| 00:07.43 | raden_work | how can i make it so i can dial *67 and stuff like that ? |
| 00:07.46 | raden_work | it dont work |
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| 00:14.35 | aurax | anyone experienced kernel panics with dahdi ? |
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| 00:16.20 | Downchuck | I'm using the Pickup() command, how do I instruct it to continue with the dial plan once the caller has hungup? |
| 00:17.31 | Downchuck | pickup sure is a weird one |
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| 00:53.43 | svm_invictvs- | raden_work: So yeah. It works with IAX2. I guess SIP is just weird. |
| 00:54.02 | svm_invictvs- | Still can't receive calls though. |
| 00:54.59 | svm_invictvs- | I'm getting this error: "Rejected connect attempt does not exist from xxx.xxx.xxx.xxx request xxxxxxxxxx@main does not exist." |
| 00:55.01 | aurax | anyone experienced kernel panics with dahdi ? |
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| 00:57.09 | svm_invictvs- | No... |
| 00:57.15 | svm_invictvs- | Probably a buggy driver. |
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| 01:00.46 | raden_work | svm_invictvs, u can call out IAX ? |
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| 01:16.36 | raden_work | exten => s,1,Answer() is 's' a wildcard ? |
| 01:17.44 | [TK]D-Fender | no |
| 01:17.55 | [TK]D-Fender | ~stdextens |
| 01:17.56 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 01:19.34 | [TK]D-Fender | [20:54]<svm_invictvs->I'm getting this error: "Rejected connect attempt does not exist from xxx.xxx.xxx.xxx request xxxxxxxxxx@main does not exist." <--- you should be looking at "xxxxxxxxxx@main " clearly. |
| 01:20.09 | eppigy | how dare you |
| 01:20.55 | [TK]D-Fender | eppigy: Might makes right, and I am very VERY right :D |
| 01:21.18 | raden_work | [TK]D-Fender, its all IP no FXO or FXS is that a standard way to recieve calls ? |
| 01:21.42 | [TK]D-Fender | raden_work: read that infobot bit AGAIN. |
| 01:22.51 | raden_work | that makes sense now sorry for the idiocy |
| 01:23.16 | [TK]D-Fender | raden_work: Its surprising how much usable info I can cram into a single line of text :p |
| 01:23.25 | raden_work | rember before when i could not get a inbound call i could just make the phone ring ? |
| 01:23.30 | raden_work | [TK]D-Fender, yes i have to agree |
| 01:23.50 | raden_work | session-timers=refuse |
| 01:23.50 | raden_work | session-expires=180 |
| 01:23.51 | raden_work | session-minse=90 |
| 01:23.51 | raden_work | session-refresher=uas |
| 01:23.52 | [TK]D-Fender | raden_work: Don't remember so much... |
| 01:24.12 | [TK]D-Fender | raden_work: Those are new parms to me... |
| 01:24.26 | raden_work | well tech got back to me with that threw in sip.conf and walla |
| 01:24.48 | raden_work | it works i cant find them anwhere in my book i dont know what they do |
| 01:25.41 | raden_work | just dont know how sessions timers and expiration could keep me from answering a phone ? |
| 01:25.54 | [TK]D-Fender | raden_work: FUBAR'd provider |
| 01:26.03 | raden_work | callcentric.com |
| 01:26.06 | [TK]D-Fender | raden_work: NO-ONE I've ever heard of requires anything like it |
| 01:26.10 | raden_work | i was thinking of calling teliax tommorow |
| 01:26.48 | raden_work | how do i go about cleaning up echo on the line using ulaw on Aastra 9133i phones |
| 01:29.15 | Kobaz | raden_work: echo? on an ip phone? |
| 01:29.20 | Kobaz | raden_work: what's the other endpoint? |
| 01:30.04 | raden_work | call phone <-> ITSP <->asterisk <-> IP Phone and 56 ms laterncy |
| 01:30.11 | eppigy | RAPIDO |
| 01:30.12 | raden_work | call = cell |
| 01:30.21 | eppigy | u a cell |
| 01:30.33 | [TK]D-Fender | raden_work: do you get it from phone to phone locally? |
| 01:30.56 | svm_invictvs- | [TK]D-Fender: http://www.pastebin.ca/1518620 |
| 01:30.57 | Kobaz | raden_work: do you get echo on a different phone off the same box? |
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| 01:31.37 | [TK]D-Fender | svm_invictvs-: And? |
| 01:31.42 | raden_work | [TK]D-Fender, no phone to phone echo |
| 01:32.03 | raden_work | does asterisk come with any sample gsm files ? |
| 01:32.04 | svm_invictvs- | [TK]D-Fender: That's what's in that context. |
| 01:32.05 | [TK]D-Fender | raden_work: then its your provider's fault and there is nothing you can do |
| 01:32.21 | raden_work | wow this provider really starting to suck :D |
| 01:32.26 | [TK]D-Fender | svm_invictvs-: I don't see 10-digit capable pattern in there anywhere.... |
| 01:32.48 | svm_invictvs- | [TK]D-Fender: DOens't it just fall through to the "s" extension? |
| 01:33.21 | [TK]D-Fender | svm_invictvs-: No, the call is coming in TARGETING a specific extension, so it'd better match. Learn from the lesson I jsut gave raden_work |
| 01:34.18 | svm_invictvs- | [TK]D-Fender: So i need to have it go to that specific extension. |
| 01:34.55 | [TK]D-Fender | svm_invictvs-: you should have something that will match the call... |
| 01:35.35 | svm_invictvs- | something like exten => _XXXXXXXXXX,n,Answer() |
| 01:35.37 | Kobaz | raden_work: do you get echo on a different phone off the same box? talking to the cell phone? |
| 01:35.52 | svm_invictvs- | Or should I just put the actual phone number in there? |
| 01:36.21 | raden_work | Kobaz, nope |
| 01:36.40 | raden_work | i get echo to anyone that i call it on there side i dont hear it at all sorry was not clear on that |
| 01:36.49 | raden_work | whats asterisk default sound directory or where do i set it ? |
| 01:37.05 | [TK]D-Fender | svm_invictvsYou should receive your calls with as specific an exten as you can in one context and then GOTO another to start an IVR, etc |
| 01:37.06 | svm_invictvs- | /var/lib/asterisk/sounds, I think |
| 01:37.17 | [TK]D-Fender | raden_work: varlib in asterisk.conf |
| 01:37.18 | svm_invictvs- | [TK]D-Fender: Ah, gotcha. |
| 01:38.30 | raden_work | can i set all conversations to be recorded somehow ? |
| 01:40.32 | ManxPower | raden_work: See the applications Monitor and MixMonitor |
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| 01:42.15 | ManxPower | svm_invictvs-: "s" means "I don't have a destination number". It does not "match any number". It does the exact opposite of "match any number" |
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| 01:43.07 | svm_invictvs- | ManxPower: I see. |
| 01:43.19 | raden_work | ManxPower, ? |
