record! don't forget to claim your trophy on the way out! |
| 03:03.32 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:03.42 | JT | melbert: only if you speak SS7, i would think |
| 03:04.48 | melbert | [TK]D-Fender, well they have gone 2 times in the past 8 monthes at over 6 hours in both of those incidents and few other time for an hour here or there....yes I am paranoid but with experience to back it up |
| 03:05.02 | melbert | SS&??? |
| 03:05.07 | JT | SS7 |
| 03:05.08 | melbert | what is SS7? |
| 03:05.14 | JT | Signalling System 7 |
| 03:05.17 | [TK]D-Fender | melbert: Maybe you should just SWITCH |
| 03:05.28 | melbert | cant...have a 3 year contract |
| 03:05.49 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 03:06.02 | JT | the signalling protocol that runs the majority of the global pstn :) |
| 03:06.14 | melbert | That was in place when I started working there |
| 03:06.18 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 03:06.30 | melbert | JT is that possible to do with asterisk? |
| 03:06.39 | melbert | what kind of equipment is needed for it? |
| 03:07.15 | JT | there are addons |
| 03:07.18 | JT | but "not really" |
| 03:07.23 | JT | and you need to be a telco |
| 03:07.32 | melbert | ha....the catch |
| 03:07.39 | JT | and have your ss7 setup certified before anyone will interconnect |
| 03:07.46 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 03:07.52 | JT | some countries use ISUP over SS7 to end users |
| 03:07.52 | melbert | ok...so not possible for me to do |
| 03:07.55 | JT | but that's rare |
| 03:09.03 | hacim | what do I need to do to get zaptel support detected so I can get asterisk to build the meetme application |
| 03:09.22 | JT | install zaptel? |
| 03:09.24 | melbert | so for me it looks like the best thing to do is get some backup service and push the 800 number to it in case of a failure on the primary side |
| 03:10.01 | osiris | i have registratiom with my sip provider, but any idea why i still get there intercept when calling in ? |
| 03:12.11 | n00dle | hacim, make and install zaptel before making and installing asterisk. |
| 03:12.31 | hacim | n00dle: I did that... |
| 03:13.38 | n00dle | Hm. |
| 03:15.21 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.129.156) |
| 03:24.56 | *** join/#asterisk f00bar80 (n=mina@196.202.92.139) |
| 03:27.22 | *** join/#asterisk Strom_C (n=strom@216.64.24.250) |
| 03:30.49 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 03:36.18 | hacim | hmm if I do ./configure --with-zaptel=/usr/include/linux (because zaptel.h is there), it tries to add a /include at the end so its: -I/usr/include/linux/zaptel.h/include |
| 03:37.58 | f00bar80 | i'm aksing about which requirements i need to setup a VOIP gateway on my hosting server , and if i need a SIP account/proxy and any extra software/hardware. \ |
| 03:38.54 | *** join/#asterisk Juggie (n=Juggie@CPE0013460a7821-CM001371886eee.cpe.net.cable.rogers.com) |
| 03:44.51 | hacim | the problem is that asterisk is failing to find the zaptel source |
| 03:48.35 | *** join/#asterisk brad[] (n=brad@gentoo/developer/brad) |
| 03:49.23 | brad[] | hi folks, I'm noticing that if I begin a call between two SIP phones (GXP-2000's in this case) and unexpectedly power off both phones mid-call, the call stays open indefinitely on the asterisk side until I intervene. Any workaround to this? |
| 03:50.51 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-112.lv.lv.cox.net) |
| 03:52.36 | *** join/#asterisk bmg505 (n=leon@196.209.183.7) |
| 03:56.07 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 03:56.53 | tzafrir_laptop | hacim, ./configure --with-zaptel=/usr |
| 03:58.47 | JT | brad[]: rtp timeout |
| 03:58.57 | JT | but how often do they power off? |
| 03:59.37 | brad[] | rarely if ever |
| 03:59.43 | brad[] | Just a corner case I don't want to hit |
| 03:59.47 | brad[] | JT: Great idea, thanks |
| 04:00.02 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.11 | hacim | tzafrir_laptop: when I do anything I get this error: |
| 04:00.12 | hacim | checking for ZT_TONE_DTMF_BASE in zaptel/zaptel.h... ./configure: line 32473: -I/usr/include: No such file or directory |
| 04:00.53 | tzafrir_laptop | hacim, which version of asterisk is it? which version of zaptel? |
| 04:00.59 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 04:01.26 | tzafrir_laptop | ah... |
| 04:01.50 | tzafrir_laptop | it was looking for tonezone.h , which should be in /usr/include/zaptel.h as well |
| 04:02.41 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 04:05.46 | hacim | tzafrir_laptop: well the location is /usr/include/linux/zaptel |
| 04:06.12 | hacim | tzafrir_laptop: i'm using svn r61760, but trying on debian etch |
| 04:14.51 | hacim | tzafrir_laptop: thanks I got it |
| 04:15.38 | tzafrir_laptop | what was it? |
| 04:21.34 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:23.17 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
| 04:24.13 | Teln1100A | do I need zaptel if all I will be using asterisk for is voip internet stuff, no hardware interface cards? |
| 04:24.59 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 04:25.56 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 04:28.51 | jablko | Teln1100A: no, zaptel will not be required |
| 04:29.52 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:35.10 | jablko | i have a zaptel fxo interface |
| 04:35.47 | jablko | when the zaptel channel rings, is it possible for asterisk to ring another channel (SIP) |
| 04:36.11 | jablko | without actually picking up the zaptel channel? |
| 04:36.29 | jablko | and only pickup the zaptel channel if the SIP channel is picked up? |
| 04:43.38 | DrAk0 | Teln1100A, only if you want make conferences |
| 04:44.20 | DrAk0 | jablko, try not pussing Answer on the zaptel dialplan |
| 04:44.25 | DrAk0 | jablko, just call |
| 04:44.26 | [TK]D-Fender | jablko, Yes. just go right ahead and start with Dial. |
| 04:45.23 | jablko | DrAk0: [TK]D-Fender: awesome, much thanks! |
| 04:46.26 | *** join/#asterisk watchy (n=watchy@h120.184.255.206.cable.cmdn.cablelynx.com) |
| 04:46.31 | watchy | hey tk you there? |
| 04:46.38 | [TK]D-Fender | watchy, yup |
| 04:46.44 | watchy | you sell phone systems right? |
| 04:48.08 | watchy | do you setup * boxes so customers can admin them themselves |
| 04:48.15 | watchy | web gui etc? |
| 04:50.26 | [TK]D-Fender | watchy, I don't sell hardware, I merely advise on it. I sell my SERVICES, which have never involved installing a GUI. I set up custom fron scratch system that aid in their techs LEARNING *. |
| 04:51.04 | watchy | oh so most companies you sell * to have phone admins? |
| 04:52.57 | jablko | sorry: if i don't answer, just Dial, will asterisk automatically "Answer" the zaptel interface when the SIP channel is answered, or do i need to somehow do that manually? |
| 04:53.33 | [TK]D-Fender | jablko, Yes, its automatic |
| 04:53.41 | jablko | [TK]D-Fender: awesome, thanks |
| 04:53.43 | [TK]D-Fender | watchy, yes |
| 04:56.42 | *** join/#asterisk Strom_C (n=strom@216.64.24.250) |
| 04:58.07 | J4k3 | watchy: remote admin for * isn't that hard if you're using server-configurable hardware |
| 04:58.53 | J4k3 | "we're adding a new desk to accounting, the new phone's mac is xx... |
| 04:58.54 | J4k3 | " |
| 04:59.34 | watchy | well the co i work for wants to setup trixbox and allow them to admin everything |
| 04:59.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:59.42 | watchy | i kinda think to a point its a bad idea |
| 04:59.47 | watchy | no residual income you know? |
| 05:00.05 | JT | actually |
| 05:00.06 | watchy | how do you feel about my theory? |
| 05:00.13 | JT | if you charge per hour after that |
| 05:00.15 | JT | it's a good idea |
| 05:00.22 | JT | as long as you have a strict agreement |
| 05:00.34 | JT | as they will surely screw things up futzing about in trixbox |
| 05:00.34 | watchy | yea. JT do you setup gui's for customers? |
| 05:00.38 | watchy | haha |
| 05:00.39 | JT | and you will need to fix it |
| 05:01.06 | [TK]D-Fender | watchy, I'm not about "residual income" personally. I set my systems up with the INTENT that they will take the reigns. |
| 05:01.13 | JT | nah, but i'm thinking of making a custom one with very limited things to control |
| 05:01.24 | [TK]D-Fender | watchy, And on bigger things I get called back, hardware maintenance, etc |
| 05:01.36 | watchy | tk: yea but your systems have REAL admins, not fucking retards in SALES or Billing |
| 05:01.58 | watchy | "hey bob in accounting, wanna try to admin our phone system" |
| 05:02.03 | watchy | thats the idiots i'd be dealing with |
| 05:02.08 | [TK]D-Fender | watchy, I do have a client or two who remains rather ignorant of * and I do the changes for. |
| 05:02.17 | JT | "numbers, i LIKE numbers!" |
| 05:02.29 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 05:02.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:03.25 | watchy | haha |
| 05:06.04 | watchy | what sucks to a point though is one of the techs i work with at work is dependant on trixbox |
| 05:07.22 | watchy | i know i aint no * jedi |
| 05:07.40 | watchy | but i have setup a complete * setup at a decently sized install manually |
| 05:07.45 | watchy | i didn't find it that hard myself |
| 05:08.10 | watchy | i had to ask TK some things but i learned alot by doing it all in .confs |
| 05:08.25 | watchy | you sure aint gonna learn shit about * haxoring a gui |
| 05:10.45 | J4k3 | yeah |
| 05:10.47 | J4k3 | very true |
| 05:10.55 | J4k3 | it depends on what your goal is |
| 05:11.10 | J4k3 | when I set up the phone system here, I needed something to work immediately |
| 05:11.10 | watchy | yea |
| 05:11.34 | watchy | i never setup trixbox but i'm sure you could have it working in an hour |
| 05:11.48 | J4k3 | I had no way to wait... I was able to get an instant-activated local DID and my line forwarded (programmed by the telco directly, as all my POTS lines were out) |
| 05:12.04 | J4k3 | so I could get my business back online |
| 05:12.15 | J4k3 | the next day I canceled all but the first pots # |
| 05:12.25 | J4k3 | left the CF programmed, and never looked back. |
| 05:12.36 | [TK]D-Fender | J4k3, And how long does it take to build an * system from a decent template? |
| 05:12.49 | J4k3 | no idea |
| 05:13.14 | J4k3 | I got trixbox up in about 4 hours... I need to actively learn * so I can migrate away from it |
| 05:13.21 | watchy | i think i could build a * box in a few hours from scratch |
| 05:13.25 | watchy | without gui now |
| 05:13.28 | J4k3 | now, I spent the next 2 months getting everything to work right, mostly due to damaged lan switches |
| 05:13.30 | watchy | fresh configs etc |
| 05:13.36 | J4k3 | yeah |
| 05:13.46 | J4k3 | thats what I'm working toward |
| 05:14.04 | watchy | but I had to learn asterisk without a gui myself because i told this co i'd setup a system and they were ok with it |
| 05:14.17 | watchy | so i installed it over the weekend and had incoming calls working monday |
| 05:14.18 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 05:14.37 | watchy | now i've learned enough this company has hired me to start doing * installs and maintains |
| 05:15.16 | watchy | jake: i think our wimax is in the 4ghz range if im correct? |
| 05:15.26 | J4k3 | watchy: 3.5? |
| 05:15.27 | watchy | i just remembered that |
| 05:15.30 | J4k3 | 4ghz is usually satellite |
| 05:15.32 | watchy | yea i think so |
| 05:15.43 | watchy | is that good or bad |
| 05:15.49 | J4k3 | sketchy |
| 05:15.52 | J4k3 | its gonna need LOS |
| 05:15.58 | watchy | oh that sucks |
| 05:16.09 | J4k3 | anything about 1.2 ghz or so |
| 05:16.16 | J4k3 | is pretty sketchy when it comes to NLOS |
| 05:16.32 | watchy | so whats the use of wimax at 3.5 |
| 05:17.21 | J4k3 | my interpretation of it is basically "802.11g with hard timeslots" |
| 05:17.43 | J4k3 | of course, its still faith-based. you have to assume the client-ends are going to act right |
| 05:18.12 | watchy | well atleast we got the freqs for free |
| 05:18.16 | J4k3 | and well, anyone thats ever used a TDMA/GSM phone knows that sometimes things don't work as they should |
| 05:18.20 | J4k3 | not bad |
| 05:18.24 | J4k3 | worth every penny ;) |
| 05:18.34 | watchy | we setup wifi across a college |
| 05:18.46 | watchy | you heard of colubris? |
| 05:19.17 | J4k3 | they sound expensive |
| 05:19.58 | watchy | yea but its kinda neat |
| 05:20.20 | watchy | it controls all of the AP's on the college campus making sure they arent killing each other frequency wise |
| 05:20.38 | watchy | + it allows centralized authentication using Radius |
| 05:20.43 | CCFL_Man2 | you would not believe how well the mechanics of this western electric 5H dial is |
| 05:20.51 | JT | watchy: got the freqs for free? |
| 05:20.58 | watchy | jt: wimax |
| 05:21.24 | J4k3 | tpc+dfs = good |
| 05:21.24 | JT | as if they're free |
| 05:22.05 | watchy | well the local college got wimax freqs for free |
| 05:22.08 | watchy | we traded for them |
| 05:24.42 | dan__t | jfc. |
| 05:24.51 | dan__t | i just want one single spare box here, on which to install asterisk. |
| 05:25.12 | dan__t | How come all this shit has to be broken. Time to go all Office Space on some stuff |
| 05:25.22 | J4k3 | might be too anemic... no transcoding, pure g729 |
| 05:26.13 | watchy | wtf is a 405GP? |
| 05:26.20 | watchy | a router box? |
| 05:28.04 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 05:29.58 | watchy | i need some warez for my iphone |
| 05:30.26 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 05:30.41 | dan__t | watchy |
| 05:30.54 | dan__t | Do you still have a crush on Sabrina |
| 05:30.59 | watchy | sabrina? |
| 05:31.17 | watchy | bos wife? |
| 05:31.20 | bintut | anyone here able to make the asterisk blf work for the grandstream gxp-2000 on an asterisk-1.2.10 ? |
| 05:31.21 | dan__t | ahahaha. |
| 05:31.30 | watchy | i never had a crush on her |
| 05:31.33 | dan__t | Bullshit. |
| 05:31.44 | watchy | i didn't she wasnt my type of chick |
| 05:31.47 | dan__t | I hate you so much. |
| 05:31.48 | watchy | not even close |
| 05:31.50 | bintut | i can't make the LEDs blink.. :( |
| 05:31.51 | dan__t | What have you been up to |
| 05:31.57 | watchy | dan: getting rich |
| 05:32.03 | dan__t | right. |
| 05:32.18 | watchy | yea i know |
| 05:32.24 | watchy | my parents already had that covered |
| 05:32.28 | dan__t | weren't you trying that like 5 years ago too :< |
| 05:32.34 | watchy | nah |
| 05:32.47 | watchy | my parents pay all my billsg |
| 05:33.01 | dan__t | Wish my parents liked me. |
| 05:33.23 | watchy | man i wish i could find a tripple LCD stand that fit new lcds |
| 05:33.49 | dan__t | me too. |
| 05:34.14 | watchy | i need one to hold 2 22inch wide screens and 1 30inch in the center |
| 05:34.22 | watchy | but they are all to small |
| 05:34.33 | dan__t | i just need one for these three 2005FPW's |
| 05:34.41 | dan__t | that would tickle my fancy. |
| 05:34.48 | watchy | ergotron makes some nice ones but they don't fit to many wide screens |
| 05:34.54 | dan__t | that's nice. |
| 05:36.09 | watchy | you got a iphone dan |
| 05:36.15 | dan__t | No, I'm straight. |
| 05:36.39 | watchy | i doubt iphones are very popular in AZ |
| 05:36.51 | watchy | i didnt see many ATT/cingular stores before i left |
| 05:37.52 | dan__t | That's nice. |
| 05:38.08 | watchy | did you ever bang saebbe |
| 05:38.13 | sparq | Hey, does anyone have a USB handset unit that they like? |
| 05:38.26 | watchy | and after she got knocked up don't count |
| 05:38.36 | dan__t | no, she reminded me of a beaver. |
| 05:38.40 | dan__t | and i don't dig beavers. |
| 05:38.43 | J4k3 | mmm beavers |
| 05:38.45 | watchy | she was nice |
| 05:38.50 | watchy | but strange |
| 05:38.50 | dan__t | no she wasn't |
| 05:38.52 | dan__t | yes |
| 05:38.53 | dan__t | very strange |
| 05:39.10 | dan__t | i remember i went over to her house once at like 3am and she was all like acting as if she wanted to ride the dan |
| 05:39.19 | watchy | haha |
| 05:39.25 | dan__t | then we get friendly and she's all "i'm hungry" and i'm trying to figure out what i did wrong |
| 05:39.35 | J4k3 | tease ass girls |
| 05:39.35 | J4k3 | suck |
| 05:39.40 | J4k3 | gah, they suck |
| 05:39.40 | dan__t | and she said something to the effect of like "oh yeah i'm celibate." |
| 05:39.42 | dan__t | Then I left. |
| 05:39.47 | watchy | thats my gs ex wife jake. you better simmer |
| 05:39.50 | watchy | j/k |
| 05:39.59 | watchy | hahaha |
| 05:40.06 | dan__t | I just.. left. |
| 05:40.06 | watchy | saebbe was a wierd girl dude |
| 05:40.09 | dan__t | yes |
| 05:40.13 | watchy | i miss hanging out with her |
| 05:40.19 | dan__t | ... |
| 05:40.29 | watchy | she was always nice to me |
| 05:40.30 | dan__t | i went out drinking a few mos ago with sabrina and caitlyn |
| 05:40.34 | dan__t | sabrina CAN DRINK. |
| 05:40.40 | watchy | caitlyn is likwids ex? |
| 05:40.53 | dan__t | yeah, she's like my cousin kindof not really :/ |
| 05:41.03 | watchy | i call her neopet girl |
| 05:41.07 | J4k3 | haha |
| 05:41.07 | dan__t | wtf? |
| 05:41.08 | J4k3 | neopets |
| 05:41.31 | watchy | when i was being investigated by the fbi i stayed at likwids a few nights |
| 05:41.32 | dan__t | you remember Caitlyn's friend Katie with the big knockers |
| 05:41.37 | *** join/#asterisk Blackthorn (n=support@76-77-161-226.smyth.net) |
| 05:41.38 | watchy | and me and cait would play neopets |
| 05:41.44 | dan__t | that's gay. |
| 05:41.45 | dan__t | really. |
| 05:41.56 | watchy | yea so now i call her neopet girl |
| 05:41.59 | dan__t | Did you have a dude neopet named Wilbert? |
| 05:42.01 | watchy | even though i never talk to her |
| 05:42.30 | bintut | anyone here using the GrandStream Phone GXP-2000 or GXP-2020 and made the LEDs work through the Asterisk BLF on an Asterisk-1.2.10 ? |
| 05:42.31 | Blackthorn | Hi, thought I would do a quick check. Anyone have a 1.4.4 server crash with [Sep 11 01:31:46] WARNING[13808]: app_dial.c:674 wait_for_answer: Unable to forward voice frame ? |
| 05:42.31 | dan__t | ah well. those were the days. |
| 05:42.39 | dan__t | I drove through BCS yesterday and thought of Sean |
| 05:42.42 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:42.45 | dan__t | I'm like. Where is that motherfscker. |
| 05:42.45 | watchy | heh |
| 05:42.46 | J4k3 | watchy: f b i? |
| 05:42.58 | watchy | j4k3: yea i did stupid shit |
| 05:43.04 | J4k3 | bcs? bryan/college station? :) |
| 05:43.08 | dan__t | One of our customers got popped bigtime here in Phoenix by the FBI and IRS and Postal Inspectors. |
| 05:43.13 | Blackthorn | that message does a continus scroll down the system and stops taking calls until I stop and restart asterisk. |
| 05:43.18 | J4k3 | ack |
| 05:43.20 | watchy | dan what did they do? |
| 05:43.28 | dan__t | Long list of things heh. |
| 05:43.39 | dan__t | Those FBI guys are pretty smart. |
| 05:43.45 | CCFL_Man2 | i knew taking this dial apart would be a bad idea |
| 05:43.46 | dan__t | Do you know they can fit 1.2TB on a thumbdrive? |
| 05:43.52 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 05:44.40 | watchy | dan: haha |
| 05:44.52 | dan__t | Well no that was the postal inspectors office. |
| 05:44.58 | watchy | haha |
| 05:44.59 | dan__t | the fbi guy was a really really nice guy, very funny. |
| 05:45.00 | threat | WARNING[7243]: chan_zap.c:1592 zt_set_hook: zt hook failed: Device or resource busy |
| 05:45.17 | dan__t | he took us out to lunch. |
| 05:45.19 | watchy | well i'm glad i never got busted by the fbi |
| 05:45.21 | dan__t | because, well, i already paid for it. |
| 05:45.23 | watchy | but i sure got close |
| 05:45.43 | dan__t | yeah i wouldn't run warez on a ... network either. |
| 05:45.50 | dan__t | but hey that's just me |
| 05:46.01 | watchy | well it wasnt that reason we got caught |
| 05:46.04 | watchy | we had a narq |
| 05:46.10 | dan__t | did you murder him |
| 05:46.29 | watchy | we woulda never got busted had that dude not narqed though |
| 05:46.40 | dan__t | So, did you murder him? |
| 05:46.56 | watchy | no hes pretty well protected |
| 05:47.04 | dan__t | haha as he should be. |
| 05:47.22 | watchy | but them was the best times of my life |
| 05:47.29 | dan__t | uh yeah mine too. |
| 05:47.50 | watchy | i'd do it again in a second |
| 05:48.02 | dan__t | im bored. |
| 05:48.12 | dan__t | I was trying to roll CentOS5 on the sparc |
| 05:48.13 | watchy | im watching porn on tv |
| 05:48.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:49.45 | sparq | CCFL_Man2: Hey -- do you know of any USB handset units are any good? |
| 05:50.08 | Blackthorn | Hi, thought I would do a quick check. Anyone have a 1.4.4 server crash with [Sep 11 01:31:46] WARNING[13808]: app_dial.c:674 wait_for_answer: Unable to forward voice frame ? |
| 05:50.23 | Blackthorn | that message does a continus scroll down the system and stops taking calls until I stop and restart asterisk. |
| 05:50.36 | Blackthorn | any ideas, some place to start... |
| 05:51.03 | sparq | Blackthorn: what sort of dialplan are you using? |
| 05:52.09 | Blackthorn | well.. i have extensions for information, and 911 calling, I have both normal ported telephone numbers, and ld service dialing through both a local pri and voicepulse |
| 05:52.42 | Blackthorn | pretty basic dialplan with about 20 entries. |
| 05:53.35 | sparq | Blackthorn: So, you've got a POTS adapter or two? |