| 01:43.35 | raden_work | [TK]D-Fender, why will asterisk pickup on _X. but not s ? |
| 01:45.15 | raden_work | i have to configure everything with _X. to work |
| 01:45.19 | raden_work | is that normal ? |
| 01:46.08 | svm_invictvs- | Howabout this: http://pastebin.ca/1518631 |
| 01:46.46 | [TK]D-Fender | svm_invictvs-: much better |
| 01:46.58 | svm_invictvs- | Let me try calling it. |
| 01:47.01 | [TK]D-Fender | raden_work: "core show application monitor" |
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| 01:52.05 | svm_invictvs- | SO the "o" extension. Is that called when somebody just presses the 0 button? |
| 01:52.27 | svm_invictvs- | [TK]D-Fender: Hey, it works now. Help is much appreciated. |
| 01:52.39 | [TK]D-Fender | svm_invictvs-: SORT of |
| 01:52.54 | [TK]D-Fender | svm_invictvsYou need to keep conditions in mind |
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| 01:54.25 | ManxPower | raden_work: the only time you EVER use "s" is with Zaptel/DAHDI and immediate=yes, macros, and FXO signaled ports. |
| 01:54.45 | svm_invictvs- | [TK]D-Fender: What conditions? |
| 01:54.46 | [TK]D-Fender | ManxPower: And IVR |
| 01:55.16 | [TK]D-Fender | svm_invictvs-: "o" is for when someone hits "0"...... while listening toa VOICEMAIL greeting <- |
| 01:55.26 | svm_invictvs- | oh |
| 01:55.42 | [TK]D-Fender | svm_invictvsotherwise for normal IVR's, etc, its just "0' |
| 01:56.31 | raden_work | ManxPower, thanks :) |
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| 01:58.26 | svm_invictvs- | [TK]D-Fender: Yeah, I'm not using voicemail at all. Basically my PBX just forwards to cell phones and then use that voicemail. |
| 02:00.21 | raden_work | i do have to record all menu items as gsm files ? |
| 02:00.34 | raden_work | or can asterisk do txt to speech ? |
| 02:01.07 | [TK]D-Fender | raden_work: Files can be in any format * can read. |
| 02:01.16 | [TK]D-Fender | raden_work: * doesn't do TTS. Other apps do |
| 02:01.30 | [TK]D-Fender | raden_work: "festival" or "cepstral" are common |
| 02:02.18 | jaytee | cepstral works well |
| 02:04.37 | raden_work | ill just have to find a chick with a nice voice :D |
| 02:06.06 | raden_work | is there alot to setting up VM in * |
| 02:08.46 | KavanS | raden_work, that's a good idea...make sure you stay in touch with her, because you will always find yourself needing additional prompts |
| 02:09.17 | raden_work | KavanS, lol :) |
| 02:09.33 | KavanS | lol just saying, it's true |
| 02:09.36 | raden_work | how can i run kate as SU ? |
| 02:09.41 | KavanS | if you think you will tinker with it |
| 02:09.44 | raden_work | KavanS, hehehe |
| 02:09.49 | KavanS | allison is cool, but everyone has that same voice |
| 02:14.08 | svm_invictvs- | What's a good mac softphone? |
| 02:14.29 | leifmadsen | I like zoiper |
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| 02:30.23 | raden_work | thanks for everything im out of here |
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| 02:38.56 | svm_invictvs- | What do the various voicemail options do? |
| 02:39.17 | svm_invictvs- | I see, tz, tld, attach, etc. They're mentioned in the Future of Telephony book, but the author doesn't clearly explain what each one does. |
| 02:40.09 | [TK]D-Fender | svm_invictvsread the sample configs |
| 02:54.42 | svm_invictvs- | hm |
| 02:54.47 | svm_invictvs- | WHy isn't email working. bizarre |
| 02:57.15 | svm_invictvs- | I bet it's an anti-spam thing |
| 02:59.41 | svm_invictvs- | d'oh sendmail |
| 02:59.52 | svm_invictvs- | [TK]D-Fender: Once again, thanks for hte help man. |
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| 03:55.02 | missnebun | hi guys ... if I want to apply a patch how I do this ... I have U linux/branches/2.2/drivers/dahdi/dahdi_dummy.c ??? |
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| 04:38.59 | Nugget | http://xkcd.com/619/ <-- Truth. |
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| 04:39.48 | [TK]D-Fender | telnet |
| 04:39.53 | [TK]D-Fender | :( |
| 04:40.05 | [TK]D-Fender | BORING |
| 04:40.19 | [TK]D-Fender | Nugget: DANCE MONKEY, DANCE! |
| 04:40.24 | Nugget | doo doo doo |
| 04:40.35 | Nugget | \o/ ^o^ /o_ /o\ |
| 04:40.40 | Nugget | (it's fun to stay at the) |
| 04:40.58 | [TK]D-Fender | ... |
| 04:41.03 | [TK]D-Fender | flees in terror |
| 04:41.15 | vousb | hehe |
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| 04:48.42 | box2 | lol |
| 04:48.45 | vousb | oops |
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| 05:39.37 | Micc_ | I have a bunch of polycom ip450's and some of them have good speaker phone and some have problems speaking while the other end is speaking too. |
| 05:39.51 | Micc_ | It cuts out the audio if there is a lot of audio being sent. |
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| 05:40.16 | Micc_ | Both have the same configuration .cfg files on the server except for the asterisk sip account. |
| 05:40.29 | Micc_ | But the sip.conf entries for both users are the same. |
| 05:40.37 | Micc_ | What could cause that? |
| 05:40.57 | Micc_ | I was thinking maybe silence suppression, but its not really anything like silence suppression. |
| 05:41.07 | Micc_ | Its more like its only half-duplex speaker phone. |
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| 05:58.16 | box2 | banana phooooone |
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| 06:39.45 | Dovid | box2: Banana phone ???????????? |
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| 06:43.33 | svm_invictvs | So... |
| 06:43.53 | svm_invictvs | I have entries in my sip.conf for phones and devices. |
| 06:45.11 | svm_invictvs | I have an entry in my sip.conf file like this....http://pastebin.ca/1518840 |
| 06:45.22 | svm_invictvs | I'm having trouble getting linphone to connect to it, though. |
| 06:45.38 | svm_invictvs | It's asking for sip proxy identity, sip proxy, and route? |
| 06:45.49 | svm_invictvs | What am I missing here? |
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| 06:47.56 | jclacherty | is anyone able to help me understand how call setups between two ip phones work with asterisk? |
| 06:48.43 | jclacherty | I thought the way it worked was that the ip phones register to the asterisk server |
| 06:49.33 | jclacherty | and then one calls the other, the sip packets come from the server, but once the call is set up the rtp/rtcp packets were transferred directly from one phone to the other |
| 06:49.52 | jclacherty | what I'm seeing though is that the rtp/rtcp is to the server |
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| 06:58.48 | jclacherty | is anyone able to help out? or have I worded the question badly? |
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| 06:59.48 | jclacherty | it looks to me as if asterisk is acting as a rtp/rtcp proxy as well |
| 07:00.37 | svm_invictvs | jclacherty: I dont' know or else I'd help. |
| 07:02.23 | jclacherty | thanks, maybe I've picked the wrong channel :) |
| 07:02.47 | jclacherty | it's sometimes hard to tell if there's anyone actually there |
| 07:03.14 | jclacherty | svm_invictvs: |
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| 07:05.12 | svm_invictvs | hm |
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| 07:34.12 | networkstudent | on Centos I see: |
| 07:34.12 | networkstudent | checking for mandatory modules: NETSNMP... fail |
| 07:34.12 | networkstudent | all package installed |
| 07:34.19 | networkstudent | any help? |
| 07:37.16 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 07:48.47 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 07:51.37 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:51.37 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:56.46 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:06.20 | Hatrix | hello, I just got "yellow alert" on some of my dahdi channels, can someone explain to me what exactly a yellow alert is? a problem on the incoming side? or a problem with the card? |
| 08:06.38 | Hatrix | tried to google but did not find anything useful yet |
| 08:06.53 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:07.46 | maskas | Do I need to make install asterisk, before I can make asterisk-addons? |
| 08:08.40 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 08:17.03 | *** part/#asterisk networkstudent (n=S@78.129.237.130) |
| 08:28.29 | fors1 | Hi guys. I have two servers connected like this: external sip provider - SIP - asterisk server 1 - IAX - asterisk server 2 - phones .. When making calls from the phones to the "outside", I don't get any outgoing sound, only incoming. Doesn't matter if the call is outgoing or incoming. If I call directly from asterisk 1 to the outside, sound is fine. If I call from asterisk 1 to asterisk 2 (not to the SIP provider), sound is fine. So the problem appears |
| 08:29.35 | fors1 | Have anyone seen a similar behaviour before? I also get a: IAX2/peer-XXXX stopped sounds |
| 08:32.59 | Stese | fors1 > Why are you using two servers? |
| 08:33.04 | fors1 | security issues |
| 08:33.54 | *** join/#asterisk ceegee (n=christia@mail.cg-networks.de) |
| 08:34.01 | ceegee | hello there |
| 08:34.29 | fors1 | my company doesn't want a server connected both to the internal network and the external network, and SIP is not easy to nat (at least that is what I'm hearing). So we have one server in DMZ, and one server on internal, firewalled with only IAX port UDP 4569 open between them |
| 08:34.41 | ceegee | i have some trouble with attented or blind transfer after a call pickup with asterisk 1.4.26 and snom 320 phone with firmware 7.3.23 |
| 08:35.39 | ceegee | if i pickup a call and then want to forward it again to another phone i can not transfer the call by hanging up my phone |
| 08:35.46 | ceegee | the call still hangs on my line |
| 08:36.55 | *** join/#asterisk phr3ak (n=nnnnphr3@gnet.hu) |
| 08:38.02 | Stese | ceegee > are you using asking asterisk to transfer the call (default, dial #) or the snom? |
| 08:40.40 | ceegee | Stese: first I pickup the call with pushing a prgrammable key on my snom, the key is programmed with e.g. 21|*8, then I push another key which is programmed with *2 followed by typing in the destination number |
| 08:41.28 | ceegee | Stese: I think this should be the same like typing in *821 and *122 directly, nor? |
| 08:49.26 | Iain_ | Hi all |
| 08:50.11 | ceegee | Stese: any idea what could be wrong? |
| 08:51.04 | Iain_ | Caould someone tell me please if there is way for checking if a priority is valid in a similar way to i checking if an extension is valid |
| 08:54.46 | *** join/#asterisk Psychobilly (n=moi@adsl176-124.kln.forthnet.gr) |
| 08:54.56 | *** join/#asterisk cnu (n=cnu@63.80-203-44.nextgentel.com) |
| 08:56.19 | *** join/#asterisk Strogg (n=jean@unaffiliated/strogg) |
| 08:57.33 | *** join/#asterisk Tarantulafudge (i=46b240a8@gateway/web/freenode/x-dsvxhimadegrqelf) |
| 08:57.41 | Tarantulafudge | hey guys |
| 08:59.07 | Tarantulafudge | I need a few pointers, we are looking for a way to provide automatic voice notifications via the phone and our colo ISP suggested that we pay for some kind of trunking line and setup an asterisk server. Is this a good idea? |
| 08:59.27 | *** join/#asterisk Kumbang (n=kites@rusnas.paume.itb.ac.id) |
| 08:59.29 | Tarantulafudge | doesn't know much about asterisk |
| 09:04.47 | *** join/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com) |
| 09:06.00 | Tarantulafudge | Anyone have suggestions for setting up a notification system? |
| 09:06.47 | Tarantulafudge | is still at the drawing board phase |
| 09:07.03 | *** join/#asterisk fiddur (n=fiddur@192.121.104.122) |
| 09:09.39 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 09:15.15 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:25.21 | *** join/#asterisk jgoo (n=r3rman@ppp174-4.adsl.forthnet.gr) |
| 09:26.00 | jgoo | why is #asterisk +i? it is so stupid. anyway, grandstreams - how do I set them to use dialplans? I've seen people asking about dialplans, but cannot find anything in manual or interface (budgetone 201) |
| 09:26.18 | jgoo | echo - simple question, which I've NEVER seen address - where is it introduced into the system? |
| 09:27.22 | jgoo | ISDN - is it possible to have one ISDN line failover to another line if it is busy? An old siemens system supposedly has 3 incoming isdn, and from calling one number, you can get 6 lines up - of course I think this is impossible, but they swear to me... |
| 09:31.22 | Chainsaw | Yes, good entrance. You're already complaining and nobody spoke yet. |
| 09:31.43 | Chainsaw | As you can see, #asterisk is +tncr. There is no +i. |
| 09:38.29 | Iain_ | Could someone tell me please if there is way for checking if a priority is valid in a similar way to i checking if an extension is valid |