| 05:54.14 | JT | sparq: he never mentioned POTS. |
| 05:55.33 | sparq | JT: I was hoping that's what he meant by normal ported telephone numbers ^_^ |
| 05:56.02 | Blackthorn | I have one port pri card. on that pri are did numbers as well as normal numbers that were ported over from the local telco. |
| 05:56.15 | JT | sparq: he said PRI though |
| 05:56.23 | Blackthorn | we have a new server + moved to 1.4.4 (from a 1.2 box) |
| 05:56.33 | JT | doesn't make much sense to pull DIDs over POTS |
| 05:57.32 | Blackthorn | And this box seems to fail every 24 to 72 hours with a continues scrolling message posted above. Can just stop and restart the box and goes away untill next ime. |
| 05:57.36 | sparq | I was just hoping it would be a kernel driver problem. ^_^ |
| 05:58.18 | sparq | Blackthorn: Does dmesg say anything interesting when that happens? |
| 06:01.18 | Blackthorn | nope.. i checked the message log and it will for example show xxx sip phone is now connected. go for some lenth of time then that message just repeats itselfone after another very quickly. |
| 06:01.43 | Blackthorn | i did find this link that shows the exact match for the error way down the page but it's in german http://www.ip-phone-forum.de/showthread.php?t=143013 |
| 06:03.52 | sparq | A codec problem, maybe? |
| 06:04.15 | Blackthorn | as far as i know i've got everything set for ulaw |
| 06:05.08 | JT | Blackthorn: why are you using such an old version of 1.4? |
| 06:05.33 | Blackthorn | er sorry thats what I was using last week. i am using 1.4.11 now |
| 06:06.17 | JT | i see |
| 06:06.23 | JT | did you upgrade zaptel? |
| 06:07.21 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 06:07.26 | Blackthorn | using zaptel 1.4.5.1 |
| 06:07.27 | *** part/#asterisk bintut (n=bintut@203.125.63.150) |
| 06:08.14 | sparq | Blackthorn: the forum discussion seems to suggest that the problem occured with the negotiated codec was ulaw, but went away when they switched to gsm. |
| 06:09.34 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:10.19 | Blackthorn | umm |
| 06:10.42 | Blackthorn | alrighty. well i'd better get off to bed. thanks for the help. |
| 06:11.49 | sparq | g'nite |
| 06:12.08 | CCFL_Man2 | i can't figure out how to get this western electric dial back together |
| 06:13.26 | threat | grrrr, phone keypad still isn't working. now what? |
| 06:13.49 | CCFL_Man2 | threat: my dial isn't working either :P |
| 06:14.38 | sparq | CCFL_Man2: You are using a 1920's candelstick style phone with Asterisk? |
| 06:15.03 | CCFL_Man2 | sparq: 1946 western electric 302 for now |
| 06:15.19 | sparq | That is awesome. |
| 06:15.33 | CCFL_Man2 | no reason why candlestick phones wouldn't work though |
| 06:15.43 | CCFL_Man2 | yeah, the 302 kicks ass |
| 06:15.44 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:15.51 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:16.31 | CCFL_Man2 | problem now is that i can't get it's dial working |
| 06:16.38 | sparq | CCFL_Man2: I have one of these sitting in a box somewhere. I bought it at a garage sale for $10. Sadly, it needs to be completely rebuilt. http://i22.photobucket.com/albums/b347/Parashuut/wecsbw.jpg |
| 06:16.45 | CCFL_Man2 | i disasembled it |
| 06:17.07 | CCFL_Man2 | sparq: the AL50> |
| 06:17.10 | CCFL_Man2 | err |
| 06:17.14 | CCFL_Man2 | AL50 |
| 06:18.26 | sparq | CCFL_Man2: It's this thing, right? http://en.wikipedia.org/wiki/Model_302_telephone |
| 06:18.33 | CCFL_Man2 | yeah |
| 06:18.49 | sparq | beautiful |
| 06:19.59 | sparq | is it stuck, or just won't return? |
| 06:20.48 | CCFL_Man2 | i can't figire out how to tention the spring |
| 06:21.15 | CCFL_Man2 | the spring that tensions the rotor |
| 06:21.59 | CCFL_Man2 | does the flywheel determin the tension of the return? |
| 06:21.59 | threat | heh |
| 06:23.35 | sparq | CCFL_Man2: I vaugly remember destroying my grandparents' rotary phone when I was a little. I seem to recall that I had to hold everything together just so while puting it back together, or the dial would just stay stuck all the way over. |
| 06:24.37 | CCFL_Man2 | sparq: see, i'm not sure if i need to rotate the rotor fully and keep that tension or just in it's free state |
| 06:24.43 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:24.48 | WilliamK | sparq, my grandparents still have a rotary phone |
| 06:24.52 | WilliamK | working one too |
| 06:25.13 | CCFL_Man2 | WilliamK: why would a western electric phone ever break? |
| 06:25.32 | WilliamK | it'd be a conspiracy by ATT if it ever did |
| 06:25.35 | CaT[tm] | because it is a device of the infidel. |
| 06:26.12 | CCFL_Man2 | WilliamK: yeah, because western electric means quality |
| 06:26.36 | CCFL_Man2 | the quality of these gears in the dial just blows me away |
| 06:27.08 | sparq | WilliamK: My greaut aunt still pays SBC a couple of bucks a month because she refuses to get touchtone service. She is convinced that it's "more expensive" and "fancy," even though they charge her extra not to have it. |
| 06:27.49 | JT | why do they charge more? |
| 06:28.10 | CCFL_Man2 | rent of the phone |
| 06:28.26 | JT | hah |
| 06:28.26 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 06:28.37 | JT | the antique phone charge |
| 06:28.45 | kiscokid | extra cost of maintaining pulse dial on her pair? |
| 06:28.52 | sparq | I have no idea. They badgered her for a few years about "upgrading," so she got used to thinking of touchtone service as an extra feature. |
| 06:29.17 | JT | kiscokid: no |
| 06:29.30 | JT | all DTMF capable switches can do decadic dialling too |
| 06:29.46 | JT | otherwise how on earth would hookflash work? |
| 06:29.48 | CCFL_Man2 | as well as channel banks |
| 06:30.30 | kiscokid | I wonder if my Cisco ATA can handle it |
| 06:32.03 | CCFL_Man2 | the linkshit pap2 cannot |
| 06:37.31 | CCFL_Man2 | but i wouldn't expect anything less |
| 06:38.11 | CCFL_Man2 | it's a shame you need to buy professional equipment to get the same quality you did years ago with consumer equipment |
| 06:39.05 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 06:41.56 | CCFL_Man2 | shit shit shit i can't get this dial back the way it was |
| 06:51.04 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 06:53.14 | sparq | CCFL_Man2: I always have better luck putting things back together in the daytime... |
| 06:55.22 | CCFL_Man2 | sparq: i'm gonna see if my boss can do it |
| 06:56.05 | CCFL_Man2 | i have no diagram and really don't know how they go together |
| 06:58.10 | sparq | CCFL_Man2: why did you take it apart in the first place? |
| 06:59.32 | CCFL_Man2 | sparq: it pulsed too fast or slow, my channel bank wouldn't reconize anything other than 1 |
| 06:59.53 | CCFL_Man2 | and in the case of a 1 it really doesn't matter |
| 07:00.03 | CCFL_Man2 | it would return too fast i think |
| 07:01.45 | sparq | oi. |
| 07:03.24 | CCFL_Man2 | i have a pink trimline that it reconizes fine |
| 07:04.15 | sparq | CCFL_Man2: Don't those old phones use resistive microphones? |
| 07:04.24 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:04.30 | CCFL_Man2 | they do |
| 07:04.44 | CCFL_Man2 | but they don't crackle |
| 07:04.50 | sparq | I seem to remember that they draw a significant current |
| 07:04.54 | CCFL_Man2 | atleast not WE |
| 07:05.33 | CCFL_Man2 | the pink trimline uses the same microphone |
| 07:05.55 | CCFL_Man2 | similar network too, upgraded, naturally |
| 07:07.53 | sparq | I'm trying to figure out if an obscure (and therefor cheap) USB handset will work on Linux, but Google is failing me. |
| 07:08.31 | CCFL_Man2 | you don't want any of that crap |
| 07:08.48 | CCFL_Man2 | restore that candlestick |
| 07:11.08 | sparq | CCFL_Man2: Yes, but then I'd need to either buy an ATA, or figure out how to get the generic firmware to work on my (now useless) Packet8 DTA-310. |
| 07:11.35 | sparq | $9.99 for a handset is hard to beat. ^_^ |
| 07:11.42 | JT | what the hell |
| 07:11.52 | JT | why not spend $9 on a pc headset? |
| 07:11.58 | JT | it does the same job |
| 07:11.59 | JT | only better |
| 07:12.38 | sparq | JT: I have one, but it bothers me for some reason. |
| 07:12.54 | JT | usb handsets are just soundcards with buttons |
| 07:13.02 | JT | they're silly |
| 07:13.09 | JT | they rely on softphones |
| 07:13.18 | sparq | of course |
| 07:13.36 | JT | headsets are far superior |
| 07:13.41 | *** join/#asterisk LukinoVoip (n=LukinoVo@gw.abanet.it) |
| 07:14.01 | sparq | JT: Unless they make your ear itch |
| 07:14.14 | JT | then get better ones |
| 07:14.24 | JT | being handsfree is really useful on long calls |
| 07:16.01 | sparq | I guess you are right. I just feel like an ass talking into the air with an ugly little plastic shrimp hanging on my ear. |
| 07:16.43 | JT | or get an ip phone... |
| 07:17.31 | *** join/#asterisk appelza (n=d@dsl-240-133-188.telkomadsl.co.za) |
| 07:17.54 | LukinoVoip | hi all, i have a setup like this: FAX T38=>ATA => AST =>iax => AST =>PRI =>PBX=>PSTN...How gen i get the fax to work? :S |
| 07:18.07 | appelza | Hi guys, my analog 'trunk' is called Zap/g1 , how can I find out which other trunks are available to me and what they are called? |
| 07:18.10 | appelza | like Zap/g2 perhaps |
| 07:18.33 | JT | LukinoVoip: get off the crack pipe maybe? ;) |
| 07:18.49 | LukinoVoip | :D |
| 07:19.02 | FlatFoot | morning all |
| 07:19.09 | JT | asterisk doesn't do T.38 termination |
| 07:19.17 | LukinoVoip | passtrough |
| 07:19.39 | JT | you said AST to PRI |
| 07:19.43 | JT | that is not passthrough |
| 07:20.32 | LukinoVoip | sorry, the fax is not working...i'm searching a solution to get it work |
| 07:20.48 | appelza | anyone? :< |
| 07:20.50 | JT | i just said it won't work using that setup... |
| 07:21.13 | LukinoVoip | :s |
| 07:21.25 | JT | appelza: checking the zapata configuration files you made |
| 07:21.32 | LukinoVoip | i'm very worried... |
| 07:21.59 | JT | LukinoVoip: of course, you did check BEFORE embarking on this project that it was even possible in Asterisk? |
| 07:23.12 | LukinoVoip | the project is already set up...but someone asks me if there is a chanche to use the fax too |
| 07:23.38 | LukinoVoip | and i'm searching for that |
| 07:23.44 | appelza | JT, I see "group 0,15" |
| 07:23.51 | appelza | does that mean I must use Zap/g15 ? |
| 07:23.59 | JT | appelza: did you set it up or not |
| 07:24.05 | JT | no, it means there are 2 groups |
| 07:24.06 | appelza | genzaptelconf |
| 07:24.09 | JT | for that channel |
| 07:24.23 | watchy | anyone here ever use virtual pc? |
| 07:24.25 | JT | LukinoVoip: it depends how you do faxing |
| 07:24.27 | appelza | ok |
| 07:24.32 | JT | but generally the answer is "not really" |
| 07:25.30 | LukinoVoip | the setup working include phones that are working..the fax is connected to the ata but naturally is not working at now... |
| 07:25.56 | JT | phones are completely different to faxes |
| 07:26.04 | LukinoVoip | i know... |
| 07:26.04 | JT | the fact that phones work is an irrelevance |
| 07:27.42 | LukinoVoip | if i use FAX => ATA => SIP =>T38 EXT PROVIDER ...can it works? |
| 07:28.03 | watchy | fax in asterisk apparently is a bitch lukin |
| 07:28.05 | watchy | :/ |
| 07:28.46 | JT | LukinoVoip: the answer there is "hopefully" |
| 07:29.01 | watchy | hahaha |
| 07:29.31 | LukinoVoip | uhm...maybe i will setup an Hylafax server ;) |
| 07:30.03 | watchy | thats pretty much the best option from what i understand lukin |
| 07:30.17 | LukinoVoip | and throw faxes out of the window |
| 07:31.19 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 07:31.21 | JT | 1.4 has T.38 passthrough |
| 07:31.23 | appelza | how would an outbound extention look like for this: |
| 07:31.25 | appelza | http://pastie.caboo.se/96007 |
| 07:31.38 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:31.39 | JT | if your ATA is ok, and your Internet connectivity isn't awful, it should work |
| 07:34.20 | appelza | anyone? :< |
| 07:34.30 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 07:34.57 | *** join/#asterisk Juggie (n=Juggie@wlanportal.aliant.net) |
| 07:35.12 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 07:38.03 | RyanW | Hello, i'm configuring FOP for the first time, i've got the server configured, running and talking to asterisk and i've got the extensions displayed in my browser. But when i dial an extension, its state in FOP does not change. |
| 07:38.18 | watchy | fop? |
| 07:38.26 | JT | flash operator panel |
| 07:38.31 | watchy | wtf is that |
| 07:38.41 | LukinoVoip | Jt: thanks a lot |
| 07:38.58 | JT | watchy: an operator panel... that uses flash |
| 07:39.03 | RyanW | What should i double check to find where i went wrong? |
| 07:39.29 | watchy | jt: got a url for it |
| 07:39.56 | JT | should be googleable |
| 07:40.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:40.16 | appelza | can anyone tell me how an outbound extention should look like for this type of channel: http://pastie.caboo.se/96007 please |
| 07:40.23 | RyanW | JT, if i pastebin some config files will you give me a hand please. |
| 07:40.41 | watchy | yea it was |
| 07:40.46 | watchy | man this intresting |
| 07:41.08 | watchy | fop looks neat |
| 07:42.00 | JT | except it uses flash :P |
| 07:42.19 | watchy | jt: what would you recommend for an operator panel? |
| 07:42.21 | JT | RyanW: sorry don't have much FOP experience |
| 07:42.38 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 07:42.46 | JT | watchy: was thinking of making my own that uses AJAX/Comet one day |
| 07:42.52 | JT | but none atm :P |
| 07:43.10 | watchy | my co work introduced me to one called HUDLite or something |
| 07:44.09 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:45.16 | watchy | you seen it jt? |
| 07:48.12 | tzafrir | FOP is nice. But its development kind of lost momentum. |
| 07:48.22 | tzafrir | And the usage of flash is a big limitation |
| 07:49.31 | tzafrir | The separation of client/server appears to be a must. I wonder if it can be done with ajax alone at the client side. Aparantly not |
| 07:52.45 | appelza | hi, is this valid: |
| 07:52.48 | appelza | exten = 0!,1,Macro(trunkdial,Zap/g0,12/${EXTEN:0}) ? |
| 07:57.32 | litage|w | does the (new?) asterisk gui work with v1.2 , or only with v1.4 ? |
| 07:59.14 | *** join/#asterisk softice (n=test@196.7.60.220) |
| 07:59.20 | softice | hmm, small problem |
| 07:59.32 | [hC] | anyone seen a polycom 601 that refuses to turn the expansion module on for some reason? |
| 07:59.47 | softice | not sure whats up with my dtmf, I dial in, press for prompt in ivr, but I have to hit the number 4/5 times for it to recognise anything |
| 07:59.52 | [hC] | the screen on my expansion module isnt turning on when i boot up the phone |
| 08:00.08 | softice | it doesn't pick up any other dtmf, only after 4/5 tried does it pick up, 1 for eg |
| 08:04.15 | *** join/#asterisk SA007 (n=sa007@ip565e0006.direct-adsl.nl) |
| 08:04.41 | SA007 | weird, asterisk is filtering dtmf tones |
| 08:05.01 | SA007 | does anyone know how toturn that off? |
| 08:05.31 | watchy | [hc]: bad news g |
| 08:05.38 | watchy | i think its a bad phone |
| 08:06.09 | [hC] | watchy: ah. thats okay ill just use another one, i have a lot of them, i just wasnt sure why it would happen, never seen it do that before |
| 08:06.14 | watchy | i had the same issue with a 601. i never did have it fixed but i think its broke |
| 08:06.27 | [hC] | i have two sidecars that ive tried so i dont think its them |
| 08:06.33 | watchy | yea when i turned this 601 i got on and it didnt turn the modules on i was like WTF |
| 08:06.46 | watchy | it made me wonder if a config i had was wrong |
| 08:07.01 | [hC] | well yeah thats what i was thinking since i just upgraded the phone to 2.2.0 |
| 08:07.12 | [hC] | ill just try the same ones on another phone |
| 08:07.15 | watchy | whats the newest out? |
| 08:07.16 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:07.27 | [hC] | 2.2.0 as of a week ago |
| 08:07.30 | watchy | ah |
| 08:07.47 | watchy | any major improvements? |
| 08:08.42 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 08:09.07 | chris_1 | hi folks! |
| 08:09.51 | chris_1 | how can I make an attended transfer from an agent who observes a queue? |
| 08:10.17 | chris_1 | unfortunately only blind transfer is work |
| 08:14.05 | *** join/#asterisk salzh (i=salzh@218.80.157.19) |
| 08:14.25 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 08:15.21 | softice | and have you set your feature code? |
| 08:16.51 | appelza | has anyone here managed to dial out using a junghanns quadbri isdn card? |
| 08:18.54 | SA007 | this is weird stuff, i've got 2 phone's on a voip router, both connected to * (as 1234 and 1235 respectively) |
| 08:19.11 | chris_1 | yes - without the queue/agent part it works! |
| 08:19.21 | SA007 | if i dial the other one i get normal audio, execpt dtmf tones... |
| 08:20.59 | softice | hmm, does nobody have any idea where i can look for an issue with dtmf, where it only picks up a digit after hitting the number a few times? |
| 08:21.42 | *** join/#asterisk yassaccan (n=yassacca@admin171.hgo.se) |
| 08:28.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 08:28.16 | appelza | what does this mean? :P zt_handle_event: Detected alarm on channel 5: Red Alarm |
| 08:29.52 | watchy | something bad |
| 08:31.44 | softice | appelza: it means channel 5 line isn;t up |
| 08:34.27 | JT | it means there is an L1 failure |
| 08:34.29 | SA007 | damnit, i can see asterisk receiving dtmf, but it doesn't pass them trough to the other phone |
| 08:36.13 | *** join/#asterisk Strom_C (n=strom@216.64.24.250) |
| 08:38.11 | appelza | :( |
| 08:38.35 | appelza | I'm struggling to dial out over my isdn card, but I can receive calls from it |
| 08:38.40 | appelza | :/ |
| 08:38.50 | appelza | And I can dial out over my analog card |
| 08:38.53 | appelza | so i dunno :| |
| 08:40.05 | appelza | Can I paste my zapata-channels.conf somewhere and someone help me create the correct extention for the first port on my isdn card to dial out? Please |
| 08:40.38 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-179-242-169.vic.bigpond.net.au) |
| 08:41.33 | appelza | http://pastie.caboo.se/96022 |
| 08:42.05 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:46.40 | kaldemar | appelza: what does your dial line look like? |
| 08:48.21 | appelza | sec |
| 08:49.23 | appelza | exten = _0!,1,Macro(trunkdial,Zap/12/${EXTEN:0}) |
| 08:49.36 | appelza | coz I want to use Span2 which is 0-12 I think |
| 08:51.03 | appelza | but im not sure if im doing the right thing |
| 08:52.55 | softice | ok I have furthered with the problem |
| 08:53.13 | softice | it seems if Iuse background(message) it doesn't pick up the dtmf, sometimes works after 4/5 tries |
| 08:53.24 | softice | if I use playback, then after the message it picks it up every time |
| 08:56.00 | appelza | anyone :< |
| 08:58.14 | softice | anyone :( |
| 09:04.08 | kaldemar | you're using a macro, have you done it yourself? |
| 09:06.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:06.14 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 09:11.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:11.51 | *** join/#asterisk Polis_ttt (n=Polis_tt@194-237-172-225-no48.business.telia.com) |
| 09:12.38 | appelza | kaldemar, whats the proper way to do it? |
| 09:12.51 | appelza | I just want to use the first port on that isdn card to make a phone call, thats all |
| 09:12.52 | *** part/#asterisk dseeb_ (n=dcb@CPE-124-179-242-169.vic.bigpond.net.au) |
| 09:15.52 | appelza | how can I have zap dial these channels? channel => 14-15 ,Zap/14-15 doesnt work |
| 09:19.50 | kaldemar | you have to use groups. |
| 09:20.00 | kaldemar | are you using some GUI to configure asterisk? |
| 09:20.21 | appelza | no, cmd line |
| 09:20.33 | tzafrir | appelza, why not just use Zap/gNN (for group=NN in zapata.conf) |
| 09:20.33 | appelza | ok so Span-5 (which is the one where the line is plugged into) |
| 09:20.53 | appelza | has group: 0,15 |
| 09:20.58 | appelza | so Zap/g0,15 ? |
| 09:21.03 | *** join/#asterisk kkn088 (n=kikoun@84.7.164.107) |
| 09:21.18 | kaldemar | well, if you for example wanted to dial out using span5 you'd use Dial(Zap/g15/${EXTEN}) |
| 09:21.22 | tzafrir | appelza, in fact, if you configured it with genzaptelconf, all the TE spans are in group 0, and each span N is in group 10+N |
| 09:21.41 | softice | yay tzafrir is here |
| 09:21.42 | kaldemar | why have you defined your groups like that? |
| 09:21.46 | softice | i bet you could help me too? |
| 09:21.56 | kaldemar | oh, genzaptelconf. |
| 09:22.10 | tzafrir | softice, what is it about? |
| 09:22.54 | appelza | so group 0,15 is should be Zap/g15 ? |
| 09:22.58 | softice | dtmf issues, if I dial in, and use background(recording_file) I can hit adigit 4/6 times before it recognises 1 |
| 09:23.14 | softice | if I use playback for some reason afterthe playback it picks it up straight away |
| 09:23.20 | kaldemar | there is no such thing as group 0,15. those channels belong to groups 0 and 15. |
| 09:23.32 | softice | also if I hit a digit its not a clean dtmf sound if I hear on another handset? |
| 09:23.44 | tzafrir | kaldemar, group can get a list of groups |
| 09:23.46 | kaldemar | if you dial group 15, you use g15. |
| 09:24.11 | tzafrir | groups is actually a bitmask, telling to which groups the channel belogs |
| 09:24.19 | tzafrir | softice, what device is it? |