| 09:38.51 | *** join/#asterisk KazaLite (n=KazaLite@94-193-98-124.zone7.bethere.co.uk) |
| 09:46.29 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 09:50.58 | KazaLite | hi all |
| 09:51.34 | KazaLite | anyone around who developed some C code/applicationcodec for asterisk 1.6 and 14 as well? |
| 09:56.39 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:10.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 10:24.38 | *** join/#asterisk cuco (n=diego@DSL212-235-59-62.bb.netvision.net.il) |
| 10:24.48 | cuco | tzafrir_laptop, ping |
| 10:24.58 | tzafrir_laptop | cuco, pong |
| 10:26.06 | tzafrir_laptop | jgoo, on chan_dahdi - use dialing through a group (g, G, r, R) |
| 10:26.48 | macko | Is there a way to make dtmf-activated features (like transfer) to local extensions only, and prevent the remote side from using them? |
| 10:27.18 | macko | Currently, the remote user can press # and transfer the call. |
| 10:37.27 | Iain_ | macko: we had similar problem to this, we got round it by changing blindxfer in features.conf to ###12345, this way people won't accidently press the # key on their handset and transfer the call, this may not be the best way to make it work but it works for us |
| 10:38.21 | *** join/#asterisk Iskorptix_ (n=iskorpti@d205.csc.lt) |
| 10:38.24 | Iskorptix_ | hello |
| 10:39.10 | Iskorptix_ | I want to send a specially crafted packet (SIP options) to asterisk, how I could do that ? |
| 10:39.10 | Iskorptix_ | do I need to send it 5060 ? |
| 10:41.41 | kron4eg | UDP port 5060 |
| 10:41.44 | kron4eg | yes |
| 10:46.51 | macko | Iain_: yes, I did something similar, but figured there must be a safer way. |
| 10:49.18 | Iain_ | macko: there probably is, we couldn't figure it out though, I'd be interested to know how you get around this eventually |
| 10:50.12 | jgoo | tzafrir_laptop, ? for which one of my three quesitons is that? |
| 10:51.35 | jgoo | you mean the third one about failing over an ISDN line? this is for incoming, I am sure it is not possible, without being a telco feature, but they say this (unmarked) seimens PBX did it |
| 10:57.17 | *** join/#asterisk Thazza (n=me@26.126.233.220.static.exetel.com.au) |
| 10:58.02 | macko | How do I Dial() a local extension? |
| 10:58.54 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 11:03.47 | Iain_ | Hi, what variable is the return code stored in after an application has completed? |
| 11:06.31 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 11:25.48 | defswork | anyone know what a CDN line is? |
| 11:26.45 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 11:30.47 | Chainsaw | defswork: I only know CDN as Content Delivery Network, which doesn't seem relevant here. |
| 11:30.58 | defswork | no |
| 11:31.02 | shido6 | or Canadian currency |
| 11:31.33 | defswork | I'm investigating a site that I am certain has been inadvertently facing the internet and has 150k worth of calls routed through it |
| 11:31.54 | defswork | the maintainers say that it has a link to H/O via a "CDN line" |
| 11:32.15 | defswork | all the calls are to Cuban destinations |
| 11:32.42 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 11:32.48 | defswork | there are reported cases of this happening with cisco kit wvia mgcp on unlocked-down setups |
| 11:33.14 | defswork | this is a nortel but it support voip so could be same issues |
| 11:33.17 | defswork | not sure what the scam is with the cuban destinations though |
| 11:34.03 | shido6 | defswork - sounds to me like someone has a wholesale business around your cisco |
| 11:34.11 | shido6 | called " connecting to the back of a switch" |
| 11:34.24 | shido6 | its sold a lot online as a "Grey" route |
| 11:34.36 | defswork | this is a nortel pbx |
| 11:34.48 | defswork | but same thing ultimately |
| 11:34.50 | shido6 | and usually traded online |
| 11:35.01 | shido6 | using third party escrow services |
| 11:35.24 | defswork | problem is that no one will admit that it is/was insecure as they will be liable |
| 11:35.46 | defswork | and the site is in italy so I can't site survey it |
| 11:35.49 | shido6 | then push the blame on the culprits |
| 11:36.05 | shido6 | and if the site is remote have someone take pix |
| 11:36.23 | defswork | shido6: need to make sure that it's secure though and the only way to do that is confirm that it's not (which it clearly isnt) |
| 11:36.47 | shido6 | usually connecting to the back of a switch is an inside job |
| 11:36.51 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 11:37.00 | shido6 | or someone noticed it and exploited the hell out of it |
| 11:37.10 | defswork | I'm hoping for the latter |
| 11:37.20 | shido6 | $150k of calls? someone probably made 25% more than that |
| 11:37.35 | defswork | 150K EU |
| 11:37.44 | Iain_ | Hi all. Could someone tell me please if there is way for checking if a priority is valid in a similar way to i checking if an extension is valid |
| 11:37.46 | shido6 | notice anyone with some new gear, car , clothes, maybe a new house? :) |
| 11:37.54 | shido6 | vacation? |
| 11:38.08 | defswork | shido6: how would they rig access on an inside job ? |
| 11:38.10 | shido6 | anyone overly excited about "working" |
| 11:38.19 | phix | hi |
| 11:38.29 | shido6 | your network is only as secure as the people who protect it |
| 11:39.13 | shido6 | and the promise of another source of income can get an extra couple of plugs added to the network |
| 11:39.34 | shido6 | does anyone on site know about the calls? |
| 11:40.05 | phix | shido6: I live in a carboard box, will a 512bit AES connection keep me safe? |
| 11:40.21 | shido6 | safe from what/whom ? |
| 11:41.02 | defswork | shido6: I think so now for sure - they locked the PBX down from making international calls |
| 11:41.24 | defswork | the exploit could still be there though |
| 11:41.57 | shido6 | whats connected to the nortel? |
| 11:42.01 | shido6 | no voip stuff , right? |
| 11:42.08 | defswork | no |
| 11:42.28 | defswork | apart from this "CDN line" to h/o which might be voip |
| 11:42.49 | shido6 | is that term on a bill , where did u get that term again? |
| 11:43.02 | defswork | the pbx maintainers send a nice diagram |
| 11:43.14 | phix | shido6: my nextdoor neighbour who lives in a box? |
| 11:43.19 | defswork | branch <-- CDN line --> h/o |
| 11:43.37 | shido6 | the pbx maintainers are they aware of the fraud? |
| 11:43.39 | defswork | they have severed the CDN line now |