| 09:24.20 | kaldemar | tzafrir: yes, i said that because i wanted to be clear on a group being a single number. |
| 09:24.22 | appelza | ah ok |
| 09:24.25 | appelza | lemme try |
| 09:25.09 | softice | tzafrir a sangoma card. and what device are you dialing, i'm calling in from my mobile phone |
| 09:25.12 | softice | to the ivr |
| 09:25.28 | kaldemar | appelza: http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+channels <-- see dialing a group in that article for more info |
| 09:25.42 | appelza | Thanks, will do |
| 09:25.50 | tzafrir | softice, for starters, get asterisk to display the detected DTMF digits |
| 09:25.51 | softice | in the cli, if I hit 1/2/3/4/5 or what ever it should pick it up right away, sometimes I have tohit 1 (5 times) |
| 09:26.06 | softice | tzafrir: it does siplay them... |
| 09:26.19 | softice | but it doesn't display them if it isn't picking it up |
| 09:26.22 | tzafrir | enable "debug" and "dtmf" for the console in /etc/asterisk/logger.conf and set: |
| 09:26.24 | softice | then after the 5th try it displays it |
| 09:26.42 | tzafrir | ah, ok |
| 09:26.51 | *** join/#asterisk kkn088 (n=kkn088@84.7.164.107) |
| 09:27.00 | tzafrir | there are a number of things to try. |
| 09:27.15 | tzafrir | you can try recording the audio with ztmonitor |
| 09:27.20 | softice | funlly like I said, if I use playback, what you cant' type digits while its playing the sound file, after the sound file it picks up each dtmf tone |
| 09:27.24 | softice | not using background though |
| 09:27.30 | tzafrir | and then trying to get a "second opinion" for it |
| 09:27.59 | softice | tzafrir: the audio of the dtmf tones? |
| 09:27.59 | tzafrir | be that as simple as playing that DTMF stream from a handset and seeing if it gets detected then |
| 09:28.08 | tzafrir | softice, yes |
| 09:28.11 | softice | the tones are not coming through, clear.. they notclearn, its like a hiss between them |
| 09:28.19 | softice | or a crackle sound |
| 09:28.37 | softice | tzafrir: local it works fine |
| 09:28.43 | softice | it picks it up every time |
| 09:28.48 | softice | but if I come from the outside it isn't working |
| 09:28.59 | softice | I have echo cancelation set on the sangona card |
| 09:29.00 | tzafrir | if what you record in ztmonitor has the same problem, then maybe the issue is not with Asterisk |
| 09:29.33 | tzafrir | try decoding them with spandsp's dtmf test utility |
| 09:29.37 | softice | i'm coming into a asterisk box, on group 1 right, then passing out to another asterisk box ong roup 4 |
| 09:29.39 | softice | through isdn |
| 09:29.40 | tzafrir | see what it thinks |
| 09:30.22 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:30.36 | softice | ok, would echo training, etc cause problems from the outside? |
| 09:30.37 | *** join/#asterisk Strom_C (n=strom@216.64.24.250) |
| 09:30.40 | tzafrir | also, make sure you don't destroy it in zsaptel - extra gains, or maybe play with the echo canceller settings |
| 09:30.53 | *** join/#asterisk ManxPower (n=manxpowe@156.sub-75-203-66.myvzw.com) |
| 09:36.22 | softice | ok |
| 09:39.07 | appelza | ok, ive tried Zap/0 to Zap/20 ; Zap/g0 to Zap/20 and none of them work for placing a call (but I can recieve calls on that card, so it does have a line) |
| 09:39.14 | appelza | I'm so frustrated :( |
| 09:39.22 | *** join/#asterisk ManxPower (n=manxpowe@156.sub-75-203-66.myvzw.com) |
| 09:39.37 | appelza | hi ManxPower |
| 09:40.35 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
| 09:40.49 | RsaMan | what is the difference between sip show users and sip show peers ? |
| 09:40.55 | RsaMan | besides the obvious |
| 09:41.08 | RsaMan | whats the diff between a user and a peer i should sya |
| 09:41.16 | RsaMan | whats the diff between a user and a peer i should say |
| 09:43.46 | ManxPower | User -> Asterisk, Asterisk -> Peer |
| 09:44.07 | ManxPower | generally "sip show peers" is more useful, as it shows registration status of phones connected to Asterisk |
| 09:44.28 | ManxPower | As you can see a peer will never send calls to Asterisk. |
| 09:44.33 | appelza | could anyone please help me make a proper outbound extention for use on this isdn card: |
| 09:44.35 | appelza | http://pastie.caboo.se/96022 |
| 09:44.40 | ManxPower | And a User will never receive calls from Aserisk |
| 09:44.42 | appelza | I've tried everything I know of :< |
| 09:45.26 | RsaMan | thanks |
| 09:46.12 | ManxPower | Normally it would be Dial(Zap/g0/thenumber), but I don't think the BRI card uses Zap drivers. |
| 09:46.23 | RsaMan | using a pap2(linksys) as a sip client |
| 09:46.30 | RsaMan | it can receive calls |
| 09:46.34 | RsaMan | but cannot make calls |
| 09:46.51 | ManxPower | RsaMan: you know what the next step is, of course. |
| 09:47.04 | RsaMan | ManxPower : no idea ? |
| 09:47.07 | appelza | Ah! |
| 09:47.13 | ManxPower | WHAT IS THE ERROR ON THE ASTERISK CONSOLE |
| 09:47.26 | appelza | Because I've tried Zap/g0/thenumber with no luck. What would the bri equivilent be? |
| 09:47.31 | ManxPower | That is always the next step. |
| 09:47.37 | RsaMan | ah, |
| 09:47.43 | ManxPower | appelza: We don't use BRI where I live, so I have no idea. |
| 09:47.57 | RsaMan | i dont see any error, so must be a problem with my linksys unit |
| 09:48.18 | ManxPower | RsaMan: what happens when you try to dial? |
| 09:48.32 | ManxPower | i.e. silence, congestion tone, busy tone, etc? |
| 09:48.44 | RsaMan | ManxPower : dialtone, then nothing |
| 09:48.56 | RsaMan | ManxPower : dont say any attempts in the console though |
| 09:48.57 | ManxPower | put a # at the end of your dialed number |
| 09:49.09 | RsaMan | OH |
| 09:49.11 | RsaMan | nice |
| 09:49.21 | ManxPower | If that works then you just need to fix the dialplan on the SIPura/Linksys |
| 09:49.29 | RsaMan | it hung up |
| 09:49.31 | RsaMan | thanks |
| 09:49.37 | ManxPower | anything on the console now? |
| 09:49.44 | RsaMan | no |
| 09:49.59 | ManxPower | how many SIP devices do you have connected to Asterisk? |
| 09:50.35 | RsaMan | the pap2 and an spa400 |
| 09:50.43 | RsaMan | fxs and fxo |
| 09:50.58 | ManxPower | can you disconnect the SIP devices you are not using while you are testing? |
| 09:50.59 | appelza | maybe BRI can only handle incoming? (I've just confirmed that it does use Zap) |
| 09:51.09 | ManxPower | appelza: BRI can use both |
| 09:51.11 | appelza | all the samples and examples I see only show incoming aswel :/ |
| 09:51.13 | RsaMan | ManxPower : i can do that |
| 09:51.14 | ManxPower | incoming and outgoing |
| 09:51.21 | appelza | meh |
| 09:51.47 | appelza | my incoming works through the bri card, but i cant get outgoing working, no matter what :| |
| 09:51.55 | ManxPower | RsaMan: do that, then do a "sip debug" and watch the console for a 404 response when you try dialing. Heck, just put the SIP debug info on pastebin.ca |
| 09:52.09 | RsaMan | kk |
| 09:52.45 | ManxPower | appelza: if it's BRI chances are it is CAPI or MISDN |
| 09:52.53 | ManxPower | look for examples on the wiki |
| 09:54.30 | RsaMan | ManxPower : :( no output in sip debug when i dial using pap2 |
| 09:54.51 | RsaMan | ManxPower: but i am able to make calls to that channel |
| 09:55.05 | ManxPower | RsaMan: the two things are TOTALLY different. |
| 09:55.35 | ManxPower | It is common to be able to have calls in one direction, but not the other direction. |
| 09:55.58 | outtolunc | i doubt appelza has a valid exten in his 'from-pstn' context which is the only context used for the bri stuff |
| 09:56.18 | ManxPower | RsaMan: double check your pap2 settings. for server and user/pasword |
| 09:57.05 | RsaMan | ManxPower: if they where not correct , surely i would not be able to make calls to the device ? as it is registered as a sip client |
| 09:57.39 | ManxPower | RsaMan: that is an incorrect assumption |
| 09:58.42 | ManxPower | That like like asking that if you can enter a prison, you should be able to leave a prison. It doesn't work that way. |
| 10:01.29 | ManxPower | RsaMan: you MUST be seeing at least occasional SIP debug info, even if you are not dialing |
| 10:02.22 | RsaMan | http://pastebin.com/d12359a91 |
| 10:02.25 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 10:02.31 | RsaMan | i get this when i plug the pap2 in |
| 10:03.15 | ManxPower | RsaMan: good, so you are seeing the SIP debug. |
| 10:03.40 | ManxPower | SOMETHING must be wrong on the PAP2, because even if Asterisk rejects the call, you should still see something on the console when you try dialing from the PAP2 |
| 10:03.48 | JT | ManxPower: err being Zap on BRI would be NEITHER CAPI or mISDN |
| 10:03.49 | ManxPower | I assume this is really a PAP2NA |
| 10:04.16 | ManxPower | JT: As I understand it the Digium BRI card does not use Zap. Therefore it must use CAPI or MISDN |
| 10:04.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:04.54 | JT | appelza: is the BRI card digium? |
| 10:05.05 | JT | didn't think the digium card could do capi |
| 10:05.30 | RsaMan | ManxPower: Yeah i believe so , it should say something like the number does not exist in that context surely |
| 10:06.26 | ManxPower | The Digium card uses mISDN (I just did a google serarch) |
| 10:06.28 | ManxPower | http://www.asteriskguru.com/tutorials/digium_b410p_installation_guide.html |
| 10:08.50 | JT | ManxPower: yes, i knew that |
| 10:08.58 | JT | hence why i don't recommend purchasing it :) |
| 10:09.28 | ManxPower | JT: I can't recommend purchasing it either. |
| 10:09.34 | ManxPower | Digium cards should all support Zap. |
| 10:09.57 | ManxPower | Any Digium card that does not support Zap, in my not so humble opinion, is a piece of crap. |
| 10:10.22 | JT | must be the case that they couldn't have been bothered to make their own zap drivers |
| 10:10.31 | JT | as the existing ones are licence incompatible |
| 10:11.07 | ManxPower | JT: Seems to me like they are not comited to the product. |
| 10:11.38 | JT | quite possibly :) |
| 10:15.09 | ManxPower | That would like Microsoft releasing a product for Linux |
| 10:18.59 | juuva | ManxPower: MS Office (or IE) for OS X? |
| 10:20.21 | ManxPower | juuva: Microsoft already has those products for Windows. |
| 10:21.57 | *** part/#asterisk LukinoVoip (n=LukinoVo@gw.abanet.it) |
| 10:22.22 | juuva | well.. yes, actually I was supposed to ask about IAX2 trunks and queues in 1.4. Can I add remote users to queues (over iax2 trunk)? |
| 10:22.24 | ManxPower | Downloading message 14289 of 37294 |
| 10:22.26 | ManxPower | That sucks |
| 10:22.45 | ManxPower | I don't use queues or IAX2 |
| 10:23.29 | juuva | ok.. then I'll continue banging my head to something |
| 10:24.50 | ManxPower | based on my understanding of queues, there should be no problems with what you are trying to do. |
| 10:24.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:25.13 | ManxPower | I would not spend much time on the IAX2 part, that should work just like any other technology in Astersik |
| 10:25.43 | ManxPower | (assuming you have non-queue calls going in both directions over that link. |
| 10:28.15 | juuva | got to get coffee, after that, some testing |
| 10:28.38 | ManxPower | coffee is good |
| 10:30.02 | ManxPower | I just realized it is sep 11 today. |
| 10:30.26 | ManxPower | I'll have to make sure to leave the radio and television off |
| 10:33.43 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 10:42.57 | *** join/#asterisk Serees (n=me@118.213-66-87.adsl-static.isp.belgacom.be) |
| 10:44.35 | Serees | hi, i'm having a DAA failed to initialize error... can somebody help? Everything worked a few day's ago, but after a system reboot it stopped working :s |
| 10:45.55 | ManxPower | Serees: power off the system, then try it again |
| 10:48.22 | Serees | ManxPower that does not help... i've tried like a dozen times |
| 10:48.46 | ManxPower | Serees: that error is usually a hardware problem -- you should contact Digium support. |
| 10:48.47 | Serees | could it be an hardware failure? |
| 10:49.34 | *** join/#asterisk ghatak (n=ghatak@84-93-217-81.plus.net) |
| 10:49.34 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-17-56.red.bezeqint.net) |
| 10:49.43 | Dovid | anyone here use the ooh323 channel driver ? |
| 10:49.56 | Dovid | haha |
| 11:06.19 | ai-a | Serees: you tried removing the power, not just pressing the button.. so the device can lose all power. |
| 11:08.09 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 11:15.04 | Serees | ai-a what do you mean??? I allready fysiclly moved the pc... And changed the card from pci slot... So it was allready fully disconected |
| 11:16.59 | *** join/#asterisk michael-i (n=michael-@W9d63.w.pppool.de) |
| 11:18.42 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:22.37 | hi365 | i installed asterisk+zaptel+libpri from source, but asterisk doesnt have any zap/pri releated commands avalible. |
| 11:22.43 | hi365 | where did they go?? |
| 11:23.00 | JT | what order did you compile in? |
| 11:23.33 | hi365 | which time? 8) |
| 11:23.44 | hi365 | lat time lib-> zap>* |
| 11:23.44 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 11:23.44 | JT | most recently. |
| 11:24.17 | JT | is chan_zap compiled? |
| 11:24.23 | hi365 | but the sangomadrivers recompiled the zaptel (after everytjing else) |
| 11:24.40 | hi365 | its loadable in asterisk |
| 11:24.50 | hi365 | (it shows in the list |
| 11:24.53 | hi365 | ) |
| 11:25.08 | JT | is it loaded? |
| 11:25.39 | hi365 | how do i check? |
| 11:27.23 | hi365 | there are no zap related commands ... :( |
| 11:28.08 | *** join/#asterisk heartones (n=heartone@196.218.34.246) |
| 11:32.24 | *** join/#asterisk sashion (n=sdgsdg@41-195-131-15.access.uunet.co.za) |
| 11:34.30 | *** join/#asterisk kkn088 (n=kkn088@84.7.164.107) |
| 11:37.47 | hi365 | JT ^^ |
| 11:39.09 | hi365 | actualy- zap is loaded |
| 11:39.21 | hi365 | it just doesnt work :( |
| 11:40.25 | sashion | i keep getting segmentation faults with ast_senddigit_end()... and I cant pin-point the cause of it... |
| 11:41.48 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:41.57 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 11:42.45 | agx | Hello, STUN support is in trunk? |
| 11:43.58 | appelza | could anyone please help me create an outbound extention based on this info (the bri card, not the analog): http://pastie.caboo.se/96022 |
| 11:44.09 | appelza | id be very very greatful |
| 11:46.52 | sashion | appelza: are you wanting to create an extension on your BRI ports, or use them for dialling out on your CO ? |
| 11:46.57 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 11:50.53 | appelza | sashion, anything thatl allow me to place a call over that card |
| 11:51.01 | appelza | I can already recieve cards over it btw |
| 11:51.17 | appelza | via the isdn line |
| 11:53.08 | sashion | appelza: ok exten => _X.,1,Dial(ZAP/g12/${EXTEN}|60|tT) |
| 11:53.10 | *** join/#asterisk kkn088 (n=kkn088@84.7.164.107) |
| 11:53.31 | appelza | oooh, ill try |
| 11:55.34 | appelza | <PROTECTED> |
| 11:55.56 | appelza | btw, ive tried Zap/g1 to 20 earlier today..but didnt do the |60|tT |
| 11:56.08 | sashion | ok the 60 is just a timeout |
| 11:56.12 | sashion | and tT is for transfering |
| 11:56.32 | sashion | are you using a xorcom or trixbox ? |
| 11:56.50 | appelza | commandline |
| 11:57.24 | appelza | with the asterisk-gui, but its useless as it doesnt see my BRI card even though asterisk does..so now just commandline |
| 11:57.51 | sashion | hmm I see... |
| 11:57.53 | sashion | ok try this |
| 11:58.04 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:58.06 | appelza | :] |
| 11:58.11 | sashion | exten => _X.,1,Dial(ZAP/8/${EXTEN}|60|tT) |
| 11:58.16 | tzafrir | support for digital cards on the asterisk-gui is still very experimental. generally work-in-progress in asterisk-gui trunk |
| 11:59.47 | appelza | nope sashion :< |
| 12:00.02 | appelza | can I paste the full error somewhere I see in asterisk console? |
| 12:00.12 | sashion | ok wait |
| 12:00.16 | sashion | do a |
| 12:00.20 | sashion | pri show spans |
| 12:00.23 | sashion | pastebin that |
| 12:00.39 | appelza | http://pastie.caboo.se/96049 |
| 12:00.41 | appelza | oh sorry |
| 12:00.45 | appelza | thats the error |
| 12:00.47 | appelza | gimme a sec |
| 12:01.12 | appelza | http://pastie.caboo.se/96050 |
| 12:02.06 | *** join/#asterisk Shido6 (n=shido6@204.126.120.132) |
| 12:02.39 | kaldemar | hi365: you're not running a pri? |
| 12:03.03 | hi365 | i *think* i am (i have the card+drivers installed, but asterisk seesm to think otherwise) |
| 12:03.04 | *** join/#asterisk coppice (n=chatzill@109.206.17.210.dyn.pacific.net.hk) |
| 12:03.21 | hi365 | i dont have any zap related options either (zap show, etc.) |
| 12:03.53 | sashion | ok appelza, try exten => _X.,1,Dial(ZAP/15/${EXTEN}|60|tT) |
| 12:04.03 | appelza | ok |
| 12:04.58 | appelza | <PROTECTED> |
| 12:04.59 | appelza | <PROTECTED> |
| 12:04.59 | appelza | <PROTECTED> |
| 12:04.59 | appelza | <PROTECTED> |
| 12:05.12 | appelza | get that every time I try, but the line isnt busy :( |
| 12:05.28 | sashion | ok appelza, do pri intense debug span 5 |
| 12:05.28 | sashion | then repeat the attempt |
| 12:05.32 | sashion | and then pastebin it please |
| 12:05.49 | appelza | ok |
| 12:06.11 | tzafrir | appelza, "red alarm" - that is - disconnected or otherwise no layer 1 connectivity |
| 12:06.53 | appelza | dont have pri intense debug, have bri intense debug tho? |
| 12:06.55 | tzafrir | head -n 1 /proc/zap/SPAN_NUM |
| 12:07.05 | appelza | ioh wait I lie |
| 12:07.21 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
| 12:07.21 | tzafrir | you need bri [intense] debug |
| 12:07.27 | sashion | bri intense debug will surfice :P |
| 12:07.52 | appelza | here you go: http://pastie.caboo.se/96054 (with pri intense debug) |
| 12:08.22 | appelza | root@asterisk:/etc/asterisk# head -n 1 /proc/zaptel/5 |
| 12:08.23 | appelza | Span 5: ztqoz/1/4 "quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 0) Layer 1 ACTIVATED (F7)" AMI/CCS |
| 12:08.56 | appelza | zaptel/5 is span/4 :O |
| 12:08.58 | tzafrir | hmm... you do have RRs in both directions |
| 12:09.13 | tzafrir | so the span is active |
| 12:09.21 | sashion | um.. appelza: Cause: Invalid number format (28) |
| 12:09.21 | sashion | same here: |
| 12:09.21 | sashion | Executing [0763938619@numberplan-custom-1:1] Dial("SIP/6000-081fb9f8", "ZAP/15/|60|tT") in new stack |
| 12:09.36 | tzafrir | to reduce the spam and maintain sanity: bri no debug span 5 |
| 12:09.39 | sashion | you sure you passing the ${EXTEN} varaible.. cause that statement says you send nothing to your CO |
| 12:09.51 | appelza | lemme check |
| 12:10.17 | agx | Anyone coming to VON Europe in Italy ? http://www.von.com/2007/rome/web/index.php |
| 12:10.30 | appelza | I wasnt! |
| 12:10.32 | appelza | but lemme test |
| 12:11.19 | appelza | same error |
| 12:11.20 | HarryR | uh I think my colleague might be |
| 12:11.34 | HarryR | not sure about me though :\ |
| 12:11.44 | sashion | appelza: did you have intense debug on? If so, please pastebin it for me |
| 12:13.05 | hi365 | what commands do you need to compile libpri? |
| 12:13.18 | sashion | make, make install |
| 12:13.43 | appelza | hold on |
| 12:13.56 | hi365 | is this normal? |
| 12:13.58 | hi365 | make: Nothing to be done for `all'. |
| 12:15.30 | *** join/#asterisk melbert (n=chatzill@66.179.79.70) |
| 12:15.43 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
| 12:16.57 | appelza | http://pastie.caboo.se/96057 |
| 12:17.07 | appelza | (btw, thanks for all the help so far!) |
| 12:17.46 | ManxPower | hi365: not make all. make install |
| 12:18.08 | hi365 | ManxPower: i did make and thats the output that i got |
| 12:18.41 | hi365 | <PROTECTED> |
| 12:18.41 | hi365 | <PROTECTED> |
| 12:18.58 | hi365 | (thats the history) |
| 12:19.01 | ManxPower | appelza: pridialplan=unknown may be what you want |
| 12:19.08 | ManxPower | I'm waiting for the "make install" |
| 12:19.26 | hi365 | the output? |
| 12:19.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:19.32 | ManxPower | no the command |
| 12:19.44 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 12:19.54 | hi365 | i did it - what about it? |
| 12:19.57 | ManxPower | 426 should be make install |
| 12:19.59 | appelza | will try ManxPower |
| 12:20.05 | ManxPower | hi365: I don't see it in your history |
| 12:20.07 | appelza | that should go in zapata.conf under span4 right? |
| 12:20.16 | *** part/#asterisk agx (n=AGX@88.34.216.63) |
| 12:20.21 | hi365 | right. i wanted to confirm that the message wasnt an error befor i went on |
| 12:20.25 | ManxPower | appelza: it should go in /etc/asterisk/zapata.conf before any channel line |
| 12:20.55 | ManxPower | just like in the example config file provided with Asterisk |
| 12:20.59 | sashion | appelza: check your dialling format... cause the number gets rejected :P |
| 12:21.21 | appelza | ok |
| 12:21.45 | ManxPower | sashion: the number would be rejected if he had pridialplan=local and was trying to make a non-local call |
| 12:22.16 | hi365 | ManxPower: do i need to compile the zap modules for the digium cards if im using a sangoma card? |
| 12:22.43 | ManxPower | There is your problem: Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] 5 < [1e 02 82 88] |
| 12:22.55 | ManxPower | hi365: What does the sangoma docs say? |
| 12:23.11 | hi365 | im pretty sur they dont require it |