| 11:43.51 | defswork | shido6: I think so as my questions have been relayed to them |
| 11:43.59 | shido6 | :( |
| 11:44.15 | shido6 | want to fly out there? |
| 11:44.26 | defswork | not really :) |
| 11:45.18 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 11:46.09 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 11:53.12 | defswork | http://www.telecomclassifiedads.com/directroutes2.html :o |
| 11:53.17 | defswork | why is cuba so desired ? |
| 11:55.18 | *** join/#asterisk yziquel (i=55da5c63@gateway/web/freenode/x-fcljlsbiycjpnyon) |
| 11:56.14 | yziquel | hi. i'm installing an Asterisk server behind a NAT, and unfortunately nat keepalive is not an option on the router, and the firewall does stateful packet inspection. |
| 11:56.29 | yziquel | how would you go about this? is siproxd a solution? |
| 12:02.51 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:06.41 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:08.47 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:10.22 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:10.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:10.45 | *** join/#asterisk s14ck (n=s14ck@190-76-98-56.dyn.movilnet.com.ve) |
| 12:14.08 | TSM | is it possable to join two conference rooms together, cant work out the CLI command to do it |
| 12:16.26 | aurax | i need help setting up fonebridge with dahdi, but i'm getting this error: http://pastebin.com/d1b68a29 can anyoen assist? |
| 12:21.47 | Psychobilly | does sending text messages over sip work in asterisk? |
| 12:21.59 | Psychobilly | im trying somehting very basic using ekiga |
| 12:22.21 | TSM | how can i dial from the CLI? |
| 12:22.32 | Psychobilly | and i get this: [Aug 5 15:21:09] WARNING[10015]: chan_sip.c:10207 receive_message: Received message to <sip:410@192.168.1.210> from "Z" <sip:ox2@192.168.1.210>;tag=b2230e20-2880-de11-8cff-000fb0773100, dropped it... |
| 12:22.32 | Psychobilly | <PROTECTED> |
| 12:22.32 | Psychobilly | <PROTECTED> |
| 12:23.17 | Psychobilly | i also tried sendtest() without much success |
| 12:23.25 | Psychobilly | sendtext() |
| 12:23.45 | Psychobilly | TSM using chan_alsa |
| 12:24.06 | Psychobilly | and trying somehting like 'console dial extension@xontext' |
| 12:25.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:25.22 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 12:25.23 | TSM | ok so i need to load it first in modules.conf |
| 12:25.37 | Psychobilly | module load chan_alsa.so |
| 12:25.39 | Psychobilly | in cli |
| 12:25.46 | TSM | can i use that dial command to join two confrences togethere? |
| 12:26.01 | Psychobilly | i dont think so |
| 12:26.24 | *** join/#asterisk yziquel (i=55da5c63@gateway/web/freenode/x-vldzihziurwgymyu) |
| 12:26.55 | TSM | any ideas how? |
| 12:27.15 | grandpapadot | TSM: I don't think you can bridge two conferences ... |
| 12:27.29 | grandpapadot | TSM: YOu might try to transfer the members of one conference to another ... |
| 12:28.15 | grandpapadot | TSM: http://www.voip-info.org/wiki/view/Asterisk+cmd+ChannelRedirect |
| 12:31.17 | TSM | when i say bridge, i mean that i want to dial in another asterisk boxes confrence room |
| 12:31.32 | TSM | in a way just adding them as a user into my conference |
| 12:32.01 | TSM | so they dont need to all connect to my server, they stay connected to theirs and we just bridge one channel across servvers |
| 12:33.16 | leifmadsen | TSM: you could do that with Originate() in 1.6.1.x+ or with callfiles (or with an AMI connection) |
| 12:33.19 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:33.23 | leifmadsen | [TK]D-Fender: WHAT?! |
| 12:33.53 | [TK]D-Fender | leifmadsen: I don't want to meet your mom! |
| 12:33.59 | leifmadsen | I just want... |
| 12:34.00 | TSM | @leifmadsen: ahh, /var/spool/asterisk and dump a SIP connection between their sip addy any my conference |
| 12:34.01 | [TK]D-Fender | leifmadsen: ! ! ! |
| 12:34.12 | leifmadsen | TSM: something like that, ya :) |
| 12:34.12 | aurax | [TK]D-Fender :) sup |
| 12:34.26 | aurax | [TK]D-Fender, what about saving my ass? |
| 12:34.55 | aurax | <- got fonebridge2 x2 pri and lame kernel panics cuz of dahdi |
| 12:35.01 | *** join/#asterisk DrunkenMaster (i=tux@195.26.24.3) |
| 12:35.28 | [TK]D-Fender | aurax: Jesus Saves...... the Devil does triple redundant off-site backups (Full & incremental) |
| 12:35.39 | aurax | lol |
| 12:35.59 | [TK]D-Fender | aurax: Get a better kernel |
| 12:36.03 | aurax | i got the servers back now, just asterisk wouldn't boot saying my dahdi is misconfugered. |
| 12:36.06 | aurax | configured* |
| 12:36.33 | aurax | [Aug 5 15:21:32] ERROR[8878] asterisk.c: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: |
| 12:36.40 | aurax | tada... |
| 12:36.48 | [TK]D-Fender | aurax: Yes, I know what a DAHDI startup error looks like... |
| 12:37.29 | aurax | mind to have a look @ my pastebin's? |
| 12:38.05 | aurax | http://pastebin.com/m199ab599 |
| 12:41.45 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 12:41.54 | [TK]D-Fender | auxI neither support nor recommend Redfone.... |
| 12:42.00 | [TK]D-Fender | aurax: I neither support nor recommend Redfone.... |
| 12:42.14 | aurax | why ? |
| 12:44.27 | [TK]D-Fender | aurax: Problems like this. |
| 12:44.35 | beek | Morning [TK]D-Fender |
| 12:44.59 | [TK]D-Fender | beek: Yes... yes it is... |
| 12:45.32 | aurax | [TK]D-Fender it's more then a dahdi issue |
| 12:46.54 | beek | [TK]D-Fender: I can't say good morning... came out to my car this morning and my iPod was stolen from the glovebox. I forgot to lock the damned car last night. I WANT BLOOD! |
| 12:49.34 | yziquel | any idea how to make an Asterisk inside a NAT to connect to the internet through a Juniper (argh!) firewall? |
| 12:49.52 | [TK]D-Fender | ~sipnat |
| 12:49.53 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:49.56 | [TK]D-Fender | yziquel: ^^^ |
| 12:50.56 | yziquel | [TK]D-Fender: i've looked at these, but i'm still at a loss. that's why i'm asking. |
| 12:51.25 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:51.46 | [TK]D-Fender | yziquel: While you're at a loss, I don't have SIP DEBUG from failed calls, or configs TO lose yet.... |
| 12:52.23 | yziquel | [TK]D-Fender: I'll paste them shortly. |
| 12:53.16 | yziquel | [TK]D-Fender: my softphones are on the same LAN as Asterisk. It's the connection to the outbound SIP server that doesn't work. Phone calls last for 30 secs, after which no voice in both directions. |
| 12:54.32 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:55.12 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 12:55.39 | *** join/#asterisk apeiron (i=apeiron@isuckatdomains.net) |
| 12:56.07 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:56.18 | *** part/#asterisk jake[work] (n=Jake@pool-173-52-144-183.nycmny.east.verizon.net) |
| 12:58.47 | *** join/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com) |
| 13:00.42 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:01.24 | tzafrir_laptop | aurax, it basically means a dahdi span exists, but does not provide timing |
| 13:01.41 | tzafrir_laptop | dahdi_test hangs |
| 13:02.24 | yziquel | [TK]D-Fender: http://paste.lisp.org/display/84814 |
| 13:02.58 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 13:03.53 | [TK]D-Fender | yziquel: Useless degub... actual CLI only with no masking. |
| 13:03.58 | [TK]D-Fender | debug* |
| 13:04.25 | yziquel | [TK]D-Fender: Ok. I'll trim down a more useful one. Too big to paste for now. |
| 13:06.34 | *** join/#asterisk shazaum (n=889uJWER@unaffiliated/shazaum) |
| 13:08.11 | *** join/#asterisk ebil|work (n=andy@wsip-98-191-211-137.dc.dc.cox.net) |
| 13:12.18 | *** join/#asterisk synthetic (n=roger@193.79.224.62) |
| 13:12.43 | synthetic | <-- looking for job oppotunities in socal or phoenix area |
| 13:12.52 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:13.13 | *** join/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 13:15.32 | aurax | tzafrir_laptop, true dahdi_test hangs |
| 13:15.40 | aurax | hwo can i stop asterisk from using dahdi |
| 13:15.52 | leifmadsen | don't load chan_dahdi.so ? |
| 13:16.10 | PMantis | modules.conf ? |
| 13:16.18 | tzafrir_laptop | in that version? a bit tricky |
| 13:16.21 | aurax | 1.4.25 |
| 13:16.29 | tzafrir_laptop | some other things depend on dahdi |
| 13:16.38 | tzafrir_laptop | you'll have to rebuild asterisk |
| 13:16.44 | aurax | it's not located in /etc/asterisk.elastix/modules.conf |
| 13:16.51 | [TK]D-Fender | ! |
| 13:16.51 | aurax | bah |
| 13:16.54 | [TK]D-Fender | \o/ |
| 13:17.04 | aurax | compiling ... bleh |
| 13:17.27 | PMantis | From source is it ONLY way I install Asterisk |
| 13:17.38 | tzafrir_laptop | ugly workaround: try using dahdi_dummy as a timing source |
| 13:17.55 | tzafrir_laptop | still, it means something is wrong with your trunk |
| 13:18.14 | tzafrir_laptop | (tdmoe-mf?) |
| 13:19.11 | [TK]D-Fender | tzafrir_laptop: RedFone D: |
| 13:20.22 | PMantis | I'm getting SO many dnsmgr_lookup entries on the screen, that the CLI is almost unusable. Scrolls like crazy. IMO, it shouldn't do that! |
| 13:21.03 | tzafrir_laptop | don't scroll, then |
| 13:21.24 | leifmadsen | turn off dnsmgr? |
| 13:22.09 | PMantis | Check this 10-second clip: http://pastebin.com/d18f089ff |
| 13:22.22 | leifmadsen | I know what it would look like |
| 13:23.22 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 13:24.02 | [TK]D-Fender | PMantis: Asterisk 1.6.1.0 <-- upgrade |
| 13:25.21 | PMantis | I don't have a problem with these entries in the log, or even the CLI, but... when I type 'sip show registry' and within 3 seconds the results scrolls off the screen from over 30 DNS lookups for the exact same hostname. |
| 13:25.56 | PMantis | [TK]D-Fender: OK, I can do that... but this has been happening for many months - I've asked before in IRC, and I'm just sick of it again. |
| 13:26.24 | [TK]D-Fender | PMantis: and your version isn't current. Similar much? :) |
| 13:26.45 | yziquel | [TK]D-Fender: http://paste.lisp.org/display/84815 |
| 13:26.57 | yziquel | yziquel: is that better? |
| 13:27.01 | PMantis | [TK]D-Fender: "similar much"?? |
| 13:27.06 | yziquel | [TK]D-Fender: is that better? |
| 13:27.36 | [TK]D-Fender | yziquel: Sending to 192.168.23.85 : 5060 (NAT) <-- why is a Class-C origin IP **NAT**? |
| 13:28.14 | [TK]D-Fender | yziquel: Reliably Transmitting (NAT) to 91.121.167.75:5060: <--- ITSP's should NEver be NAT |
| 13:28.35 | leifmadsen | using NAT won't hurt anything... |
| 13:28.37 | [TK]D-Fender | yziquel: Go fix. |
| 13:29.03 | [TK]D-Fender | leifmadsen: on the ITSP-side it will |
| 13:29.11 | yziquel | [TK]D-Fender: will fix shortly. thanks. but i do not expect it is the main issue. |
| 13:29.15 | PMantis | [TK]D-Fender: I would argue that the destination doesn't matter... it's what's in between that does. |
| 13:29.21 | leifmadsen | [TK]D-Fender: why? |
| 13:30.01 | [TK]D-Fender | leifmadsen: because when you tell * NAT it also overrides the IP for RTP to the signalling source and many ITSP's carry that on other servers. |
| 13:30.20 | leifmadsen | I don't think that is really true. Pretty sure it only affects the SIP headers |
| 13:30.21 | [TK]D-Fender | \o/ big deployments |
| 13:30.28 | [TK]D-Fender | leifmadsen: I've seen this many times... |
| 13:30.44 | [TK]D-Fender | leifmadsen: yziquel here should have results in a moment either way |
| 13:31.00 | leifmadsen | I've never seen that. Usually the only problem with RTP is when Asterisk is behind NAT and doesn't get audio from the other end, it doesn't know where to send audio to. |
| 13:31.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:31.14 | leifmadsen | doesn't really car. I've never seen that though. |
| 13:31.35 | PMantis | leifmadsen: Right, since the "contact: " line i the SIP message didn't use the external IP of the originating * system. |
| 13:31.52 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 13:32.22 | leifmadsen | I always use nat=yes on internal phones, and have zero issues |
| 13:32.49 | leifmadsen | the value on the IP header and the Contact will be the same |
| 13:32.52 | leifmadsen | so it doesn't matter |
| 13:32.57 | [TK]D-Fender | leifmadsen: Of for phones its usually OK, its the ITSP side that kills. |
| 13:33.00 | [TK]D-Fender | Oh* |
| 13:33.10 | leifmadsen | I have nat=yes on an ITSP -- zero issues. |
| 13:33.16 | Iain_ | Could someone tell me please if there is way for checking if a priority is valid in a similar way to i checking if an extension is valid |
| 13:33.19 | [TK]D-Fender | leifmadsen: Depends on the ITSP too ;) |
| 13:33.36 | [TK]D-Fender | leifmadsen: But you should always let them take care of their own business anyway... |
| 13:33.50 | [TK]D-Fender | yziquel: You should also have "canreinvite=no" for all of your peers |
| 13:34.31 | yziquel | [TK]D-Fender: canreinvite is set to no for all my peers |