| 12:23.12 | ManxPower | In fact that information in right in the install document for sangoma |
| 12:24.39 | *** join/#asterisk HarryR (n=harryr@77.240.56.18) |
| 12:25.11 | pHnz | Hello, wich user may i used to System Configuration in the AsteriskNOW Web interface I used my user: admin and the admin passwd and that didn't work's. |
| 12:26.05 | ManxPower | pHnz: I suggest you try #AsteriskNOW because we don't use GUIs on this channel |
| 12:26.41 | ManxPower | hi365: so the wanpipeinstallation and readme.install files in the doc dir of the wanpipe source dir does not say? |
| 12:27.00 | ManxPower | and the Sangoma Wiki install files does not say either? |
| 12:27.43 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 12:30.46 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:30.50 | lirakis | morning |
| 12:31.10 | ManxPower | hi365: I know the answer. If you can't find the answer then you have far more serious issues. |
| 12:31.44 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 12:32.04 | *** join/#asterisk melbert (n=mmelbert@66.179.79.70) |
| 12:32.16 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 12:32.18 | *** part/#asterisk melbert (n=mmelbert@66.179.79.70) |
| 12:33.03 | hi365 | ManxPower: i dont think it says specificaly that you do or dont need the zap modules (the ones for the digium cards) |
| 12:33.18 | ManxPower | hi365: Then I guess you don't. |
| 12:34.54 | hi365 | so why doesnt astersk show any zap or pri options? |
| 12:35.06 | *** join/#asterisk melbert (n=IceChat7@66.179.79.70) |
| 12:35.38 | ManxPower | if you don't have zaptel installed, it won't show them |
| 12:35.56 | ManxPower | but that was NOT your question |
| 12:36.21 | *** part/#asterisk melbert (n=IceChat7@66.179.79.70) |
| 12:36.28 | appelza | what does this mean: |
| 12:36.30 | appelza | <PROTECTED> |
| 12:36.45 | ManxPower | appelza: it means the call did not go thru or was hungup |
| 12:36.57 | agx | sorry, where i can find digium developers? |
| 12:37.04 | ManxPower | agx: #asterisk-dev |
| 12:37.08 | agx | ty |
| 12:37.30 | agx | website is incorrect, it say to join #asterisk :-P |
| 12:37.34 | *** part/#asterisk agx (n=AGX@88.34.216.63) |
| 12:37.52 | ManxPower | hi365: there is also a README.asterisk in the Sangoma doc directory. |
| 12:38.00 | ManxPower | But you knew that already, right? |
| 12:39.28 | ManxPower | In fact the main page of the Sangoma Wiki has install instructions for Asterisk |
| 12:39.40 | ManxPower | hi365: did you read ANY Sangoma docs? |
| 12:40.00 | appelza | what would the trunkstyle be for an isdn card? digital? |
| 12:40.00 | hi365 | ManxPower: which i followed. and the sangoma stuff installed correctly (or so it seems) |
| 12:40.28 | hi365 | however, asterisk, although it loaded chan_zap doesnt have an zap options avalible |
| 12:41.12 | ManxPower | hi365: Did you follow the generic Sangoma install information or the specific install information that Sangoma provides for using their cards with Asterisk? |
| 12:41.15 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:41.49 | hi365 | i followed this: |
| 12:41.49 | hi365 | http://wiki.sangoma.com/wanpipe-linux-asterisk-install |
| 12:42.03 | ManxPower | I give up. Here is a quote from the Sangoma docs: "First install: |
| 12:42.04 | ManxPower | <PROTECTED> |
| 12:42.04 | ManxPower | <PROTECTED> |
| 12:42.14 | ManxPower | Note: |
| 12:42.14 | ManxPower | <PROTECTED> |
| 12:42.14 | ManxPower | <PROTECTED> |
| 12:42.16 | *** part/#asterisk Rawplayer (n=kevin@cp108757-a.landg1.lb.home.nl) |
| 12:42.17 | ManxPower | There! |
| 12:43.00 | ManxPower | It says right on that wiki page that you pasted the URL to. |
| 12:43.13 | hi365 | your point being? |
| 12:43.48 | ManxPower | my point being that if you follow those instructions and install in order zaptel, libpri, asterisk before installing wanpipe you would not be having the problem of no zap or pri support in asterisk |
| 12:44.30 | akirch_ | wait |
| 12:44.33 | hi365 | hmm, so the fact that i indeed folowed the instructions and never the less im stuck - does it say that anywhere? |
| 12:44.36 | akirch_ | you mean it works if you follow the instructions?!?!?! |
| 12:44.45 | akirch_ | hi365, call sangoma |
| 12:44.57 | ManxPower | I suspect you installed Asterisk before you installed zaptel or libpri and that is obviously not going to work |
| 12:44.59 | akirch_ | hi365, their support will walk you through... well anything you could possibly imagine needing |
| 12:45.20 | appelza | can anyone please help me with this error: http://pastie.caboo.se/96061 |
| 12:45.39 | ManxPower | they don't list the packages in random order, BTW. |
| 12:45.41 | akirch_ | hi365, CALL SANGOMA |
| 12:46.17 | akirch_ | By Phone: |
| 12:46.17 | akirch_ | Toll Free in North America: 1 800·388·2475 ext. 3 or |
| 12:46.17 | akirch_ | Internationally 1 905 474 1990 ext. 3 |
| 12:46.30 | akirch_ | their support is beyond phenomenal |
| 12:46.32 | ManxPower | hi365: and you will be installing IN ORDER: zaptel, libpri, asterisk, wanpipe and not in any other order |
| 12:46.35 | akirch_ | almost cisco like in it's completeness |
| 12:46.46 | ManxPower | akirch_: their support can't even fix a simple flex/lex/yacc problem |
| 12:46.54 | hi365 | sure |
| 12:46.57 | akirch_ | not in my experience |
| 12:47.16 | akirch_ | and anyway |
| 12:47.16 | hi365 | any need to uninstall befor i reinstall? |
| 12:47.23 | ManxPower | hi365: no |
| 12:47.32 | akirch_ | flex/lex/yacc sounds like a new bulemic weight loss plan |
| 12:47.38 | akirch_ | but that's just me |
| 12:47.57 | hi365 | thanks for the direct answer :) (didnt see it in the docs :) ) |
| 12:48.42 | ManxPower | hi365: that's because I don't think that info is in the docs 8-) |
| 12:48.52 | ManxPower | I only yell at people that ask questions that are in the docs. |
| 12:49.03 | file | why am I being threatened? |
| 12:49.11 | akirch_ | because they are muffins |
| 12:49.12 | ManxPower | akirch_: use english muffins, he's terrified of those. |
| 12:49.15 | akirch_ | give in to the threat! |
| 12:49.15 | Wonka | eep eep shneep eep eep |
| 12:49.30 | akirch_ | ManxPower, nah, file's ok |
| 12:50.16 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 12:50.19 | ManxPower | The last time file was confronted with an English Muffin he ran away screaming "Where's the fruit!!?" |
| 12:50.28 | akirch_ | ManxPower, I could see that |
| 12:50.46 | akirch_ | darned non-fruit-having English |
| 12:50.52 | *** join/#asterisk ming_zym (n=ming_zym@124.254.54.9) |
| 12:50.56 | akirch_ | though I did have a really tasty apple cinnamon English Muffin recently |
| 12:51.24 | akirch_ | (made to not taste like crap) |
| 12:52.05 | akirch_ | does that mean 1.2 is finally stable? |
| 12:52.07 | akirch_ | (and ducks) |
| 12:52.27 | file | it's perfectly stable for lots of people |
| 12:52.37 | akirch_ | I'm kidding |
| 12:52.39 | file | but every installation is not setup equal |
| 12:52.45 | akirch_ | some are more equal than others! |
| 12:55.18 | Nugget | 1.4 is doubleplusgood. |
| 12:55.48 | file | Nugget: all has been well? |
| 12:56.05 | Nugget | now that we're on the PRI, yeah |
| 12:56.23 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 13:01.34 | hi365 | No such command 'zap show' (type 'help' for help) |
| 13:01.51 | hmmhesays | type help |
| 13:02.00 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:02.16 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 13:02.50 | hi365 | ManxPower: correct me if im wron, but im pretty sure these are the steps in the correct order: http://pastebin.ca/691877 |
| 13:03.35 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 13:03.58 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 13:05.13 | [TK]D-Fender | hi365: Nope |
| 13:05.25 | appelza | I hate asterisk. :( |
| 13:05.50 | [TK]D-Fender | hi365: libpri, zaptel, wanpipe (zaptel should get compiled again automatically, it not do it manually), THEN asterisk |
| 13:05.57 | *** part/#asterisk dg (i=dgl@otherwize.co.uk) |
| 13:06.06 | [TK]D-Fender | appelza: Whats YOUR child-hood trauma? |
| 13:06.14 | appelza | asterisk :( |
| 13:06.15 | hi365 | lol |
| 13:06.23 | Wonka | lol |
| 13:06.35 | Wonka | is it allowed to mention callweaver here? |
| 13:07.09 | Corydon76-dig | You just did |
| 13:07.13 | file | sure |
| 13:07.16 | [TK]D-Fender | Wonka: Mention is fine, going zealotous would be ill-advised |
| 13:07.49 | appelza | Just when I think I understand asterisk, the next problem makes me cry :( |
| 13:07.52 | appelza | *sigh* |
| 13:07.55 | Wonka | hehe |
| 13:08.44 | Wonka | i made some asterisk-related stuff work here and am now tasked with looking into openser as a "sip firewall" in front of sip "hardware" |
| 13:08.54 | hi365 | [TK]D-Fender: do i need to redo the whole list frm the begining, or just recompile asterisk again? |
| 13:08.58 | Wonka | and privately, i want to look into callweaver |
| 13:09.12 | Wonka | how does callweaver handle chan_misdn and chan_capi? |
| 13:09.16 | hi365 | nevermind, ill just redo it |
| 13:09.57 | Corydon76-dig | Wonka: you're going to have to ask them |
| 13:09.59 | [TK]D-Fender | hi365: I'd advise the last 2-3 |
| 13:10.10 | Corydon76-dig | Wonka: they don't hang out here |
| 13:13.22 | hmmhesays | the bad guys on walker are so hopelessly retarded |
| 13:14.09 | Corydon76-dig | That's because if they were really smart, Walker would have been dead 5 minutes into the first season |
| 13:14.22 | hmmhesays | oh give him some credit |
| 13:14.37 | Corydon76-dig | 7 minutes? |
| 13:14.39 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.25.53) |
| 13:17.54 | Wonka | texas ranger? or which walker? |
| 13:18.49 | Corydon76-dig | bingo |
| 13:19.00 | hmmhesays | is there any other? |
| 13:19.59 | hi365 | so whats the [CC] and [LD] stuff mean? |
| 13:20.24 | Wonka | invocations of cc and ld |
| 13:20.28 | Wonka | in the build process |
| 13:21.54 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:23.47 | hmmhesays | good lord I hate the centos postgresql package |
| 13:24.07 | hmmhesays | it doesn't install a pg_hba.conf |
| 13:26.47 | hi365 | [TK]D-Fender: i actualy tryied compiling asterisk 3 times, still no reference to zap or pri :( http://pastebin.ca/691912 |
| 13:27.17 | hi365 | and chan_zap seems to be loaded |
| 13:27.31 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 13:27.50 | [TK]D-Fender | hi365: your * build is MANGLED. |
| 13:27.56 | tzanger | good morning from Frederickton! |
| 13:28.00 | tzanger | file: you around? |
| 13:28.06 | hi365 | how so? or better yet what to do? |
| 13:28.09 | [TK]D-Fender | cd asterisk-1.4.11 |
| 13:28.10 | [TK]D-Fender | make && make install |
| 13:28.26 | file | tzanger: yes |
| 13:28.32 | hi365 | i tired that befor |
| 13:28.40 | tzanger | where are you at these days? |
| 13:28.42 | file | tzanger: I'm not actually in New Brunswick though :D |
| 13:28.48 | file | I'm in Montreal till Saturday night |
| 13:28.56 | [TK]D-Fender | hi365: Yeah, you tried to ISNTALL if before eben doing ./configure , and make menuselect! |
| 13:29.00 | tzanger | dammit, I was just there yesterday |
| 13:29.14 | tzanger | I was going to see if I could visit [TK]D-Fender and junk-y but we didn't stay long |
| 13:29.15 | [TK]D-Fender | hi365: www.asterisk.org |
| 13:29.40 | [TK]D-Fender | tzanger: Shoulda called, we'd have grabbed that bear local! |
| 13:30.01 | tzanger | file: anything historical or interesting I should be visiting while out here? We're on our way to Halifax but anyway :-) |
| 13:30.02 | hi365 | [TK]D-Fender: a. ive done thouse befor b. ive tried compiling asterisk with a ./configre and menuselect |
| 13:30.06 | hi365 | history 335+336 |
| 13:30.24 | tzanger | [TK]D-Fender: perhaps on the way back? I have no idea wha my scheule's like, but yeah that would have been awesome |
| 13:31.25 | [TK]D-Fender | hi365: Funny I don't see you modprobing your car, starting wanrouter, I see no attempt to verify that zaptel is READY. |
| 13:31.48 | hi365 | hmm, good point! |
| 13:32.58 | [TK]D-Fender | And somebody at Digium deperately needs to rewrite : http://www.asterisk.org/support/install |
| 13:32.58 | [TK]D-Fender | thats jsut TRGIC |
| 13:33.08 | [TK]D-Fender | So bad its missing an "A"! |
| 13:33.22 | *** join/#asterisk Corydon76-dig (i=ten@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM] |
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| 13:33.23 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 13:33.23 | *** mode/#asterisk [+oo Corydon76-dig file] by irc.freenode.net |
| 13:33.24 | akirch_ | they're...BAAAAAAAAACK! |
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| 13:34.25 | hi365 | [TK]D-Fender: we were all there at one point... |
| 13:34.35 | hi365 | what am i modprobing for? |
| 13:34.47 | hi365 | zaptel (comes up blank)? |
| 13:35.00 | coppice | shouldn't it be exorcism for zombies? |
| 13:35.11 | [TK]D-Fender | hi365: zaptel. and you should already have started wanrouter and tested with ztcfg |
| 13:35.40 | hi365 | ztcfg comes up just fine |
| 13:35.41 | [TK]D-Fender | coppice: Zombies are soul-less, not possessed. |
| 13:35.56 | coppice | oh, support staff |
| 13:35.57 | hi365 | blame it on the dementors |
| 13:36.03 | [TK]D-Fender | hi365: Show some backup, then redo * in the right order |
| 13:36.17 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 13:36.24 | hi365 | again?! |
| 13:39.13 | *** join/#asterisk the_lalelu (n=lalelu@geek-at-work.org) |
| 13:39.18 | the_lalelu | ehlo |
| 13:39.19 | ManxPower | [TK]D-Fender: I think he's using 1.4 and the menuconfig did not detect zaptel / libpri on the first run and is not detecting it on future runs |
| 13:39.22 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:39.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:39.35 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 13:39.44 | tzafrir | agx, just ask |
| 13:39.48 | [TK]D-Fender | hi365: Sure, why not... trash your whole FOLDER and start over. |
| 13:40.08 | agx | anyone know why snom phones does not resend subscriptions until they are rebooted? |
| 13:40.08 | agx | someone has snom phones and know why they do not resend subscriptions ? |
| 13:40.27 | agx | tzafrir, uff, never use ViRC, its evil software :) |
| 13:40.29 | ManxPower | at least I assume he's running configure and make menuselect, he's not very good at finding and following the docs |
| 13:42.33 | [TK]D-Fender | ManxPower: Actually the instructions on asterisk.org are ancient and crap.... |
| 13:42.45 | [TK]D-Fender | ManxPower: maybe theres a better readme. |
| 13:43.14 | coppice | anyone tried one these new motherboards appearign with build it VoIP hardware? |
| 13:43.55 | agx | coppice: link? |
| 13:44.11 | coppice | Asus and MSI both have them |
| 13:44.48 | *** join/#asterisk mog (i=mog@nat/digium/x-0dc59284f58f8dbb) |
| 13:44.48 | *** mode/#asterisk [+o mog] by ChanServ |
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| 13:51.37 | tzafrir | agx, which specific snom phones? |
| 13:52.55 | agx | tzafrir, snom 300 fw:6.5.10, 320 fw:6.5.10 and .12 |
| 13:54.21 | hi365 | [TK]D-Fender: my god bless you! |
| 13:56.03 | [TK]D-Fender | hi365: You're welcome. |
| 13:56.14 | hi365 | [TK]D-Fender: really - thanks! |
| 13:57.37 | ai-a | agx: we use snom300 |
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| 14:03.07 | agx | ai-a, does it resend subscriptions? i just want to know into the webinterface if there is a flag to force it |
| 14:04.16 | ai-a | they are cheap crap. |
| 14:04.54 | coppice | cheap? |
| 14:06.57 | appelza | "< |
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| 14:11.54 | *** join/#asterisk zippytech (n=ron@71.155.129.244) |
| 14:11.58 | appelza | does DID_ before a context have any special purpose? |
| 14:12.08 | zippytech | any one know how to increase the voice mail emial messsage |
| 14:12.18 | zippytech | volume |
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| 14:13.03 | ai-a | appelza: its just a name. |
| 14:14.28 | [TK]D-Fender | zippytech: read the sample voicemail.conf. |
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| 14:15.32 | zippytech | cool thanks |
| 14:21.46 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 14:21.50 | arcanine | hello |
| 14:22.33 | zippytech | i don't see any thing in the voicemail.conf to control volume, any idea where to look |
| 14:23.02 | appelza | thanks |
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| 14:23.42 | [TK]D-Fender | zippytech: its there, look AGAIN |
| 14:24.37 | arcanine | can i assign a local nos. that he can only do outbound calls on only specific set of prefix |
| 14:24.55 | ai-a | zippytech: cat voicemail.conf.sample | grep volume |
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| 14:25.04 | Qwell | `uuoc |
| 14:25.07 | Qwell | ~uuoc |
| 14:25.08 | jbot | extra, extra, read all about it, uuoc is Useless Use of Cat Award. Given out for years by Randal Schwartz on the newsgroup comp.unix.shell. Basically, most constructions that look like "cat filename | grep pattern" can be more easily written as "grep pattern filename". Works for grep and most other Unix utilities. Easier to type and marginally more efficient. |
| 14:25.16 | deegan | I'm looking for something like queuemetrics and the like for a callcenter that places outgoing calls, any opensource alternatives (or free if you like the term better). |
| 14:25.20 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:25.57 | ai-a | Qwell: ;) |
| 14:26.32 | ai-a | actually, i dont like doing pattern filename, because to change the grep pattern i have to go back more words.. pain.. i prefer the cat file | grep patterrn, then i can modify the patten easy. |
| 14:27.26 | ai-a | locate -i voicemail.conf.sample | xargs grep volume |
| 14:27.32 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 14:27.40 | agx | any good TAPI driver around as alternative to xtelsio.com? |
| 14:29.54 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 14:29.58 | arcanine | ex: local 1201-1211 can only use 21+ prefix.... |
| 14:30.19 | arcanine | only this prefix can he use |
| 14:31.25 | [TK]D-Fender | arcanine: yes, that is the entire POINT for dialplan patterns. You choose what a given device/channel is allowed to dial and what it does. |
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| 14:35.48 | awannabe | hello, is it possible with ParkAndAnnounce() that you can park the call, and instead of having it ring back to the person who parked it, you can just announce the park slot on the same call, just like with DTMF parking |
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| 14:44.49 | Aeudian | Anyone have an experience with a logging application for Asterisk? I am looking for a way to log each phone call, sorta like a call center operation. I am looking for if the call was inbound/outbound, duration, and termination. |
| 14:47.23 | [TK]D-Fender | Aeudian: CDR <---- |
| 14:48.07 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:50.19 | Aeudian | perfect, thanks |
| 14:51.44 | Aeudian | [TK]D-Fender: ever used the Asterisk-Stat: CDR Analyser? I need a front end to this so i can show the clients |
| 14:51.52 | *** join/#asterisk UVSoft (n=jnk467@motorola154-31.ip.PeterStar.net) |
| 14:52.03 | [TK]D-Fender | Aeudian: Nope. Go TRY them and see |
| 14:52.05 | hi365 | on asterisk 1.4.11 i cannot spy on any channels that are allready in a call using the following options: exten => s-spy,1,chanspy(SIP|bw) |
| 14:52.35 | [TK]D-Fender | hi365: and why not? |
| 14:52.46 | hi365 | thats my question |
| 14:53.03 | appelza | ^_______^ |
| 14:53.06 | appelza | kbai |
| 14:53.25 | [TK]D-Fender | hi365: that isn't a question, its a STATEMENT |
| 14:53.52 | hi365 | [TK]D-Fender: why not? <---- now thats a question! |
| 14:54.16 | [TK]D-Fender | hi365: How about you show some CLI output and describe what HAPPENS. |
| 14:55.31 | hi365 | [TK]D-Fender: it doesnt recognize that there is a call in progress to spy on |
| 14:55.31 | hi365 | http://pastebin.ca/692056 |
| 14:56.15 | [TK]D-Fender | hi365: I should see a channel dump in there... |
| 14:56.30 | hi365 | [TK]D-Fender: if i dont specify the SIP part, the it works |
| 14:56.31 | hi365 | hold |
| 14:56.42 | *** part/#asterisk agx (n=AGX@88.34.216.63) |
| 14:56.45 | *** part/#asterisk UVSoft (n=jnk467@motorola154-31.ip.PeterStar.net) |
| 14:57.07 | hi365 | asterisk*CLI> core show channels |
| 14:57.07 | hi365 | Channel Location State Application(Data) |
| 14:57.07 | hi365 | Zap/1-1 (None) Up Bridged Call(SIP/229-092461a8) |
| 14:57.07 | hi365 | SIP/229-092461a8 s@macro-dialout-trun Up Dial(ZAP/g0/6400000|300|Tt) |
| 14:57.07 | hi365 | 2 active channels |
| 14:57.08 | hi365 | 1 active call |
| 14:58.28 | ManxPower | hi365: you don't specify a sip port when using spy |
| 14:59.32 | *** join/#asterisk lbow (n=lbow@41-195-77-184.access.uunet.co.za) |
| 15:00.12 | hi365 | ManxPower: 'chanprefix' only requres the prefix of the channel |
| 15:01.48 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.31) |
| 15:04.55 | hi365 | anyone? |
| 15:05.11 | *** join/#asterisk melbert (n=IceChat7@66.179.79.70) |
| 15:05.47 | ai-a | any way to park a call on a holding line.. ie 1->N so someone else can pick up that call when they are free ? |
| 15:06.27 | *** join/#asterisk Skaag (n=skaag@194.90.216.102) |
| 15:06.29 | melbert | I have verizon PRI's right now and have had some reliability issues with them. I am considering and Verizon is trying to sell me on their VOIP service. Has anyone had any experience with Verizon VOIP vs. PRIs? |
| 15:07.16 | Qwell | melbert: Their PRI service is supposed to be very high reliability. If they can't even get close to that on copper, how are they going to do so over the internet? |
| 15:07.29 | melbert | ???? |
| 15:08.04 | melbert | Qwell We have had 2 outages over 6 apiece in 8 months |
| 15:08.10 | melbert | 6 hours that is |
| 15:08.33 | Qwell | so, what, your internet connection is going to go over DSL, on a less reliable pair of copper, then over the public internet... |
| 15:08.45 | Qwell | I don't see how that could be more reliable at all |
| 15:09.05 | melbert | It would a dedicated T1 from them |
| 15:09.19 | *** join/#asterisk [intra]lanman (n=lanman@va-76-6-213-8.dhcp.embarqhsd.net) |
| 15:09.35 | Qwell | your PRI is already a "dedicated T1"... |
| 15:10.40 | melbert | So maybe we just want a second provider to switch our number over to in case of emergency |
| 15:10.42 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 15:10.50 | Qwell | or get a better PRI provider |
| 15:11.27 | melbert | unfortunately I inherited a 3 year contract with them |
| 15:11.36 | Qwell | break it. they did |
| 15:12.05 | Qwell | If they can't give you the reliability that they guaranteed, the contract is void. |
| 15:12.18 | hi365 | using the following, i cannot spy on any calls that are allready in progress. any idea why? |
| 15:12.19 | hi365 | exten => s-spy,1,chanspy(SIP|bw) |
| 15:13.06 | melbert | I mentioned that to the president of our company...but he is concerned about taking a "chance" with another provider. I told him we were taking a chance stay with out current provider |
| 15:13.16 | Qwell | option 2: Quit. :) |
| 15:13.19 | melbert | ha |
| 15:13.33 | Qwell | you'd be better off |
| 15:13.37 | cellphone | break down the rate of failures and how much it costs your company versus how much it'll cost to switch. |
| 15:14.00 | Qwell | cellphone: should he also include the cost of him doing the breakdown? :p |
| 15:14.05 | cellphone | haha. |
| 15:14.24 | cellphone | if you can't appeal to logic, appeal to emotion. |
| 15:15.00 | [TK]D-Fender | Qwell: Don't forget to break down this outside consultation time you're spending while you're at it... or my time AUDITING the time you're spending for that matter :p |
| 15:15.01 | Qwell | cellphone: You must be in sales. |
| 15:15.07 | melbert | yeah...I need to get quotes from other providers |
| 15:15.09 | Qwell | [TK]D-Fender: precisely |
| 15:15.20 | cellphone | no, I just have a boss that doesn't always act on logic. |
| 15:15.48 | cellphone | and I know how to get him to move on issues :) |
| 15:16.27 | *** join/#asterisk Phuntom (n=Phuntom@80.233.159.254) |
| 15:16.34 | Phuntom | hi ya! |
| 15:16.49 | rickross | hi guys. Does anyone know the feature code to record a call? |
| 15:16.56 | rickross | I cant seem to find it listed anywhere |
| 15:18.15 | [TK]D-Fender | rickross: features.conf <--- |
| 15:18.27 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 15:18.35 | [TK]D-Fender | rickross: And learn to use the WIKI for this stuff too.... |
| 15:18.43 | rickross | thx |
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| 15:34.32 | mohsen | asteriskdocs.org is down? I have not been able to connect to it for several days. |
| 15:35.03 | _x86_ | voip-info.org is better anyway ;) |
| 15:35.59 | [TK]D-Fender | ~book |
| 15:35.59 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 15:36.26 | mohsen | maybe. but it can not fix lots of broken links around |
| 15:37.02 | mohsen | and does not have a copy of that book, i guess :| |
| 15:37.27 | Phuntom | why is voip-info.org better then asteriscdosc? |
| 15:37.32 | hmmhesays | haha this is the episode where walker kicks through a windshield |
| 15:37.46 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:38.33 | [TK]D-Fender | mohsen: I've mirrored the book and added it to the end of that reference |
| 15:39.58 | mohsen | [TK]D-Fender: that's great. actually I was just answering _x86_. I could find the book on asterisk-france.com. Though not sure how out-dated my version is. |
| 15:40.45 | awannabe | hey guys, anyone used ParkAndAnnounce quite a bit? |
| 15:42.45 | ai-a | does asterisk support Overlap dialing ? |
| 15:42.46 | *** part/#asterisk md1024 (n=martin@host86-143-58-94.range86-143.btcentralplus.com) |
| 15:44.35 | Qwell | Phuntom: it isn't |
| 15:46.16 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:46.51 | [TK]D-Fender | awannabe: um... trying to follow what it is you want to do... |
| 15:50.15 | awannabe | [TK]D-Fender, well i want to transfer the call/start the parking, at that point instead of having it call the phone back i would like it to just announce the parking space right htne |
| 15:50.43 | [TK]D-Fender | awannabe: Why not just park the call like NORMAL then? |
| 15:50.47 | awannabe | instead of having it end the call, then call you back, a extra step id hate to have to do |
| 15:51.05 | awannabe | [TK]D-Fender: like normal? using DTMF digits? we are having GOBS of problems with that |
| 15:51.19 | [TK]D-Fender | awannabe: what do you mean with DIGITS? |
| 15:51.35 | [TK]D-Fender | awannabe: how are you planning on sending this call to be parked in the first place? |
| 15:51.41 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:51.52 | awannabe | transfer the call to a extension, which then start ParkAndAnnounce |
| 15:52.06 | *** join/#asterisk Corydon76-dig (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 15:52.06 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 15:52.21 | Trevor_B|Away | with a transfer button or a * code? |
| 15:52.22 | [TK]D-Fender | awannabe: how is that different than transferring to 700 for normal parking? |
| 15:52.23 | awannabe | we have been using call parking via DTMF, as in you hit #701 and it starts the "transfer" but we are having serious issues with that |
| 15:53.04 | awannabe | normal parking is from features config, well the DTMF parking we have been using |
| 15:53.09 | [TK]D-Fender | awannabe: the point of parkandannounce is so you can do things like setting up an automated PAGING of where it was parked so nervous shmucks can park without having to have a nervous breakdown on a loudspeaker |
| 15:53.22 | awannabe | lol, gotcha |
| 15:53.25 | [TK]D-Fender | awannabe: Who said anythign about having to use DTMF for this? |
| 15:53.45 | [TK]D-Fender | awannabe: you just transfer to 700. How you want to go about doing that is irrelevant |
| 15:53.51 | awannabe | i guess all the docs i read have said that is why, we just need to park a call, have it announce the parking space on the same call, and thats it |
| 15:54.12 | [TK]D-Fender | awannabe: then you misunderstood its purpose |
| 15:54.24 | [TK]D-Fender | awannabe: just to straight parking. |
| 15:54.39 | awannabe | see right now we are having issues with you park a call, you get a number, then you go to pick that call up, and it says "2" |
| 15:54.40 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:56.40 | [TK]D-Fender | awannabe: huh? |
| 16:00.25 | awannabe | yeah, our problem right now is, you park a call, it says slot "1", then you go to pick up slot 1, and it says "2" |
| 16:00.37 | awannabe | its soooo weird, and i have no ideal whats doing it |
| 16:01.53 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 16:01.58 | [TK]D-Fender | awannabe: you'd have to pastebin your config & CLI and maybe we'll see |
| 16:02.28 | *** part/#asterisk mohsen (n=chatzill@81.31.160.140) |
| 16:02.35 | *** join/#asterisk ZackTek (n=zzumbaug@70.244.109.129) |
| 16:02.41 | ZackTek | hi |
| 16:02.49 | nny | so whats the legal side of the http interface atserisknow uses? I don't need it per se, but it would be a nice addition to something built for a client. Is the interface GPLd? |
| 16:03.11 | awannabe | and its random, very bizarre. so thats why i was trying to get away from it complety. its 1.2.13 build as well, and that might be it |
| 16:04.32 | awannabe | [TK]D-Fender: so your saying just use normal park that is built in, and what is the way you recommend to do a transfer so the person hears what space it is parked on? |
| 16:05.00 | ZackTek | anyone have experience fixing a "We think we're the CPE, but they think they're the CPE too" issue? |
| 16:05.21 | ZackTek | with a TE120P and a 10 channel PRI |
| 16:05.48 | awannabe | ZackTek: you need to change the type, what is it set to right now? |
| 16:05.55 | alrs | ZackTek: It sounds like you might be just getting loopback |
| 16:07.06 | ZackTek | span = 1,1,0,esf,b8zs |
| 16:08.01 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:08.36 | dlynes | I guess the BLINDTRANSFER variable only works in those cases, where you're doing an analogue blind transfer, not a sip transfer, using the transfer key on the phone? |
| 16:09.33 | *** join/#asterisk Y0da^ (n=Home@70.159.118.70) |
| 16:10.08 | *** part/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 16:10.47 | ZackTek | awannabe: what do you mean by type? |
| 16:11.46 | ZackTek | switchtype is set to 5ess which is what the phone people told me |
| 16:11.52 | ZackTek | signalling is set to pri_cpe |
| 16:12.27 | MrMister2 | Hi. I'm having a problem with a vanilla instalation of Asterisk. * picks up the call, reports that it's playing a message and hangups. That's what it should do, problem is I get no sound :( This is from a mobile phone to a sip trunk. http://pastebin.ca/692164 |
| 16:12.41 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 16:13.02 | MrMister2 | Anyone had this issue before? |
| 16:13.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:13.29 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:13.33 | MrMister2 | I _think_ I have everything setup but since I'm a newbie I may be doing something dumb.. |
| 16:13.52 | *** join/#asterisk heartones (n=heartone@196.218.34.246) |
| 16:16.49 | dlynes | Does ParkAndAnnounce not work as documented? |
| 16:17.22 | dlynes | I'm trying to get it to timeout back to a certain extension at a certain priority in a certain context, and yet it still insists on timing back out to the extension that parked the call |
| 16:17.27 | *** part/#asterisk lukketto (n=lukketto@host80-193-dynamic.7-87-r.retail.telecomitalia.it) |
| 16:18.06 | dlynes | My line is as follows: exten => *74,2,ParkAndAnnounce(pbx-transfer:PARKED|30|Local/dead|ring_all|${CALLERID(num)}|1) |
| 16:18.37 | alrs | ZackTek: Sometimes the dorks at the carriers have no idea what they are talking about. Try switchtype=national |
| 16:18.41 | alrs | for sport |
| 16:18.42 | dlynes | I don't care about announcing the parkedcall extension, because it should ring back in 30 seconds to every extension |
| 16:18.56 | ZackTek | ive tried that as well |
| 16:19.11 | alrs | ZackTek: What are you using for a cable? |
| 16:19.19 | ZackTek | straight ethernet |
| 16:19.27 | ZackTek | i tried a crossover but then the light on the card turns red |
| 16:19.45 | alrs | ZackTek: and the card goes red if you unplug it at the smartjack? |
| 16:19.49 | ZackTek | yes |
| 16:20.09 | alrs | ZackTek How many spans are on that T1 card? |
| 16:20.46 | dlynes | MrMister2: can you repastebin as a non-polluted log, with verbosity set to 100? |
| 16:20.55 | dlynes | MrMister2: i.e. no sip debug? |
| 16:21.02 | MrMister2 | dlynes: OK. Just a sec |
| 16:21.41 | ZackTek | one Span |
| 16:21.42 | dlynes | MrMister2: also, is anything behind a firewall? |
| 16:21.47 | ZackTek | 9 bchannels and 1 dchannel |
| 16:21.57 | ZackTek | the rest of the channels are for dat |
| 16:21.58 | ZackTek | a |
| 16:22.27 | ZackTek | they said 24 was the dchannel and 1-9 were the bchannels |
| 16:23.07 | *** join/#asterisk vykarian (i=risadinh@200.138.30.10) |
| 16:23.14 | dlynes | ZackTek: this is a new install? |
| 16:24.56 | ZackTek | yes |
| 16:25.19 | dlynes | ZackTek: can you pastebin zap show status and zap show channels? |
| 16:25.33 | ZackTek | yes |
| 16:25.38 | ZackTek | give me a sec |
| 16:26.08 | dlynes | ZackTek: also, is it a digium card, sangoma card, rhino card, ...? |
| 16:26.11 | MrMister2 | dlynes: http://pastebin.ca/692175 |
| 16:26.12 | ZackTek | i took down the server to double check the jumper on the te120p :-P |
| 16:26.16 | ZackTek | Digium TE120P |
| 16:26.38 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@032-390-340.area5.spcsdns.net) |
| 16:27.01 | MrMister2 | dlynes: I dont think so, how can I check on linux? I'm a newbie on * and Linux :) |
| 16:27.54 | dlynes | MrMister2: as root user: iptables -nL |
| 16:28.14 | dlynes | MrMister2: if you get more than about 12 lines or so, you've got a firewall |
| 16:29.00 | dlynes | MrMister2: or just pastebin the result of iptables -nL |
| 16:29.32 | MrMister2 | dlynes: http://pastebin.ca/692181 |
| 16:30.06 | [intra]lanman | is there any way to get * to route a NOTIFY from another source to a user registered locally? |
| 16:30.13 | MrMister2 | Any configuration there has been setup automatically by the instalation of Plesk |
| 16:30.53 | dlynes | MrMister2: I would fire plesk...with a setup like that, you may as well not even have a firewall |
| 16:31.03 | MrMister2 | dlynes: :) LOL |
| 16:31.19 | dlynes | MrMister2: your windows networking is completely exposed to the Internet |
| 16:31.26 | [TK]D-Fender | MrMister2: I don't see you allowing SIP or RTP in that firewall |
| 16:31.32 | *** join/#asterisk f00bar80 (n=mina@196.202.91.215) |
| 16:31.47 | MrMister2 | dlynes: windows networking? You lost me there. |
| 16:31.52 | dlynes | [TK]D-Fender: that was going to be my next point :) |
| 16:32.04 | f00bar80 | i want to setup my VOIP gateway as to be a VOIP long distance calls service provider , i want to know the software/hardware i'll need for both Client/Server sides , and any further accounts registration needed like SIP account or something else |
| 16:32.07 | MrMister2 | [TK]D-Fender: What do I need to run to allow it? |
| 16:32.08 | dlynes | MrMister2: windows networking/samba file services/... |
| 16:32.25 | [TK]D-Fender | MrMister2: 5060,10000-20000 UDP |
| 16:32.30 | MrMister2 | also, If no SIP allowed should I still be able to receive the call? |
| 16:33.19 | dlynes | [TK]D-Fender: btw...he's got ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 |
| 16:33.27 | dlynes | [TK]D-Fender: just to make it extra secure |
| 16:33.45 | dlynes | [TK]D-Fender: so the policy of drop all is null and void |
| 16:34.05 | generalhan | anyone here use an IPCop box between their * box and the world ? |
| 16:34.25 | [TK]D-Fender | I also see in your log output that is hangups 1 SECOND after answering. maybe you should LENGTHEN your test call for SANITY reasons... |
| 16:35.11 | dlynes | [TK]D-Fender: btw...you have any experience with the ParkAndAnnounce() application? |
| 16:35.25 | generalhan | im trying to setup some QoS stuff and i cant seem to find good documentation on how to get it done. And when ever im doing a large download, or there are a lot of people on at the same time, or remote users' connection drops to 600ms+ |
| 16:35.31 | ZackTek | dlynes: here is my zap show status and zap show channels ---> http://pastebin.com/d6cde209d |
| 16:35.40 | [TK]D-Fender | dlynes: Not really, but feel free to ask anyways |
| 16:35.59 | generalhan | s/ or / our / |
| 16:36.00 | MrMister2 | [TK]D-Fender: K. let me put a longer message there. But disregarding the non-existance of security on the firewall, there should be nothing there to stop the sound, correct? |
| 16:36.16 | f00bar80 | ppl, any comment ? |
| 16:36.21 | dlynes | [TK]D-Fender: Yeah...it keeps timing back out to the extension that parked for me, rather than the context,extension,priority that I specify |
| 16:36.27 | generalhan | uhh ... thats what i said the first time ! lol |
| 16:36.31 | [TK]D-Fender | MrMister2: if you believe that allow-all takes precedence |
| 16:37.22 | dlynes | ZackTek: have you been getting any warnings or errors on your console or in your log from asterisk for the pri card? |
| 16:37.50 | dlynes | ZackTek: also, anything peculiar in your dmesg about that card? |
| 16:37.54 | [TK]D-Fender | dlynes: who said anything about it allowing you to specify an exten & prio? |
| 16:38.31 | ZackTek | dlynes: i just get the We think we're the CPE, but they think they're the CPE too |
| 16:38.33 | dlynes | [TK]D-Fender: return_context: the goto style label to jump the call back into after timeout. default=prio+1 |
| 16:38.34 | ZackTek | dmesg looks ok |
| 16:39.18 | dlynes | ZackTek: sounds like they've got their end misconfigured |
| 16:39.26 | [TK]D-Fender | dlynes: it still only says CONTEXT, otherwise it continues on the CURRENT exten in the current context. It does not say you can provide a NEW exten to land on. |
| 16:39.27 | dlynes | ZackTek: get them to doublecheck their end |
| 16:39.44 | [TK]D-Fender | ZackTek: pastebin your zapata.conf & zaptel.conf |
| 16:39.57 | [TK]D-Fender | ZackTek: And dmesg while you're at it |
| 16:40.48 | dlynes | [TK]D-Fender: the name of the field is return_context, but the description is a 'the goto style label' |
| 16:41.10 | dlynes | [TK]D-Fender: and all the examples i've found all show a goto style label, not a context |
| 16:41.25 | [TK]D-Fender | dlynes: pastebin an attempt |
| 16:42.15 | dlynes | [TK]D-Fender: nvm...I see what the stupid thing is doing |
| 16:42.30 | dlynes | [TK]D-Fender: it's ignoring the extension i'm giving it, and only taking the context and priority |
| 16:42.49 | ZackTek | my dmesg: http://pastebin.com/d31ea566a |
| 16:43.09 | dlynes | [TK]D-Fender: erm...nvm...that's cause I told it to :) |
| 16:43.31 | ZackTek | zaptel.conf: http://pastebin.com/d2c5e1e96 |
| 16:43.49 | [TK]D-Fender | ZackTek: -- DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM400P?? <--------- |
| 16:44.12 | ZackTek | zapata.conf: http://pastebin.com/d7ddd1208 |
| 16:44.23 | ZackTek | ya it's not plugge din but im not using it anyway |
| 16:45.34 | [TK]D-Fender | ZackTek: And now : ztcfg -vvvv |
| 16:46.52 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:47.27 | ZackTek | ztcfg: http://pastebin.com/d3c715917 |
| 16:48.01 | hmmhesays | godaddy is so nice |
| 16:48.06 | Qwell | godaddy sucks |
| 16:48.14 | hmmhesays | haha they are friendly and cheap |
| 16:48.20 | Qwell | ~cheap |
| 16:48.20 | jbot | methinks cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 16:48.23 | f00bar80 | i want to setup my VOIP gateway as to be a VOIP long distance calls service provider , i want to know the software/hardware i'll need for both Client/Server sides , and any further accounts registration needed like SIP account or something else |
| 16:48.28 | Qwell | Vonage used to be friendly too |
| 16:48.31 | hmmhesays | I take it you've had problems with them |
| 16:48.35 | Qwell | nop |
| 16:48.36 | Qwell | e |
| 16:48.46 | Qwell | I've just seen what they've done |
| 16:48.53 | [TK]D-Fender | ZackTek: Ok, and whats the error you're getting now? |
| 16:48.56 | hmmhesays | and what is it they have done that you don't like? |
| 16:49.14 | ZackTek | WARNING[3009]: chan_zap.c:9151 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. |
| 16:49.21 | Qwell | hmmhesays: google them :) |
| 16:49.42 | [TK]D-Fender | ZackTek: What exactly are you plugging your system into? |
| 16:49.59 | ZackTek | DSX-1 port on an ADTRAN 608 |
| 16:50.10 | dlynes | [TK]D-Fender: my mistake...that had nothing to do with why it wasn't working...it's still not working...the pastebin is: http://pastebin.ca/692215 |
| 16:50.30 | [TK]D-Fender | ZackTek: maybe they ARE backwads. Just change your zapata signalling and see if it stop whining. |
| 16:50.39 | ZackTek | ya ive tried that |
| 16:50.42 | hmmhesays | every company makes mistakes |
| 16:50.58 | Qwell | repeatedly? |
| 16:51.03 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 16:51.48 | ZackTek | FIXED |
| 16:51.52 | ZackTek | it works now |
| 16:52.07 | ZackTek | there was a dang loopback on the DSX-1 port that the phone company "has no idea how it got there" |
| 16:52.14 | Qwell | ZackTek: nice |
| 16:52.34 | hmmhesays | all of the complaints are see are related to seclists.org |
| 16:52.37 | ZackTek | thanks a lot guys |
| 16:52.51 | *** part/#asterisk ZackTek (n=zzumbaug@70.244.109.129) |
| 16:52.56 | *** join/#asterisk drutlandxpt (n=drutland@c-75-65-167-27.hsd1.ms.comcast.net) |
| 16:52.57 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-ab52ce91e57ab923) |
| 16:53.39 | [TK]D-Fender | dlynes: Why exactly are you doing an announce with no announce? |
| 16:53.55 | f00bar80 | am i talking to myself ? |
| 16:53.56 | *** join/#asterisk gubluntu (i=46130682@gateway/web/cgi-irc/ircatwork.com/x-ccf26030cb1e7647) |
| 16:53.56 | drutlandxpt | I am having issues with dtmf. can anyone help? |
| 16:54.20 | [TK]D-Fender | f00bar80: Your wustion is just so wonderfully vague and huge. |
| 16:54.34 | [TK]D-Fender | drutlandxpt: details would help. |
| 16:54.39 | [TK]D-Fender | question* |
| 16:54.45 | hmmhesays | everyone one of the headlines on nodaddy.com are related to seclists.org |
| 16:54.52 | gubluntu | hey everyone |
| 16:55.07 | dlynes | [TK]D-Fender: because that application doesn't seem to have an option to announce the parked call extension to the user without calling them back, and because the peopel that will be using it just plain don't care what extension it's parked on...it'll ring back in 30 seconds to all phones, anyways, so someone will pick it up |