| 13:34.37 | yziquel | [TK]D-Fender: already. |
| 13:34.43 | [TK]D-Fender | yziquel: Ok. So whats the result? |
| 13:35.02 | *** join/#asterisk x86 (n=porteb1@p3m/member/x86) |
| 13:35.16 | yziquel | [TK]D-Fender: working remotely, so it's a bit of a pain to get results quickly. will post asap. |
| 13:37.35 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 13:41.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:41.47 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:43.47 | *** join/#asterisk RobiGo (n=R@85.186.103.66) |
| 13:43.51 | RobiGo | hi |
| 13:43.58 | RobiGo | i have a question |
| 13:44.02 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 13:44.23 | RobiGo | i buyed an account for siptraffic.com |
| 13:44.42 | RobiGo | how i can configure my Asterisk to use it? |
| 13:44.55 | mweichert | in my dialplan, can I reference a global variable or something similiar which provides information about how many active channels exist? |
| 13:44.59 | [TK]D-Fender | RobiGo: They don't ahve a config sample for you? |
| 13:45.11 | RobiGo | nop |
| 13:45.27 | [TK]D-Fender | mweichert: You'll need to use some external script to figure that out |
| 13:45.43 | RobiGo | i received only a username and passw |
| 13:45.47 | [TK]D-Fender | RobiGo: then go look at how other ITSP's samples look like and begin with their style |
| 13:45.48 | SuPrSluG | can polycom's get vlan id from dhcp? |
| 13:45.50 | RobiGo | nothing else |
| 13:45.52 | synthetic | RobiGo: you pay some consultant to do it for you :) |
| 13:46.07 | [TK]D-Fender | RobiGo: Hree : |
| 13:46.11 | [TK]D-Fender | ~itsplist-us |
| 13:46.11 | infobot | well, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 13:46.11 | mweichert | [TK]D-Fender, ok, thanks |
| 13:46.13 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 13:46.16 | RobiGo | :)) synthetic nice :D |
| 13:46.24 | synthetic | last i used polycomes vlan ID must be set manually |
| 13:47.10 | SuPrSluG | k, thanks |
| 13:48.14 | RobiGo | where i can start the configuration? it is a trunk? or an extension? i dont know.. |
| 13:48.42 | [TK]D-Fender | RobiGo: sip.conf |
| 13:50.27 | RobiGo | bad idea, the sip conf start with "Don't Edit... " |
| 13:51.08 | *** join/#asterisk jgoo (n=r3rman@ppp16-218.adsl.forthnet.gr) |
| 13:51.28 | RobiGo | what windows voip software is recomanded for Asterisk? |
| 13:51.34 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 13:52.02 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 13:52.07 | RobiGo | what windows voip software is recomanded for Asterisk? |
| 13:52.31 | [TK]D-Fender | RobiGo: GUI's are NOT supported here, check with their channel |
| 13:52.31 | synthetic | gives cookie points to TK-D |
| 13:52.34 | [TK]D-Fender | RobiGo: and don't repeat questions just because noone has answered you in ONE MINUTE |
| 13:52.40 | jgoo | Right, I wrote 3-4 questions earlier, the fucktards who wrote xchat were retarded enough to let my channel window get spammed with server reconnect messages (instead of having a new tab for that shit) - so I have no idea if someone answered |
| 13:52.41 | *** join/#asterisk nightrid3r (n=borgirc-@78-20-232-172.access.telenet.be) |
| 13:52.56 | [TK]D-Fender | RobiGo: Any soft-phone will work. go try X-Lite first |
| 13:52.59 | jgoo | Hey TK |
| 13:53.00 | synthetic | xchat is brilliantly written |
| 13:53.07 | jgoo | why is #asterisk +i? it is so stupid. anyway, grandstreams - how do I set them to use dialplans? I've seen people asking about dialplans, but cannot find anything in manual or interface (budgetone 201) |
| 13:53.16 | jgoo | echo - simple question, which I've NEVER seen address - where is it introduced into the system? |
| 13:53.19 | RobiGo | ok, thank's <[TK]D-Fender> |
| 13:53.20 | synthetic | i agree +i chan mode = FTL |
| 13:53.22 | zeeesh | compiling asterisk-1.4 ubuntu-8.04 getting error 'make[1]: *** [editline/libedit.a] Error 2'? |
| 13:53.27 | jgoo | ISDN - is it possible to have one ISDN line failover to another line if it is busy? An old siemens system supposedly has 3 incoming isdn, and from calling one number, you can get 6 lines up - of course I think this is impossible, but they swear to me... |
| 13:53.32 | synthetic | or registration requirement rather |
| 13:53.37 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 13:53.41 | jgoo | Those three questions |
| 13:54.00 | ricko73 | morning |
| 13:54.02 | jgoo | Grandstreams - how do you dialplan a 201? Echo - where is it introduced? ISDN - can you have line failover |
| 13:54.21 | *** part/#asterisk RobiGo (n=R@85.186.103.66) |
| 13:54.32 | ricko73 | Is there some way with Zaptel cards to force them to wait for a dialtone before attempting to dial? |
| 13:54.39 | jgoo | room full of people using voip - does anyone know why we have echo cancellation? Why do normal phones not have echo - or, why do voip phones have echo - what causes it? |
| 13:54.40 | synthetic | jgoo yes you can have lien roll over |
| 13:54.51 | jgoo | synthetic - how does that work? |
| 13:54.52 | [TK]D-Fender | jgoo: GS's don't HAVE dialplans. their that shit. |
| 13:54.59 | [TK]D-Fender | they're* |
| 13:55.15 | Stese | jgoo > http://pastebin.com/d19ae3a47 |
| 13:55.23 | [TK]D-Fender | jgoo: And GS's firmware itself has been known to be the cause for echo |
| 13:55.24 | jgoo | [TK]D-Fender, shit. I wondered. I have people hitting # all the time - can I ask, where the fuck is all this documented (like, don't buy grandstream, they are shit |
| 13:55.32 | [TK]D-Fender | ~gs |
| 13:55.33 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:55.36 | [TK]D-Fender | ~grandstream |
| 13:55.37 | infobot | i guess grandstream is the Yugo of VoIP hardware. Run. Run away now.. Though therealcircut says that they're not that bad |
| 13:55.38 | synthetic | jqoo provider must configure the capability |
| 13:55.58 | [TK]D-Fender | ~echo |
| 13:55.59 | infobot | well, echo is an issue which can be best fixed using this link: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-5.html, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 13:56.10 | jgoo | aaah :p damn |
| 13:56.18 | [TK]D-Fender | jgoo: it 2>4 wire conversion, impedence, etc |
| 13:56.47 | PMantis | jgoo: ALL phones echo, but if the echo comesback without delay, our brain ignores it as a normal consequence of talking. A longer delay causes the need to filter so our brain doesn't "hear" the echo. |