| 16:55.16 | Qwell | hmmhesays: well, yeah, that's the guy who created nodaddy.org |
| 16:55.16 | gubluntu | i was wondering if someone could point me in the right direction |
| 16:55.23 | f00bar80 | [TK]D-Fender: ok simply is asterisk the only thing i have to use to start a VOIP long distance calls servie provider ? |
| 16:55.39 | gubluntu | what kind of hardware do i need to be running asterisk and freepbx besides the computer and an internet line? |
| 16:56.06 | [TK]D-Fender | dlynes: So they use it as a raww time-out to dial them back later with the call? |
| 16:56.26 | dlynes | [TK]D-Fender: yeah...one person will park it, and it should ring back to all 11 extensions |
| 16:56.31 | [TK]D-Fender | f00bar80: have you ever even USED * before? |
| 16:56.44 | drutlandxpt | ok. on my sip phone connecting to asterisk, when i use rtp or sip infndo as my selection, I can use dtmf with asterisk. However an outside line doesn't hear my tones. WHen I use inband, the outside line sorta works. It hears the tone, but then it stops all sound. I cannot hear anything coming from the other side |
| 16:56.56 | [TK]D-Fender | dlynes: Then screw parking. if you want to have it ring back everyone later make a dialplan script for that. |
| 16:57.11 | f00bar80 | [TK]D-Fender: only i know the main features |
| 16:57.21 | dlynes | [TK]D-Fender: then how am i going to park the call? |
| 16:57.28 | [TK]D-Fender | dlynes: do a blind transfer to a local channel that will answer, play Moh for a specified time and jsut dial everyone! |
| 16:57.44 | [TK]D-Fender | f00bar80: again very vague. |
| 16:57.45 | hmmhesays | Qwell, for a company that handles millions of registrations, there are bound to be some problems |
| 16:57.50 | f00bar80 | [TK]D-Fender: i'm asking cause i don't have enough info |
| 16:58.02 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 16:58.03 | Qwell | hmmhesays: when was the last time google had a problem? |
| 16:58.15 | f00bar80 | [TK]D-Fender: even correct me and point me to a guide for what i'm asking about |
| 16:58.21 | Qwell | or, you know...most other registrars? |
| 16:58.23 | [TK]D-Fender | f00bar80: We can't answer questions you don't have. Go play with * some more and tell us what you think is MISSING and we'll tell you what will do the job. |
| 16:58.29 | Qwell | I've never seen a problem with enom |
| 16:58.50 | dlynes | [TK]D-Fender: ah, that'll work (I think)....thanks for the idea |
| 16:58.54 | [TK]D-Fender | f00bar80: And there is no miracle guide for using * to setup an ITSP. What size? What kind of hardware are you considering, etc |
| 16:59.06 | [TK]D-Fender | dlynes: 6 lines of code. TOPS. |
| 16:59.10 | [intra]lanman | [TK]D-Fender: i think a _good_ solution to realtime and mwi is missing, can you tell me what will do the job? |
| 16:59.26 | dlynes | [TK]D-Fender: and that will keep the original caller on hold for the entire duration, right? |
| 16:59.28 | [TK]D-Fender | [intra]lanman: Whats the actual problem you're trying to solve? |
| 16:59.30 | hmmhesays | google, google checkout horror stories |
| 16:59.39 | [TK]D-Fender | dlynes: if you BLIND transfer, yes |
| 16:59.51 | dlynes | [TK]D-Fender: and if you do a regular transfer? |
| 16:59.52 | hmmhesays | as I said, Everyone has had some problems somewhere |
| 17:00.10 | hmmhesays | even google |
| 17:00.10 | f00bar80 | [TK]D-Fender: i'm considering small ISTP serving not more than 20 user, and already i have my hosting server, i'm asking what else i need |
| 17:00.12 | [TK]D-Fender | dlynes: Rules of physics... what do you THINK that local channel would see? YOUS <---- |
| 17:00.35 | dlynes | [TK]D-Fender: yous? |
| 17:00.39 | [TK]D-Fender | f00bar80: depends on what services you intend to use in providing them back to your clients. |
| 17:00.44 | [TK]D-Fender | YOURS * |
| 17:00.55 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 17:00.57 | f00bar80 | [TK]D-Fender: long distance calls , nothing more |
| 17:01.02 | [TK]D-Fender | dlynes: BLIND transfer = your inbound callers ID, Attended = YOURS. |
| 17:01.04 | [intra]lanman | [TK]D-Fender: my problem is that mwi doesn't get sent to users if i'm using realtime without rtcache (which kinda defeats the purpose) |
| 17:01.08 | [TK]D-Fender | dlynes: C'mon, this is * 101! |
| 17:01.49 | dlynes | [TK]D-Fender: so iow, it's not foolproof |
| 17:01.49 | [TK]D-Fender | [intra]lanman: Sorry, not RT experience |
| 17:01.49 | dlynes | [TK]D-Fender: the end user can still screw things up :) |
| 17:01.49 | gubluntu | can someone at least point me to a decent website with the information im looking for |
| 17:01.49 | [TK]D-Fender | dlynes: users can ALWAYS screw stuff up... |
| 17:01.53 | gubluntu | i dont quite understand how everythings connected once im running asterisk |
| 17:02.06 | [TK]D-Fender | dlynes: If you do an attended transfer to park & annouce you'll gt screwed the same way as well. |
| 17:02.30 | [TK]D-Fender | gubluntu: "everything"? huh? |
| 17:02.43 | [TK]D-Fender | dlynes: "get over it". |
| 17:02.52 | gubluntu | specifically, if i have asterisk and freepbx running on a linux box |
| 17:02.59 | gubluntu | what else is needed beside s the internet line? |
| 17:02.59 | hmmhesays | fun |
| 17:03.07 | gubluntu | how do phones jack in etc.. |
| 17:03.11 | hmmhesays | what a fantastically vague question |
| 17:03.33 | gubluntu | its not like im asking how long a piece of string is |
| 17:03.59 | drutlandxpt | [TK]D-Fender: do you have any ideas on my situation? |
| 17:04.05 | gubluntu | my current nortel pbx runs terminates into a patch panel |
| 17:04.20 | [intra]lanman | [TK]D-Fender: know of any way to get * to route a NOTIFY if i send the MWI from elsewhere (namely a sipsak on the same host) |
| 17:04.23 | gubluntu | do i need something similar for my linux box? |
| 17:04.46 | [TK]D-Fender | [intra]lanman: Nope, not a clue |
| 17:05.34 | [TK]D-Fender | gubluntu: You clearly need a T3 line, an Adtran de-mux and a 2 Sangoma A108d PRI cards.. |
| 17:06.11 | Qwell | Every time you buy Sangoma, their CEO eats a baby. |
| 17:06.25 | [TK]D-Fender | drutlandxpt: You should check on the mode used by your OUTSIDE link. |
| 17:06.35 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 17:06.49 | drutlandxpt | [TK]D-Fender: where is that at? |
| 17:06.51 | [TK]D-Fender | gubluntu: Oh yes... and a large supply a babies to feed their CEO |
| 17:07.07 | [TK]D-Fender | drutlandxpt: What are you using for this outside PSTN access? |
| 17:07.32 | drutlandxpt | [TK]D-Fender: a pri that is fed from my company. it is directly off a class 5 switch |
| 17:08.03 | [TK]D-Fender | drutlandxpt: what hardware are you using? |
| 17:08.10 | [TK]D-Fender | drutlandxpt: and what signalling? |
| 17:08.54 | drutlandxpt | [TK]D-Fender: zaptel 4 pri card pri_cpe |
| 17:09.12 | gubluntu | anyway i can just use asterisk,freepbx on a linux box and a virtual lan to terminate to regular cat5e jacks around the offce and just use some polycom ip phones |
| 17:09.28 | [TK]D-Fender | drutlandxpt: DTMF should be fine by default.... you should only have to concern yourself with your phones |
| 17:09.31 | gubluntu | everything running down a voip circuit provided by verizon? |
| 17:09.51 | gubluntu | =1.5 both ways |
| 17:09.53 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 17:09.55 | gubluntu | t1/e1 |
| 17:10.15 | [TK]D-Fender | gubluntu: What exactly is a "VoIP Circuit"? Thats a nifty new term... |
| 17:10.28 | gubluntu | whatever internet circuit they provided me for voip |
| 17:10.33 | dlynes | [TK]D-Fender: ah, ok...didn't even know about it |
| 17:10.46 | drutlandxpt | [TK]D-Fender: the only thing that seems to work fully is inband, but when i press a number, it goes totally silent. it doesn't drop, just sits there |
| 17:10.50 | dlynes | [TK]D-Fender: but on another note, i've figured out to deal with that issue |
| 17:10.51 | gubluntu | its a regular t1.e1 line |
| 17:11.03 | [TK]D-Fender | gubluntu: You clearly need to get a grip with what you HAVE and what you WANT. |
| 17:11.27 | [TK]D-Fender | gubluntu: T1 DATA?! Just a normal data link for connectivity to the INTERNET? |
| 17:11.40 | [TK]D-Fender | dlynes: Which issue? |
| 17:13.18 | gubluntu | what do you not understand... i was provided with a t1/e1 circuit that is capable of running a hosted voip solution provided by veriuzon... circuit connects to adtran, adtran to lan, lan to patch, patch to plycom phones |
| 17:13.27 | gubluntu | can i get asterisk in there some how |
| 17:14.13 | drutlandxpt | [TK]D-Fender: have you heard of that before? |
| 17:14.57 | [TK]D-Fender | gubluntu: T1 is merely a carrier technology. PRI is a voice signalling often used. There are others, and there is also jsut raw data, etc. You were so vague it could have meant ANYTHING. |
| 17:14.58 | dlynes | [TK]D-Fender: where the callerid takes on the parker's callerid on an attended transfer |
| 17:15.17 | [TK]D-Fender | gubluntu: And if its jsut a link to the internet, sure you can run *, why the hell not? |
| 17:15.29 | [TK]D-Fender | dlynes: And how would you solve that? |
| 17:15.41 | dlynes | [TK]D-Fender: asterisk db |
| 17:16.00 | [TK]D-Fender | dlynes: Oh, do share :) This outta be nifty! |
| 17:16.07 | [TK]D-Fender | oughtta* |
| 17:16.21 | gubluntu | [TK]D-Fender: im trying to figure out where the * needs to sit.. on the lan? |
| 17:16.37 | [TK]D-Fender | gubluntu: sure |
| 17:17.49 | dlynes | [TK]D-Fender: grab the callerid when the call comes in, put it in asterisk db, for the channel identifier; when the call gets transferred into the local extension, and then times out, grab the callerid back from asteriskdb, set it, and ring all phones |
| 17:19.12 | [TK]D-Fender | dlynes: And what if MULTIPLE calls get parked? |
| 17:19.34 | [TK]D-Fender | dlynes: And if EVERY CID gets pushed, only the LATEST gets pulled. |
| 17:19.38 | lirakis | <PROTECTED> |
| 17:19.44 | lirakis | http://ca.prweb.com/releases/2007/9/prweb552202.htm |
| 17:19.52 | dlynes | [TK]D-Fender: i assign it to the channel number/identifier |
| 17:20.09 | dlynes | [TK]D-Fender: read my original message...you'll see I already said that |
| 17:20.26 | lirakis | wrong link ... http://www.grandstream.com/gxp1200.html |
| 17:20.28 | [TK]D-Fender | lirakis: Ohhh yeah, sign me up for 0 of those!!! |
| 17:20.31 | dlynes | i.e. Zap/1-1 |
| 17:20.35 | lirakis | <PROTECTED> |
| 17:20.40 | lirakis | i thought you would like it |
| 17:20.46 | [TK]D-Fender | Craptastic! |
| 17:20.49 | dlynes | [TK]D-Fender: or SIP/321-3F08DE |
| 17:20.53 | [TK]D-Fender | dlynes: SHOW ME THE MONEY :) |
| 17:21.15 | dlynes | [TK]D-Fender: I'm too poor already...can't afford to give anyone else my moolah |
| 17:21.52 | [TK]D-Fender | dlynes: Its a Jerry McGuire line... geez... |
| 17:22.09 | [TK]D-Fender | dlynes: I mean pastbin how you think you've "solved" this, so I can punch some holes in your bubble! |
| 17:22.30 | Nugget | heh |
| 17:22.56 | dlynes | [TK]D-Fender: i know that, sheesh |
| 17:23.04 | dlynes | [TK]D-Fender: i was just being facetious |
| 17:23.09 | [TK]D-Fender | dlynes: Because the second I hear ASTDB< I KNOW this ship is gonna sink :p |
| 17:23.20 | dlynes | [TK]D-Fender: astdb is crap? |
| 17:23.42 | dlynes | [TK]D-Fender: the other solution i was thinking of was global variables, but I don't like using those animals |
| 17:23.55 | [TK]D-Fender | dlynes: jsut show me how you have set this up..... |
| 17:24.02 | dlynes | i don't yet |
| 17:24.08 | dlynes | It was just a solution I had thought of |
| 17:24.22 | [TK]D-Fender | dlynes: Globals can't handle MULTIPLE "parked" calls. |
| 17:24.37 | [TK]D-Fender | dlynes: Thats going to be the big "catch" |
| 17:24.45 | dlynes | [TK]D-Fender: one global for each parked call |
| 17:25.06 | [TK]D-Fender | dlynes: You'll never know how to pair them up again... keep trying... |
| 17:25.18 | dlynes | i.e. PARKEDCALL1_CHANIDENT=, PARKEDCALL1_CALLERIDNUM=,PARKEDCALL1_CALLERIDNAME= |
| 17:25.42 | dlynes | and i parse ouit the zaptel channel, to determine which one to use |
| 17:25.55 | [TK]D-Fender | dlynes: even nastier as an attended transfer is a new call and you won't have a match for an originating channel as thats the answering side. |
| 17:26.02 | dlynes | because in this scenario, there are no voip calls |
| 17:26.26 | [TK]D-Fender | dlynes: ok, try and code it up and then jsut show me. |
| 17:26.39 | dlynes | Will do, but it'llhave to wait a couple of hours |
| 17:26.44 | dlynes | Need to head off to a job site first |
| 17:26.47 | [TK]D-Fender | dlynes: I'm patient :) |
| 17:28.12 | *** join/#asterisk vadiml1024 (n=vadim@LAubervilliers-153-52-29-171.w217-128.abo.wanadoo.fr) |
| 17:30.12 | vadiml1024 | hi i need a little assitance to set a loopback between 2 port of Wildcard TE210P |
| 17:30.55 | [TK]D-Fender | vadiml1024: Just google up "t1 cross-over cable" and you'll find specs in about 10 seconds flat |
| 17:31.23 | vadiml1024 | i did that, i've problem with a zaptel.conf: |
| 17:31.27 | vadiml1024 | span=1,0,0,ccs,hdb3 |
| 17:31.31 | vadiml1024 | # termtype: te |
| 17:31.33 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:31.35 | vadiml1024 | bchan=1-15,17-31 |
| 17:31.39 | vadiml1024 | dchan=16 |
| 17:31.43 | vadiml1024 | # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4 |
| 17:31.47 | vadiml1024 | span=2,1,0,ccs,hdb3 |
| 17:31.48 | vadiml1024 | # termtype: te |
| 17:31.48 | vadiml1024 | bchan=32-46,48-62 |
| 17:31.48 | vadiml1024 | dchan=47 |
| 17:31.48 | vadiml1024 | loadzone = fr |
| 17:31.48 | vadiml1024 | defaultzone = fr |
| 17:32.01 | [TK]D-Fender | vadiml1024: PASTEBIN |
| 17:32.05 | Corydon76-dig | ~pb |
| 17:32.05 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:32.07 | [TK]D-Fender | vadiml1024: do NOT spam in here |
| 17:32.31 | vadiml1024 | sorry.... |
| 17:32.43 | [TK]D-Fender | vadiml1024: So whats the problem? |
| 17:33.25 | file | [TK]D-Fender: going to the montreal asterisk user's group meeting thingy? |
| 17:33.43 | [TK]D-Fender | file: Whcih, when? |
| 17:33.45 | vadiml1024 | with above zaptel.conf when doing ztcfg i get |
| 17:33.48 | vadiml1024 | ZT_CHANCONFIG failed on channel 49: No such device or address (6) |
| 17:33.51 | file | [TK]D-Fender: this Friday |
| 17:34.08 | [TK]D-Fender | vadiml1024: Almost gauranteed taht you didn't set the E1 jumper on your cad |
| 17:34.17 | [TK]D-Fender | vadiml1024: card |
| 17:34.22 | [TK]D-Fender | file: Time & location? |
| 17:34.28 | vadiml1024 | ahhh....... Thanksa ton!!!!! |
| 17:34.41 | file | [TK]D-Fender: ummm |
| 17:34.54 | [TK]D-Fender | NEXT!@!@@ (c) BKW |
| 17:35.19 | file | [TK]D-Fender: http://forums.amug.ca/viewtopic.php?t=3162 |
| 17:35.39 | drutlandxpt | [TK]D-Fender: do you know of how I can have the dtmf between sip client and asterisk as sip info, but when it comes to the zaptel it is inband? |
| 17:36.11 | *** join/#asterisk TrevorSHarrison (n=trevorsh@24-49-36-218-st.chvlva.adelphia.net) |
| 17:36.45 | [TK]D-Fender | file: And you're speaking too? Very cool, definately in. |
| 17:36.54 | file | I was voluntold |
| 17:37.21 | [TK]D-Fender | file: voluntold = ordered to come willingly? |
| 17:37.26 | file | :D |
| 17:37.28 | Qwell | somebody buy me a plane ticket, and I'll be there :p |
| 17:37.40 | [TK]D-Fender | drutlandxpt: test each INDEPENDANTLY. |
| 17:38.00 | file | Qwell: get Christen to put it against the coffee budget! |
| 17:38.09 | Qwell | can do |
| 17:40.28 | tristanbob | does asterisk have a voice directory, so that I can simply speak the name I want? |
| 17:41.12 | [TK]D-Fender | tristanbob: No |
| 17:41.14 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:41.21 | putnopvut | Qwell: wouldn't you need a passport? |
| 17:41.27 | file | ha! he would |
| 17:41.29 | Qwell | foiled! |
| 17:41.36 | tristanbob | [TK]D-Fender: any third-party extensions that provide that? |
| 17:41.45 | [TK]D-Fender | tristanbob: Go lookup "voice recognition" on the WIKI and you can start with Sphinx |
| 17:41.58 | tristanbob | [TK]D-Fender: ok - will do - thanks |
| 17:42.25 | [TK]D-Fender | tristanbob: Free ones suck back, payed ones suck less, all are more trouble than they're worth |
| 17:42.52 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:42.52 | tristanbob | I love the one we use, not sure what brand -( avaya pbx, audix voice mail) |
| 17:43.01 | Qwell | ugh, audix |
| 17:43.11 | Qwell | I *hate* audix |
| 17:43.45 | tristanbob | Qwell: not sure who does the voice directory |
| 17:44.08 | Qwell | probably the phone lady |
| 17:44.34 | Qwell | (at my last job, the phone person was a woman, so it's always the phone lady to me... not trying to be sexist or anything) |
| 17:44.47 | mrempire | can aterisk ignore one msn |
| 17:44.57 | Qwell | what a weird lady she was too... |
| 17:45.07 | Qwell | shook her head when she walked...quite bizarre |
| 17:45.21 | Qwell | ...but I digress |
| 17:46.01 | viKing78 | How do you do a voice mail distribution in Asterisk? |
| 17:46.05 | [TK]D-Fender | mrempire: rephrase please.... |
| 17:46.11 | Qwell | viKing78: explain |
| 17:46.23 | [TK]D-Fender | viKing78: Voicemail(1&2&3&4&5&6,b) |
| 17:46.57 | mrempire | fender, I have 4 msns but i don't want to use 1 msn on asterisk |
| 17:47.05 | viKing78 | [TK]D-Fender: Is that a dispatch or broadcast? What I mean is if one user deletes dies it clear the message for the rest? |
| 17:47.19 | viKing78 | *does |
| 17:47.21 | [TK]D-Fender | viKing78: No, each is independant |
| 17:47.23 | drutlandxpt | [TK]D-Fender: SIPDtmfMode can be used to set the dtmf from a sip channel to inband when it goes to a zap channel |
| 17:47.24 | drutlandxpt | ? |
| 17:47.25 | mrempire | I want to use it with my standard isdn phone, or else my wife will kill me |
| 17:47.37 | [TK]D-Fender | drutlandxpt: No, you should never have to do anything like that |
| 17:47.40 | viKing78 | [TK]D-Fender: Is there a way to have it work the other way? |
| 17:47.47 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 17:48.18 | [TK]D-Fender | mrempire: Just don't ahve a parrtern match for that MSN. |
| 17:48.32 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 17:48.36 | [TK]D-Fender | viKing78: No normal way..... |
| 17:48.56 | viKing78 | [TK]D-Fender: What do you mean by "normal"? |
| 17:49.06 | [TK]D-Fender | viKing78: closest alternative si make a shared box that your phones get MWI for, but its still ANOTHER box you have to check. |
| 17:49.26 | *** join/#asterisk lukketto (n=lukketto@host80-193-dynamic.7-87-r.retail.telecomitalia.it) |
| 17:49.43 | *** join/#asterisk CVirus (n=GoD@196.205.192.229) |
| 17:49.51 | flujan | join #macosx |
| 17:50.04 | [TK]D-Fender | flujan: I'd rather not :p |
| 17:50.04 | *** part/#asterisk TrevorSHarrison (n=trevorsh@24-49-36-218-st.chvlva.adelphia.net) |
| 17:50.17 | mrempire | Fender, If i don't have a match in extensions.conf it gives a bussy tone |
| 17:50.23 | [TK]D-Fender | I'd sooner switch to a Linux distro than to MacOS :) |
| 17:50.32 | viKing78 | [TK]D-Fender: When you call Voicemail(1&2&3&4,b) which greeting is played? |
| 17:50.36 | flujan | [TK]D-Fender: lol... just forget to put a / |
| 17:50.38 | flujan | :P |
| 17:50.43 | russellb | viKing78: 1 |
| 17:50.57 | viKing78 | [TK]D-Fender: Thought so but wanted to check, thanks |
| 17:50.58 | flujan | man I am amused... asterisk is working for 15 days with no problems... |
| 17:51.08 | [TK]D-Fender | mrempire: show me the CLI output of the failed attempt at verbose 10, and channel debug enabled. |
| 17:51.08 | flujan | it never happened to me before... :) |
| 17:51.14 | [TK]D-Fender | ~pb |
| 17:51.14 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:51.24 | Qwell | [TK]D-Fender: You wouldn't run osx on a normal x86 laptop if you could? |
| 17:51.47 | flujan | I must have learned something about asterisk configuration and deployment. :D |
| 17:52.28 | [TK]D-Fender | Qwell: MacOS really doesn't offer me anything interesting. Heck its hard to find a good free OSS FTP app and so much more. Ubuntu offers tons of stuff right up front. |
| 17:52.37 | Qwell | fair enough |
| 17:52.47 | [TK]D-Fender | flujan: You learn quickly my young Jedi.... |
| 17:53.27 | [TK]D-Fender | Qwell: Especailly as I'm not some Garageband / Photoshop chump. |
| 17:53.29 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 17:53.42 | flujan | [TK]D-Fender:thanks master... and I refuse to join the dark side of the force... |
| 17:53.43 | flujan | :D |
| 17:53.54 | [TK]D-Fender | Qwell: I WOULD like to get QSynth, and Rosegarden fully operational on my laptop however.... |
| 17:54.05 | Qwell | yeah...rosegarden is seriously buggy for me |
| 17:54.13 | *** join/#asterisk Op3r (n=Op3r@121.97.193.51) |
| 17:54.13 | Qwell | it locks up my whole machine sometimes (most of the time...) |
| 17:54.22 | Qwell | hard lock - even ssh dies |
| 17:54.27 | [TK]D-Fender | Qwell: Ouch |
| 17:54.29 | Qwell | yeah |
| 17:54.56 | Qwell | it might have something to do with 64-bittedness...no idea |
| 17:55.07 | [TK]D-Fender | Qwell: I basically need multi-track audio + MIDI with a live soft-synth thats SF2 compatible. |
| 17:55.17 | mrempire | fender, please have a look at http://pastebin.ca/692302 |
| 17:55.25 | Qwell | yeah, rosegarden should do that |
| 17:55.30 | Qwell | it's really nice when it works |
| 17:55.43 | Qwell | it's pretty impressive, really |
| 17:56.03 | mrempire | in my /etc/misdn.conf i did not included that msn |
| 17:56.04 | [TK]D-Fender | mrempire: -- Executing [i@default:1] Playback("mISDN/1-u0", "invalid") in new stack <--- you are ANSWERING THE CALL! |
| 17:56.37 | [TK]D-Fender | mrempire: [Sep 11 19:44:41] WARNING[3298]: chan_misdn.c:4269 cb_events: Extension can never match, So jumping to 'i' extension. port(1) <------------- do NOT fail through to "i" |
| 17:56.52 | [TK]D-Fender | mrempire: Comment that out and you should be fine |
| 17:58.11 | mrempire | fender, but if I didn't included that msn in misdn.conf than i should not have answered |
| 17:58.47 | mrempire | in the mean time i will search for the 1 in extensions.conf |
| 17:59.08 | mrempire | sorry i mean i extension |
| 18:00.07 | hmmhesays | does the polycom ip-320 not have a headset jack? |
| 18:01.14 | [TK]D-Fender | hmmhesays: it does, a 2.5 mm on the right-hand side |
| 18:01.33 | [TK]D-Fender | hmmhesays: 320/330 = 2.55 side mount, all others = RJ9 |
| 18:01.36 | [TK]D-Fender | (on back) |
| 18:01.36 | mrempire | Thanks Fender |
| 18:02.13 | GlobeTrotter | hola.. getting this error at the console mpg123: no process killed Asteris ended with error code 1, automatically restating |
| 18:02.20 | GlobeTrotter | anyone seen that before? |
| 18:02.31 | GlobeTrotter | im using a .gsm file for moh |
| 18:02.58 | [TK]D-Fender | GlobeTrotter: And since when would you use mpg123 to play GSM files?! |
| 18:03.13 | [TK]D-Fender | mrempire: you're welcome. I take it that everything woks now? |
| 18:03.26 | hmmhesays | [TK]D-Fender a single, not one for mic, one for speaker? |
| 18:03.44 | GlobeTrotter | thats my point.. in npt using mpg123, but i still get this error |
| 18:03.49 | [TK]D-Fender | hmmhesays: Correct. like a cell-phone headset, and like the SPA's use |
| 18:04.00 | hmmhesays | oh 2.5mm |
| 18:04.02 | hmmhesays | right? |
| 18:04.03 | [TK]D-Fender | hmmhesays: You can get an adapter for that though I'm sure |
| 18:04.08 | mrempire | Fender, no not yet, I'm looking trough the extensions.conf |
| 18:04.15 | [TK]D-Fender | hmmhesays: Are you even READING my answers? :) |
| 18:04.27 | [TK]D-Fender | [14:01]<[TK]D-Fender>hmmhesays: 320/330 = 2.55 side mount, all others = RJ9 |
| 18:04.48 | Qwell | when're they gonna add bluetooth headset support? |
| 18:05.06 | GlobeTrotter | i am wondering why am i getting this error when im not using mp3 files or mpg123 |
| 18:05.08 | [TK]D-Fender | Qwell: Whent he market actually gives a shit :p |
| 18:05.34 | hmmhesays | [TK]D-Fender yeah sorry distracted |
| 18:05.44 | [TK]D-Fender | GlobeTrotter: apparently you ARE |
| 18:07.52 | GlobeTrotter | ok thanks,, ill try to see what i cn figure out.. muchas gracias |
| 18:13.24 | mrempire | Fender, can you please have a look at my extensions.conf http://pastebin.ca/692335 |
| 18:13.30 | *** join/#asterisk Kurin- (n=Kurin@lithium.delete.org) |
| 18:13.49 | Kurin- | Does anyone here have experience setting up MWI on Polycom phones? |
| 18:13.57 | Kurin- | I can't for the life of me figure out how to get that stupid light to blink |
| 18:14.14 | Kurin- | and neither can I find any documentation on how the phone checks for new messages |
| 18:14.45 | mrempire | The extensions.conf does not work as I expect ;( |
| 18:15.56 | *** join/#asterisk lbow (n=lbow@41-195-77-184.access.uunet.co.za) |
| 18:16.07 | *** join/#asterisk pacneil (n=pacneil@68.15.17.81) |
| 18:17.48 | Shido6 | Kurin |
| 18:17.55 | Shido6 | what are you using for voicemail? |
| 18:18.11 | Kurin- | Just the asterisk scripts |
| 18:18.31 | Shido6 | does regular vmail work for you yet? |
| 18:18.35 | Kurin- | Yeah |
| 18:18.46 | Shido6 | in sip.comf what do you have set for "mailbox" for that phones peer ? |
| 18:19.00 | Kurin- | 114@voicemail |
| 18:19.06 | Kurin- | 114 is the extension and the mailbox name |
| 18:19.11 | Kurin- | voicemail is the context |
| 18:19.40 | Kurin- | Although `sip show peers` shows nothing |
| 18:19.48 | Kurin- | even though the phones, except for this, appear to work fine |
| 18:20.08 | Shido6 | um |
| 18:20.10 | Shido6 | weird |
| 18:20.16 | Shido6 | you should see your phone |
| 18:20.18 | Shido6 | s ip |
| 18:20.20 | Kurin- | yeah |
| 18:20.23 | Shido6 | when u do a sip show peer |
| 18:20.32 | Shido6 | and when u make a phone call it works / |
| 18:20.36 | Kurin- | Yep |
| 18:20.42 | Shido6 | and receive a call |
| 18:20.43 | Shido6 | ? |
| 18:20.46 | Kurin- | I can even plug in the PRI line and get outgoing and incoming |
| 18:20.47 | Kurin- | Yep |
| 18:20.53 | [TK]D-Fender | mrempire: show us WHERE since we aren't PSYCHIC and don't know what you "expect" |
| 18:21.09 | Shido6 | do u have mwi msg.mwi.1.subscribe= to anything in your cfg file? |
| 18:21.23 | [TK]D-Fender | Shido6: Shouldn't have to. |
| 18:21.37 | [TK]D-Fender | Kurin-: pastebin your voicemail.conf and your sip.conf entrey for that phone. |
| 18:21.40 | [TK]D-Fender | ~pb |
| 18:21.40 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:21.51 | Kurin- | Yeah I wasn't quite sure what the subscribe should be, I've tried it both ways |
| 18:21.53 | Shido6 | might want to pastebin your phone cfg, too |
| 18:22.06 | Kurin- | yeah one sec |
| 18:22.53 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:22.54 | Kurin- | The sip.conf is actually via odbc, so I have to format it |
| 18:22.56 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 18:24.15 | Shido6 | aha |
| 18:24.21 | Shido6 | in odbc eh? |
| 18:24.33 | Kurin- | yeah |
| 18:24.43 | Kurin- | but it registers and even updates the database with the correct IP |
| 18:25.15 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:25.28 | *** join/#asterisk agx (n=badpengu@81-174-8-228.dynamic.ngi.it) |
| 18:25.29 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 18:27.45 | Qwell | 15 minutes to upgrade firmware...that's pretty ridiculous |
| 18:28.05 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 18:28.16 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:28.40 | Qwell | saving application, checking application, loading application...wtf |
| 18:28.50 | Qwell | if it had to save it, wouldn't it have had to have been loaded? |
| 18:29.24 | nny | soo... I know that the asteriskgui is generally frowned upon. We are working on setting up something for our clients to do basic user changes etc. and we are checking it out. |
| 18:29.37 | nny | quick q about it. It seems to read users.conf. Why? |
| 18:29.41 | Kurin- | god I hate giant db tables |
| 18:29.51 | *** join/#asterisk m0t3jl (n=m0t3jl@ip103.galance.net) |
| 18:30.00 | Kurin- | sip_conf has like 30 relations |
| 18:30.00 | Qwell | nny: because that's how it was written |
| 18:31.06 | nny | Qwell: lol |
| 18:31.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-177-39.red.bezeqint.net) |
| 18:31.19 | nny | Qwell: so is users.conf a alt for files like sip.conf? |
| 18:32.01 | deeperror | Anyone know what could cause this? http://pastebin.ca/692375 |
| 18:32.02 | Kurin- | Shido6, [TK]D-Fender: http://paste.lisp.org/display/47586 |
| 18:33.34 | deeperror | messages reaches a gig in about a minute |
| 18:35.10 | mcab | Qwell: downloading and saving is working with a compressed app; loading application is decompressing it from flash and running it. (but, yeah, it's a PITA) |
| 18:36.51 | [TK]D-Fender | Kurin-: LOOKS fine. how about checking that there is in fact NEW mail waiting? |
| 18:37.13 | Kurin- | yeah there is |
| 18:37.20 | Kurin- | "you have" "one" "new message" |
| 18:37.49 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 18:39.11 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 18:39.22 | *** join/#asterisk roxy_ (n=roxy_@4.249.97-84.rev.gaoland.net) |
| 18:40.01 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:40.11 | *** join/#asterisk Poehali (n=actionma@74.93.5.186) |
| 18:42.27 | Poehali | hey guys |
| 18:42.42 | Poehali | I still can't get SPA3102 to work on asterisk |
| 18:42.45 | [TK]D-Fender | Kurin-: Hrm.... |
| 18:43.03 | [TK]D-Fender | Poehali: And do you have anything more to say for yourself than you did last night? |
| 18:43.23 | Poehali | I put in proxy, outbound proxy of asterisk box ip, make call and answer without reg:yes, and subscriber information, but subscription always fails |
| 18:43.32 | Poehali | [TK]D-Fender: yes |
| 18:43.44 | deeperror | anyone ever seen errors like this http://pastebin.ca/692375 or have any clues what to look into on this |
| 18:43.47 | Poehali | [TK]D-Fender: and I got in a fenderbender last night |
| 18:44.14 | _ShrikE | Im having an issue with username/authname mismatch when registering multiple lines from the same device in 1.4.11. Bug 9044 seems to acknowledge the issue, does anyone know if this is going anywhere? |
| 18:44.19 | MrMister2 | Question: I keep getting "Remote UNIX connection |
| 18:44.43 | MrMister2 | on the CLI of Asterisk. How can I get rid of those lines? |
| 18:45.06 | Kurin- | maybe the phones are somehow not registering as peers? |
| 18:45.36 | Poehali | TK? |
| 18:45.40 | Kurin- | even though they should be "friends" |
| 18:45.49 | MrMister2 | [TK]D-Fender: BTW, your sugestion to increase the time led me to the solution to my lack of sound. Once I inserted a delay before playing a sound it worked fine. |
| 18:46.39 | [TK]D-Fender | MrMister2: Good to hear |
| 18:46.55 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 18:47.33 | Kurin- | sip show users is equally blank |
| 18:48.48 | Nugget | I'm getting lots of dtmf problems on our pri where callers can't navigate the ivr because asterisk isn't properly decyphering the dtmf |
| 18:48.55 | Nugget | any hints on where to look to debug that? |
| 18:49.21 | Nugget | naturally it works whenever I call in to test it |
| 18:51.15 | Poehali | [TK]D-Fender: so PTSN line status shows line voltage: -50 (V), registration state: failed |
| 18:51.44 | *** part/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 18:53.37 | agx | i experience DB corruption in astdb, there is a way i can check it before starting asterisk start so i can zero it? |
| 18:53.46 | MrMister2 | I'm getting a msg on the CLI of * roughly every second. Very annoying and it fills the log file like crazy, any idea on how I can deactivate it? |
| 18:55.16 | agx | MrMister2 1) don't use CLI 2) comment it out into the code 3) check manager.conf sample there is an option for it i suppose |
| 18:57.22 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 18:57.24 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 18:57.24 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:57.34 | Kurin- | What's wrong with the CLI? |
| 18:58.20 | Kurin- | Interesting |
| 18:58.20 | MrMister2 | agx: The message I meatn was the one I said above. "Remote UNIX connection" I assume this might be because I have the verbosity too high? |
| 18:58.24 | MrMister2 | *meant |
| 18:58.39 | Kurin- | Now that I've put my info in sip.conf directly, it works and I get a vmail light |
| 19:01.13 | *** join/#asterisk heartones (n=heartone@196.218.34.246) |
| 19:04.02 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 19:05.25 | agx | MrMister2 as i said, Manager.conf, try with displayconnects = yes/no |
| 19:06.37 | *** join/#asterisk pjz (n=pj@66.219.59.183) |
| 19:07.02 | pjz | anyone have suggestions on an asterisk appliance? |
| 19:07.16 | Strom_M | the asterisk appliance :) |
| 19:07.18 | pjz | I wouldn't mind Digium's, if I can find one |
| 19:07.21 | Strom_M | that's my suggestion |
| 19:07.36 | pjz | okay, so where can I order one with 4 FXS lines? |
| 19:07.53 | pjz | or do I need FXO? bah, I always forget |
| 19:07.59 | pjz | I need to use 4 POTS lines |
| 19:08.05 | pjz | all internal phones will be SIP |
| 19:08.14 | dasuberdavid | Order a Digium TDM04B |
| 19:08.18 | dasuberdavid | 4 FXO modules |
| 19:08.31 | pjz | okay, so I need FXO not FXS then? |
| 19:08.44 | dasuberdavid | If you are connecting the card to POTS lines, you need FXO modules |
| 19:08.45 | dasuberdavid | correct |
| 19:08.59 | pjz | okay |
| 19:09.14 | pjz | so I'd like an asterisk appliance with 4+ FXO lines |
| 19:09.21 | pjz | suggestions anyone? |
| 19:09.38 | Strom_M | pjz: digium appliance |
| 19:09.43 | dasuberdavid | exactly |
| 19:09.46 | pjz | Strom_M: from where? |
| 19:09.49 | Nugget | http://www.digium.com/en/products/hardware/asteriskappliance.php |
| 19:10.10 | pjz | I looked at digium's website, but they don't sell direct, and their distributors all seem to suck |
| 19:10.27 | pjz | though I'm tempted to get the AADK and use it in production |
| 19:10.34 | *** join/#asterisk tomcontr3 (n=tomcontr@82-161-246-201.adsl.terra.cl) |
| 19:10.42 | Strom_M | pjz: well it did just start shipping last week |
| 19:10.51 | pjz | Strom_M: ah, that would explain it :) |
| 19:11.01 | tomcontr3 | hi, I just bought a Digium TDM400 with to FXO modules |
| 19:11.05 | pjz | voipsupply.com only has 'VOIP only' versions |
| 19:11.11 | *** join/#asterisk YoYo (n=chatzill@12.196.144.37) |
| 19:11.24 | YoYo | can anyone here help me get a sangoma card set up on freebsd? |
| 19:12.11 | pjz | Strom_M: any word on if you can, say, buy FXO modules and switch out the FXS modules in one? or are they hard wired? |
| 19:12.13 | Corydon76-dig | Have you tried Sangoma? |
| 19:12.23 | Corydon76-dig | Because they should know how to set up their cards |
| 19:12.25 | Kurin- | So why would `sip show peers` not show realtime peers? |
| 19:12.30 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@015-802-211.area5.spcsdns.net) |
| 19:12.35 | YoYo | been digging on their site for a while... looking for peer support before calling them :) |
| 19:12.40 | jwh | it does realtime lookups |
| 19:12.44 | agx | pjz: you should ask yourself why distributor does not sell it :) |
| 19:12.48 | jwh | it will only show the realtime peers if its cached |
| 19:12.52 | Corydon76-dig | YoYo: if they say no, then that's that |
| 19:12.53 | jwh | rtcachefriends in sip.conf |
| 19:12.54 | [TK]D-Fender | Poehali: if your registration failed then you set the wrong credentials. |
| 19:12.56 | pjz | agx: because it just started shipping last week |
| 19:13.00 | Kurin- | jwh: thanks |
| 19:13.00 | tomcontr3 | but I dont know how to install it |
| 19:13.01 | YoYo | who said no? |
| 19:13.05 | tomcontr3 | I mean configur it |
| 19:14.46 | MrMister2 | agx: mmm... I can't sem to find anything on manager.conf that seems to be related to it. Any hints on where it could be? |
| 19:14.57 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:14.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:15.07 | agx | MrMister2 put displayconnections=no in [general] section |
| 19:15.21 | MrMister2 | agx: ah. Thnks :) |
| 19:15.35 | agx | MrMister2 btw check manager.conf sample or manager.c in asterisk source; i cannot remember the spell of the option |
| 19:15.53 | tomcontr3 | can anyone help me to configure my TDM400P? |
| 19:15.55 | MrMister2 | agx: I just googled it and got no hits |
| 19:16.07 | tomcontr3 | Im getting some errors when starting asterisk |
| 19:17.17 | MrMister2 | agx: the correct is displayconnects=no Thanks for the answer :) |
| 19:18.09 | tzafrir_laptop | tomcontr3, what errors do you get? |
| 19:18.22 | agx | MrMister2, /usr/src/asterisk/CORE-1.4/asterisk-1.4.11/main/manager.c (displayconnects = yes/no) |
| 19:18.29 | *** part/#asterisk agx (n=badpengu@81-174-8-228.dynamic.ngi.it) |
| 19:19.00 | tomcontr3 | something about zaptel and IAX |
| 19:19.14 | tomcontr3 | I can send you the logs via pastebin |
| 19:19.34 | tomcontr3 | http://pastebin.ca/692443 |
| 19:19.43 | *** join/#asterisk limbje (n=root@limbique.xs4all.nl) |
| 19:19.47 | drwelby | In iax.conf, for a Zoiper client, should DTMFMODE=AUTO work ok, or is there a picky setting? |
| 19:19.59 | ManxPower | tomcontr3: Sep 11 11:15:34 ERROR[2331] chan_zap.c: Unable to load config zapata.conf |
| 19:20.08 | ManxPower | you don't have /etc/asterisk/zapata.conf or it is not valid |
| 19:20.14 | limbje | hi |
| 19:20.37 | ManxPower | drwelby: iax does not have a dtmfmode |
| 19:20.57 | ManxPower | IAX only has one DTMF mode, you can't turn it off, you can't change it. |
| 19:21.03 | *** join/#asterisk sashion (n=sdgsdg@dsl-241-202-136.telkomadsl.co.za) |
| 19:21.15 | tomcontr3 | that file exist in /etc/zapata.conf |
| 19:21.22 | drwelby | ManxPower: Well, that simpifies the troubleshooting then! |
| 19:21.24 | ManxPower | (btw, this is not an issie as you should never have to turn it off or change it. |
| 19:21.41 | ManxPower | tomcontr3: you need TWO files. /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 19:21.44 | drwelby | Must be on the Zoiper end |
| 19:22.02 | tomcontr3 | ohhh you are right |
| 19:22.14 | *** join/#asterisk los415 (i=los415@LAX-DHCP-64-201-109-227.race.com) |
| 19:22.24 | tomcontr3 | now I get this |
| 19:22.24 | tomcontr3 | http://pastebin.ca/692450 |
| 19:22.32 | sashion | anyone had segmentation faults in ast_senddigit_end() ? |
| 19:26.06 | tomcontr3 | this is what I have in the zapata.conf file |
| 19:26.06 | tomcontr3 | http://pastebin.ca/692456 |
| 19:26.15 | *** join/#asterisk smace (n=chatzill@200.220.198.107) |
| 19:26.24 | smace | hello !! I am using Monitor for recording calls. But I would to have calls saved in mp3 format instead of wav. Is it possible? |
| 19:26.27 | limbje | i need to register to my provider with a user and a peer |
| 19:26.47 | sashion | smace, lookup MONITOR_EXEC |
| 19:26.49 | limbje | can't find if my peer and user is connected... |
| 19:26.50 | YoYo | smace: edit res_monitor.c to tell soxmix to use mp3 on the outfile |
| 19:27.15 | sashion | YoYo: whats the cpu implications of encoding directly to mp3 ? |
| 19:27.34 | YoYo | can't encode directly to mp3... can only mix the -in and -out streams |
| 19:28.00 | roxy_ | does someone knows of a command line softphone ? I just want to test if I can register with asterisk. |
| 19:28.16 | YoYo | and on my box, it's not noticible... but, I only ever have 4-5 calls at a time, and I'd expect it to be a rare event when 2 calls end at the same moment |
| 19:28.37 | tzafrir_laptop | tomcontr3, you can get a working sample by running xpp/utils/genzaptelconf in the zaptel directory |
| 19:28.51 | tzafrir_laptop | look at /etc/asterisk/zapata-channels.conf then |
| 19:29.08 | smace | YoYo: should this file be in /usr/include/asterisk/? |
| 19:29.08 | sashion | YoYo: I have a system running about 80 - 180 simultaneous calls at a time, and I've been looking at mp3 recording, but am too worried about performance impact |
| 19:29.20 | sashion | smace: Asterisk source code |
| 19:29.26 | sashion | where you compiled asterisk from |
| 19:29.33 | smace | sashion: but it means I should recompile asterisk no? |
| 19:29.39 | sashion | yes |
| 19:29.49 | sashion | you've done it once, I'm sure you can again |
| 19:29.50 | limbje | sip show users |
| 19:29.51 | sashion | :) |
| 19:29.52 | limbje | lol |
| 19:30.00 | smace | sashion: the asterisk I have was not compiled by me. |
| 19:30.15 | smace | is there any other way? |
| 19:30.46 | limbje | is a user to make calls or to receive calls |
| 19:30.46 | sashion | hmm get someone with the same version of asterisk, compile it, and then copy their /usr/lib/asterisk/modules/res_monitor.so to yours |
| 19:30.57 | jfitzgibbon | smace: transcode from wav to mp3 after the fact with whatever tool you're comfortable using |
| 19:31.17 | sashion | hence lookup MONITOR_EXEC in voip-info.org |
| 19:31.38 | sashion | will give you some tips about how to run commands against your recordings after the channel ends |
| 19:32.01 | *** join/#asterisk heartones (n=heartone@196.218.34.207) |
| 19:32.25 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:32.30 | smace | thank you, it is helpful :) |
| 19:32.43 | sashion | smace: No problem |
| 19:35.24 | *** join/#asterisk Al_Berto (i=Al_Berto@bandsal.at) |
| 19:35.52 | Al_Berto | hi! can i stop asterisk (1.2) from forwarding dtmf signals? |
| 19:37.25 | Al_Berto | i'd like asterisk to just process dtmf-signals according to features.conf, without retransmitting them to other peers |
| 19:40.04 | *** join/#asterisk webtech_m33 (i=webtech-@webtech.m33access.com) |
| 19:41.19 | webtech_m33 | anyone know of a good web based program to interface with asterisk? |
| 19:42.02 | webtech_m33 | i am running a ubuntu 7.04 server |
| 19:45.52 | *** part/#asterisk pjz (n=pj@66.219.59.183) |
| 19:46.38 | tomcontr3 | I configured my TDM400P but now Im getting http://pastebin.ca/692492 |
| 19:47.08 | sashion | webtech_m33: asterisk_gui or checkout freepbx |
| 19:49.50 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
| 19:50.23 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 19:51.03 | tomcontr3 | does anyone knows this? |
| 19:51.07 | tomcontr3 | #/sbin/ztcfg -vvvv |
| 19:52.13 | *** join/#asterisk Luch0 (n=lucho@host121.201-253-167.telecom.net.ar) |
| 19:52.19 | deeperror | lsmod | grep zt |
| 19:52.41 | tomcontr3 | ztdummy 7944 0 |
| 19:52.42 | tomcontr3 | zttranscode 12424 0 |
| 19:52.42 | tomcontr3 | zaptel 184228 3 ztdummy,zttranscode,wctdm |
| 19:53.25 | deeperror | zaptel.conf & zapata.conf been setup? |
| 19:53.30 | webtech_m33 | so FreePBX is put on top of asterisk? |
| 19:53.44 | tomcontr3 | yep |
| 19:54.24 | [TK]D-Fender | tomcontr3: pastebin dmesg, I have a rather strong suspicion... |
| 19:55.57 | tomcontr3 | http://pastebin.ca/692504 |
| 19:55.59 | deeperror | how would i find out which revision is used in a final release? |
| 19:57.43 | Nugget | man, I really need to just start from scratch and make a clean dialplan. Three years of cruft and evolving asterisk functions has left this one pretty messy. |
| 19:58.25 | Trevor_B|Away | webtech_m33: Yes, freepbx is a LAMP stack that uses the data to write the asterisk config and reload when required. #freepbx should be able to field answers on it. |
| 19:59.28 | tomcontr3 | any idea <[TK]D-Fender? |
| 20:01.15 | [TK]D-Fender | tomcontr3: modprobe your card, then pastebin "ztcfg -vvvv" , then try starting * |
| 20:01.34 | tzafrir_laptop | tomcontr3, what is the output of: lszaptel # or cat /proc/zaptel/* |
| 20:02.01 | [TK]D-Fender | deeperror: Which version of what? Zaptel? |
| 20:02.55 | deeperror | [TK]D-Fender: yes |
| 20:02.56 | tomcontr3 | lszaptel: http://pastebin.ca/692513 |
| 20:03.31 | tzafrir_laptop | tomcontr3, this is after running ztcfg? |
| 20:03.34 | tomcontr3 | <[TK]D-Fender: I have already done modprobe wctdm |
| 20:03.40 | tomcontr3 | right |
| 20:03.45 | tzafrir_laptop | if so, your /etc/zaptel.conf is empty or something |
| 20:04.24 | tomcontr3 | in zaptel.conf I have fxoks=1-2 loadzone = cl defaultzone=cl |
| 20:04.30 | tomcontr3 | in different lines ofcourse |
| 20:05.21 | tzafrir_laptop | what is the output of ztcfg ? |
| 20:05.49 | tomcontr3 | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) ......... |
| 20:06.05 | tzafrir_laptop | please use genzaptelconf |
| 20:06.18 | tzafrir_laptop | xpp/utils/genzaptelconf in the zaptel dir |
| 20:06.27 | jfitzgibbon | tomcontr3: you have FXO modules, so you need FXS signaling, right? |
| 20:06.42 | tzafrir_laptop | maybe you got wrong module numbers or wrong types or something |
| 20:06.54 | tomcontr3 | I have olny 2 FXO modules |
| 20:07.07 | tzafrir_laptop | fxsks=1-2, then |
| 20:07.28 | tzafrir_laptop | but then again, if you used genzaptelconf you wouldn't have needed to remember this |
| 20:07.35 | tomcontr3 | so I delete fxoks=1-2 ? |
| 20:08.01 | deeperror | it seems backwards but is correct |
| 20:08.07 | tzafrir_laptop | replace fxoks with fxsks |
| 20:08.20 | tzafrir_laptop | or run genzaptelconf |
| 20:08.43 | tomcontr3 | ok ztcfg said nothing |
| 20:09.08 | tomcontr3 | but now Im getting this http://pastebin.ca/692524 |
| 20:09.50 | deeperror | probably need to make the same update in zapata |
| 20:10.19 | tomcontr3 | signalling=fxs_ks |
| 20:10.24 | tomcontr3 | that was the problem |
| 20:10.33 | tomcontr3 | I had fxo_ks instead |
| 20:13.21 | tomcontr3 | thanks guys |
| 20:15.29 | los415 | does anyone have sample configs of getting asterisk to talk to a nextone via sip |
| 20:15.46 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@015-802-211.area5.spcsdns.net) |
| 20:16.05 | deeperror | how should one go about asking a question related to specific lines of code? |
| 20:18.28 | tomcontr3 | I pluged my PSTN line to the TDM400, but when I call in... it says NoOp("Zap/1-1", "No DID or CID Match") |
| 20:19.25 | ManxPower | tomcontr3: sounds to me like you are using some GUI |
| 20:20.00 | ManxPower | Before you answer: The last someone admitted to using a GUI here, we tossed their body in the canal for the 'gators. |
| 20:20.18 | ManxPower | There are at least 3 other channels for GUI stuff. |
| 20:20.42 | ManxPower | deeperror: try #asterisk-dev if the lines of code are from Asterisk |
| 20:21.43 | tomcontr3 | but how to I make my TDM400 Card to check for the CallerID |
| 20:22.36 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 20:24.38 | jablko | i have a zaptel fxo interface |
| 20:24.39 | tomcontr3 | ?? |
| 20:24.56 | jablko | my dial plan is configured so as soon is it rings, asterisk calls a sip interface |
| 20:25.03 | jablko | with timeout 20seconds |
| 20:25.12 | ManxPower | tomcontr3: do you have callerid enabled in zapata.conf? |
| 20:25.12 | jablko | without actually answering the zaptel interface |
| 20:25.37 | ManxPower | Also put in a Noop(CALLERID(all) is ${CALLERID(all)}) to see what you are getting |
| 20:25.49 | ManxPower | jablko: astrisk will not answer unless you tell it to. |
| 20:25.54 | jablko | this gives people 20 seconds to answer the zaptel interface by picking up an extension |
| 20:26.04 | jablko | ManxPower: right, this is the intention |
| 20:26.05 | ManxPower | Specific APPLICATIONS will answer, check the docs for the app |
| 20:26.06 | jablko | problem: |
| 20:26.43 | jablko | if the SIP interface is not available, asterisk answers the call immediately (goes to voicemail) |
| 20:26.55 | ManxPower | jablko: then don't do that. |
| 20:27.06 | jablko | this doesn't give people 20 seconds to pickup an extension on the PSTN |
| 20:27.12 | ManxPower | Voicemail has to answer, if you don't want it to answer, then don't run Voicemail when the Dial ends |
| 20:27.30 | ManxPower | jablko: Uh, if the SIP device is not available, you can't use it. |
| 20:27.56 | [TK]D-Fender | jablko: If no channel is left available to ring its not going to sit around doing NOTHINIG for those 20 sec you know... |
| 20:28.06 | jablko | how can i make asterisk dial the SIP interface immediately, but wait 20seconds before ansering the call and going to voicemail, regardless of whether the SIP interface is avaialble? |
| 20:28.30 | ManxPower | jablko: you check the value of DIALSTATUS and decide if you want to run Voicemail or run Wait(20) |
| 20:28.37 | jablko | i think i need to combing Dial(...,20) and Wait(20) somehow... |
| 20:28.45 | jablko | ManxPower: ah |
| 20:28.52 | ManxPower | jablko: Asterisk is not designed and will never support picking up lines OUTSIDE of Asterisk. |
| 20:28.57 | [TK]D-Fender | jablko: NO |
| 20:29.38 | [TK]D-Fender | ok, heading out, BBIAB |
| 20:29.39 | ManxPower | chances are DIALSTATUS will be CHANUNAVAIL if the SIP device is not reackable |
| 20:29.53 | Qwell | reackable...interesting typo |
| 20:30.31 | jablko | (eventually we don't want picking up the line OUTSIDE asterisk, but we're currently limited by hardware - making a slow transition from analog system to asterisk) |
| 20:30.31 | Nugget | USERSTATUS is DIDNTREADTHEDOCS. |
| 20:30.35 | ManxPower | If it is busy, then DIALSTATUS will be BUSY, if it does not answer then it will be NOANSWER. All this is documented in "show application dial" and examples of doing stuff with DIALSTATUS is in macro-std-exten in the extensions.conf.sample |
| 20:30.54 | *** join/#asterisk s1gny|wrk (n=s1gny@91.64.105.151) |
| 20:31.01 | jablko | ManxPower: can do, thanks |
| 20:31.06 | *** part/#asterisk s1gny|wrk (n=s1gny@91.64.105.151) |
| 20:31.17 | ManxPower | jablko: unfortunatly "slow transitions" when it comes to PBXs are seldom slow, nor transitions. Usually they are causes of heart attacks, ulcers, and unemployment |
| 20:31.18 | jablko | couldn't figure out what was the "right" way to make this work... |
| 20:31.37 | jablko | checking the DIALSTATUS works, thanks |
| 20:36.31 | *** join/#asterisk potsboy (n=chrisg@vc-196-207-32-228.3g.vodacom.co.za) |
| 20:41.45 | tomcontr3 | how can I detect when the line is beeing used by someone else, so I dont interrupt the call when Im using 2 FXO modules |
| 20:41.46 | tomcontr3 | ? |
| 20:42.09 | deeperror | cli> zap show channels |
| 20:42.12 | jfitzgibbon | tomcontr3: you won't interrupt the call. try it |
| 20:43.40 | webtech_m33 | anyone know what the ubuntu package for perl-CPAN |
| 20:43.43 | webtech_m33 | is? |
| 20:44.04 | jfitzgibbon | tomcontr3: in a strangely apropos moment, the answer is to check ${DIALSTATUS} |
| 20:46.47 | Poehali | I have the right "secret" but SPA3102 still fails to register to asterisk |
| 20:50.45 | Poehali | is there a iptables rule I need to set on asterisk? |
| 20:51.41 | tomcontr3 | it did |
| 20:51.46 | tomcontr3 | thats why im asking |
| 20:55.14 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:56.18 | tomcontr3 | should I configure something for that in the zapata.conf? |
| 20:56.19 | *** join/#asterisk smace (n=chatzill@200.220.198.107) |
| 20:58.52 | *** part/#asterisk potsboy (n=chrisg@vc-196-207-32-228.3g.vodacom.co.za) |
| 21:00.30 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:02.45 | smace | I could not understand till now how to forward calls. I am using SPA-3102 and Asterisk. But I am not sure about if I should forward in the ATA or in Asterisk. |
| 21:04.22 | smace | I would like when someone calls to the Public Line this call to be redirected to one Ramal. I could already record the call. This way I would have the call redirected and recorded. |
| 21:05.43 | [TK]D-Fender | smace, "one Ramal." huh? |
| 21:06.20 | smace | consider it, one termination. one Telephone. I thought it was ENglish. Sorry. |
| 21:07.59 | [TK]D-Fender | smace, Still not making any sense |
| 21:09.22 | watchy | anyone here run 3 displays + on a windows box? |
| 21:09.39 | Qwell | I don't think anybody here runs windows |
| 21:09.44 | watchy | i do :/ |
| 21:09.57 | watchy | linux isn't a very good desktop os |
| 21:10.18 | smace | [TK]D-Fender: Just a sec. I'll write with diferent words. :) |
| 21:10.33 | [TK]D-Fender | watchy, I have a friend who runs 4 |
| 21:10.47 | [TK]D-Fender | smace, Oh, and try not to pick them at random this time :p |
| 21:11.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:11.33 | watchy | tk: my friend is having trouble with running 2 PCI-e video cards |
| 21:12.01 | [TK]D-Fender | watchy, Not sure how that works outside SLI mode. |
| 21:12.31 | [TK]D-Fender | Then again... I'm on ONBOARD video now on all my systems :) |
| 21:13.58 | watchy | yea apparently not well |
| 21:14.08 | watchy | his 8800GTX and a 6600 won't work for some reason |
| 21:14.38 | [TK]D-Fender | watchy, which one fails? |
| 21:14.46 | *** join/#asterisk |omni| (n=rob@c-67-185-70-220.hsd1.wa.comcast.net) |
| 21:15.22 | watchy | the 6600 never starts i guess windows never talks to it |
| 21:15.34 | watchy | but if he removes his 8800 and puts the 6600 in the main slot it works |
| 21:16.17 | [TK]D-Fender | watchy, and if he simply inverts the cards? |
| 21:16.47 | watchy | no work |
| 21:16.52 | watchy | brb gotta take a shower |
| 21:16.53 | [TK]D-Fender | watchy, NONE? |
| 21:16.54 | watchy | http://forums.nvidia.com/lofiversion/index.php?t31779.html |
| 21:16.59 | watchy | thats some good info about it |
| 21:17.12 | watchy | whichever cards in the main slot boots |
| 21:17.13 | watchy | brb |
| 21:17.50 | [TK]D-Fender | watchy, oh well, that post really says it all. |
| 21:20.26 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
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| 21:22.42 | Poehali | anyone here configured a SPA3102 here? I'm really stuck now |
| 21:24.54 | *** join/#asterisk webavant (n=freenode@c-76-27-193-53.hsd1.or.comcast.net) |
| 21:25.03 | smace | I would like to forward all calls coming from PSTN (SPA-3102 -> FXO) to one remote voip/sip phone (SPA-2000 -> FXS). All calls will be recorded when are being forwarded (it is already working). Asterisk is working fine for recording and calling (when dialing numbers manually). I do not know where I should do the forward in the SPA-3102 and Asterisk. I have tried some setup but did not work... |
| 21:25.05 | smace | ...as expected, and I am feeling a little lost now. Does it sound clear now? :) |
| 21:26.14 | *** join/#asterisk Yourname`` (n=IM@unaffiliated/yourname/x-837320) |
| 21:26.27 | webavant | any of you use GrandCentral.com with Skype? When I tell GrandCentral to call my Skype for recording voice mails, it does not detect my key-presses, although I can hear the tone during the recording playback... anyone know how to configure Skype to properly generate the tones or something? |
| 21:26.30 | deeperror | smace: do you answer your incoming context? |
| 21:26.53 | Yourname`` | Hi, so while we're doing some dialing. I change the sip.conf to go through another provider. And when I do sip reload on the CLI, it'll automatically switch to the changed provider, correct? Or will I need to restart? |
| 21:27.17 | Poehali | so basically SPA3102 doesn't work? |
| 21:28.01 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 21:28.21 | Winkie | hey guys, i'm getting quite big times reported from rtp.c, can someone give me a hand understanding this? |
| 21:28.35 | smace | I cant manage to have it working properly. It works, but not as I expected. THe point is where I should do the forward. Dialing numbers manually works fine, but I would like to have automatically forwarded. |
| 21:29.10 | Poehali | oh please help me |
| 21:29.19 | Poehali | I can't even get it to work as not as I expected |
| 21:30.44 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:30.57 | smace | Poehali: What do you wanna do? |
| 21:31.36 | ido | what's the preferred SIP/voip provider for small businesses looking to take incoming 800 calls? i'm looking into phone support options for a company i'm starting and i would love any advice from you seasoned veterans. :) |
| 21:31.46 | Poehali | so what's the proper way to do it? |
| 21:32.45 | Poehali | I want to get sipura to talk to asterisk |
| 21:32.45 | Strom_M | ido: i like teliax |
| 21:33.47 | ido | Strom_M: two issues come to mind: if i don't like them, how easy is it to port/keep the same number when moving providers (for 800 numbers)? and, IAX vs. SIP -- for multiple simultaneous incoming calls, any differences worthy of mention? |
| 21:34.14 | Strom_M | ido: 800 numbers are universally portable, usually |
| 21:34.23 | Strom_M | and teliax uses iax and sip; i like both. |
| 21:34.26 | ido | Strom_M: do you have experience with 800 numbers? |
| 21:34.35 | *** join/#asterisk badcfe (i=christia@irc.slengpung.com) |
| 21:34.36 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 21:34.56 | *** join/#asterisk mohsen (n=chatzill@82.99.234.205) |
| 21:35.14 | Strom_M | ido: yes |
| 21:35.25 | Trionnis | can someone help with getting * to allow unregistered/unauthenticated calls to be sent out an IAX channel? |
| 21:35.35 | mohsen | mmm, how can I check the status of a sippeer in real time? SipPeer doc says it does not function with real time. |
| 21:35.55 | Trionnis | I have allowguest=yes in sip.conf, and it's still throwing 407 when a call is placed |
| 21:35.56 | ido | fantastic. i am completely new to the 800 number deal. would you be interested in compensation in exchange for a little hand-holding sometime next month? |
| 21:37.13 | Strom_M | sure |
| 21:37.32 | ido | sweet. privmsging you shortly. |
| 21:44.51 | *** join/#asterisk tc3driver (n=huh@rrcs-67-52-113-254.west.biz.rr.com) |
| 21:46.50 | *** join/#asterisk Stormfr (n=Stormfr@sgc91-2-82-237-76-2.fbx.proxad.net) |
| 21:47.52 | webavant | hmmm... I thought grandcentral could add custom ringbacks |
| 21:49.04 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584865.dsl.bell.ca) |
| 21:52.40 | *** join/#asterisk Stormfr (n=Stormfr@sgc91-2-82-237-76-2.fbx.proxad.net) |
| 21:53.32 | the_Goat | ichung%r4 |
| 21:58.18 | smace | I need some tip to forward calls. I am lost here. SPA-3102 manual is not saying much clearly how-to. |
| 22:16.03 | *** join/#asterisk Lonie (n=lonie@dslb-088-074-224-007.pools.arcor-ip.net) |
| 22:20.27 | SA007 | anyone know how to convince asterisk to translate sip info calls to dtmf tones? |
| 22:23.58 | _ShrikE | good afternoon folks |
| 22:24.31 | SA007 | afternoon? its almost half past midnight over here |
| 22:24.42 | Qwell | mmm, cow nuggets |
| 22:24.47 | Qwell | PATENT PENDING |
| 22:24.48 | Nugget | ]:8) |
| 22:26.12 | Toerkeium | hello guys. If I initiate a call from AST mananger from (lets say) PHP.. is there any way to monitor if the call was answered? would anyone point me in the right direction? I am pretty lost here |
| 22:26.33 | Qwell | you could watch the manager events... |
| 22:26.37 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:27.35 | Toerkeium | Qwell.. and how can I identify each call? I mean, does the AST manager return any ID once I initiate a call? |
| 22:29.12 | SA007 | old one, but funny ;) http://home.hetnet.nl/~carthago/fun/ft031003.gif |
| 22:29.27 | Toerkeium | I ask this because lets say I initiate 3 calls at the same time from a web page, and based on if the call is answered I need to return some text to the web caller |
| 22:31.30 | *** join/#asterisk _10nix_ (n=hyjnx@user-160u96o.cable.mindspring.com) |
| 22:33.36 | _10nix_ | hello, im having a bit of a problem with the voicemail module, i was wondering if anyone might offer a suggestion |
| 22:34.03 | _10nix_ | when i call into the voicemail main, it automatically playsback goodbye |
| 22:35.25 | _10nix_ | im calling in on a SIP extension |
| 22:37.13 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:39.36 | *** join/#asterisk the_goat (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net) |
| 22:39.40 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 22:40.37 | the_goat | anyone here handy with call parking |
| 22:40.39 | the_goat | ? |
| 22:42.36 | the_goat | anyone here? |
| 22:43.58 | RipeR-81 | the_goat seems everyone is pretty busy |
| 22:44.07 | RipeR-81 | i also need help on outgoing calls |
| 22:44.13 | JT | busy or not here |
| 22:44.17 | JT | or unable to help |
| 22:44.27 | RipeR-81 | JT or newbie like us |
| 22:44.27 | RipeR-81 | :D |
| 22:52.52 | the_goat | ok, since i am finished eating, i can ask my ;-) |
| 22:52.54 | the_goat | ? |
| 22:53.12 | *** join/#asterisk sivana (n=sivana@gromit.mixdown.ca) |
| 22:53.29 | sivana | tzanger: ping |
| 22:54.22 | the_goat | ok, i am having an issue with call parking. when someone calls in, and i put them on park, i go to the other phone and dial the parked extension. the issue i am having, i can't hear the caller from the phone i just picked up the parked extension on. they can hear me just fine though |
| 22:55.17 | RipeR-81 | ? |
| 22:56.55 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-38-118.lns3.syd7.internode.on.net) |
| 22:59.58 | the_goat | yeah....when i pickup the parked extension....ie go to the phone and dial 702 or what ever. when i talk on that phone the can hear me, but i can't hear them |
| 23:04.31 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 23:10.29 | *** join/#asterisk ssokol (n=ssokol@216.64.24.250) |
| 23:13.16 | *** join/#asterisk fmueller (n=user@p548F67B7.dip.t-dialin.net) |
| 23:17.57 | the_goat | hey ripe, iforgot to ask what your ? was |
| 23:19.06 | Qwell | ssokol: hey |
| 23:21.13 | the_goat | hi quell, are you handy with call parking? |
| 23:22.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:23.24 | *** join/#asterisk shido6 (n=shido6@74-130-125-252.dhcp.insightbb.com) |
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| 23:34.30 | RipeR-81 | the_goat u forgot to aske me??? |
| 23:35.06 | RipeR-81 | ask* me that is .. mispelling is a killer |
| 23:40.32 | *** join/#asterisk heartones (n=heartone@196.218.34.207) |
| 23:41.08 | Poehali | I figured out how to read asterisk logs now |
| 23:41.17 | the_goat | you said you had a question about outbound calls |
| 23:41.25 | Poehali | it appears the actual error for me not being to log in is "device does not match ACL" |
| 23:44.50 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:47.39 | RipeR-81 | the_goat yeah.. im trying to configure to call thru a sip server |
| 23:47.45 | RipeR-81 | aka touchstar |
| 23:47.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:48.28 | RipeR-81 | the_goat but right now im with tech support from them trying to set asterisk so we can make outbound calls thru them |
| 23:48.56 | RipeR-81 | the nortel pbx i have here it seems so outdated |
| 23:49.24 | the_goat | do you have your peers defined in sip.conf |
| 23:50.07 | Toerkeium | people, I am looking events and I see a Uniqueid which is useful to me, if I could some way send within the originate action some other or same uniqueid... is this possible? |
| 23:50.17 | RipeR-81 | the_goat yep i guess the problem was on the dialing rules |
| 23:50.18 | RipeR-81 | :D |
| 23:51.10 | the_goat | exten => _1NXX-NXX-XXXX,2,Dial(SIP/sipprovider/${EXTEN}) |
| 23:51.26 | the_goat | i use this in my dialplan. i dial one and the number and it dials out |
| 23:52.02 | Toerkeium | anyone? any idea? |
| 23:53.10 | the_goat | sorry toerkeium. i am fairly new to asterisk so i am still learning |
| 23:53.19 | the_goat | do you know anything about call parking |
| 23:54.13 | TJNII | Toerkeium: I think you can do that in the dialplan, but I haven't played with it. |
| 23:54.32 | TJNII | A friend was taking about agi scripts to do that. |
| 23:55.01 | Toerkeium | I am using the AMI with php |
| 23:55.05 | RipeR-81 | the_goat im using exten => _NXXNXXXXXX,1,Dial(SIP/sipprovider/${EXTEN}) |
| 23:55.41 | Poehali | the_goat: yes I have peers defined in sip.conf |
| 23:56.07 | Poehali | the_goat: I can connect using portsip but not with the sipura device, same login |
| 23:56.37 | Toerkeium | I am trying something with "ActionID" but it doesn't appear in the event responses |
| 23:56.48 | the_goat | what happens when you try to connect with the sipura device |
| 23:56.56 | Poehali | the_goat: failed |
| 23:57.25 | Toerkeium | I think that if I make it appears I could just do this setting up the ActionID with a timestamp |
| 23:58.15 | Poehali | the_goat: [C[Sep 11 15:51:18] NOTICE[3024] chan_sip.c: Registration from <sip:john@192.168.1.7>' failed for '192.168.1.205' - Device does not match ACL |
| 23:59.07 | *** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:59.52 | Toerkeium | hmm if I telnet to asterisk it shows the actionID in the response, but not when I request "Action: events" from the AMI |
| 23:59.52 | Nugget | telnet is eeeeeeevil! |