| 13:56.48 | eliyahud | echo comes from impedance mismatch |
| 13:56.56 | [TK]D-Fender | jgoo: First ensure you're on the latest firmware on the phones. Then test with jsut phones local to *. THEN involve outside resources. You need to pin down the culprit(s) |
| 13:57.12 | jgoo | aah, right, so the other persons phone is causing the echo. |
| 13:57.22 | [TK]D-Fender | jgoo: No, could be YOU |
| 13:57.30 | jgoo | Can't they just put a noise cancellation circuit into the mouthpiece that negates what is in the earpiece? |
| 13:57.32 | [TK]D-Fender | jgoo: Now go break this down. |
| 13:57.54 | jgoo | I mean, I don't care if I am causing echo on their side, I mean the echo I am hearing is from their mic picking up my voice |
| 13:57.58 | jgoo | and vice versa |
| 13:58.08 | jgoo | Thanks [TK]D-Fender and PMantis |
| 13:58.08 | PMantis | jgoo: yes |
| 13:58.50 | jgoo | So, if all phones had a $.1 signal processor that removed the immediate pickup of the earpiece into the mic, the world would be a better place? |
| 13:59.20 | Stese | jgoo > PSTN in the UK does |
| 13:59.20 | [TK]D-Fender | jgoo: ..... stop placing blame on hardware alone, let alone WHOSE. |
| 13:59.25 | jgoo | [TK]D-Fender, I'll go with that approach you describe tomorrow |
| 14:00.23 | jgoo | [TK]D-Fender, no, I generally do that - and I am right, there is the problem - if I had designed a phone, essentially a speaker and a mic, my first thought would be, let's not have the speaker interfere with the mic... I guess back in the day that would have been expensive... but I've seen circuits that do stuff like that. Just procrasinating at the weakness of human science |
| 14:00.35 | *** join/#asterisk jtodd (i=hqghyef6@ns.fox-den.com) |
| 14:00.35 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:00.37 | jgoo | Stese, PSTN in UK does what? has echo cancellation in handsets? |
| 14:00.50 | Stese | Yeah, IIRC |
| 14:01.02 | *** join/#asterisk ta^3 (n=tacvbo@67.201.69.184) |
| 14:01.58 | jgoo | Wouldn't that depend on the handset? since there is no 'in and out' pipe on a phone line... they'd need to phase shift very finely... sound improbably |
| 14:03.00 | jgoo | synthetic, line roll over - can this be a telco feature? I have 2 ISDN lines, they say, the number on line 1 went to line 2 - I can't imagine how the PBX did this (!?) since the physical wire wouldn't be in use - but if it is a telco feature - wouldn't it work now? |
| 14:03.06 | *** join/#asterisk _gm (n=_gm@203.215.176.22) |
| 14:03.14 | _gm | hello |
| 14:03.17 | Stese | what type of ISDN lines? |
| 14:03.19 | jgoo | I only tried calling one ISDN line 3 times, only 2 went through, the line was reserved for the third |
| 14:03.25 | synthetic | yes its a telco feature |
| 14:03.34 | jgoo | erm. 3 channel BBD... BRI |
| 14:04.12 | jgoo | synthetic, so I need to actually pickup the first two lines... then it should let that other number act like an msn on the second line? I'll try it... 3 ringing lines didn't work, will try the other |
| 14:04.35 | Stese | jgoo > http://communication.howstuffworks.com/telephone1.htm |
| 14:05.22 | *** join/#asterisk c4rg (i=crg@lagoon.freebsd.lublin.pl) |
| 14:05.24 | jgoo | Wow, you answered all my questions. cool. Damn, grandstream suck so bad. unusable crap - either waits too long to dial, or times out... much better to keep analogue handsets and slap PAP2Ts in there.... hrm |
| 14:05.46 | jgoo | I did get two call management working ok on a 201... using flash... works... but , meh |
| 14:06.36 | c4rg | anyone having problems with "rx overrun errors" on Sangoma's E1 card? |
| 14:06.44 | jgoo | Right.... wait.. so this duplex coil... why can't it make the other persons voice not reach the mic? hrm... |
| 14:06.54 | *** join/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk) |
| 14:07.44 | [TK]D-Fender | jgoo: BT=shitty ATA w/ handset duct-taped to it |
| 14:07.54 | x86 | hahahaha |
| 14:07.59 | PMantis | ROFL |
| 14:08.01 | x86 | [TK]D-Fender++ |
| 14:08.10 | PMantis | I use Polycom for my clients |
| 14:08.23 | x86 | polycom phones are the only ones I'll even deal with |
| 14:08.43 | x86 | i've tried linksys, they are ok, but provisioning them is a huge nightmare |
| 14:08.55 | Stese | Jgoo > sorry, i got the wrong end of the stick there... turn the output volume down, :P |
| 14:09.04 | [TK]D-Fender | x86: Really? the config format looked pretty simple. What are some "gotchas"? |
| 14:09.13 | x86 | everyone says Snom and Aastra phones are decent, but I've not messed with them |
| 14:09.18 | PMantis | I wrote some PHP scripts that work with mod_rewrite and auto provision polycom phones. |
| 14:09.33 | x86 | [TK]D-Fender: you found documentation somewhere? I was not able to find any |
| 14:09.51 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:09.52 | x86 | [TK]D-Fender: also I couldn't see how to make them get their config from an FTP server |
| 14:09.54 | [TK]D-Fender | x86: Plenty, even on the WIKI |
| 14:10.12 | [TK]D-Fender | x86: I never really dug in since I sold the only one I ever had, but its around |
| 14:10.13 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:10.33 | ariel_ | sipura and linksys spa are fairly easy to mass deploy and the wiki even has a link to the software to setup the configs for them |
| 14:10.52 | x86 | [TK]D-Fender: I know you can backup / restore an XML config from the phone's internal webserver, but I don't think you can make it grab config from an FTP server |
| 14:11.18 | x86 | ariel_: yeah, which sucks if you don't run windows :) |
| 14:11.20 | [TK]D-Fender | x86: They can do TFTP & HTTP last I checked. Not sure about others. |
| 14:11.37 | x86 | [TK]D-Fender: well I already had the FTP setup from the Polycom phones |
| 14:11.40 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:11.58 | x86 | [TK]D-Fender: since polycom does TFTP, FTP, FTPS, HTTP, and HTTPS :P |
| 14:12.58 | yziquel | [TK]D-Fender: http://paste.lisp.org/display/84820 |
| 14:13.03 | yziquel | [TK]D-Fender: is it better? |
| 14:13.23 | yziquel | [TK]D-Fender: the sounds still drops on both directions. |
| 14:14.56 | [TK]D-Fender | yziquel: Your firewall should not be doing any SIP transform |
| 14:15.11 | [TK]D-Fender | x86: Oh no need to preach to me about Polycom ;) |
| 14:15.23 | [TK]D-Fender | IS the choir |
| 14:15.35 | PMantis | lol |
| 14:15.56 | yziquel | [TK]D-Fender: Is it doing such an SIP transform? |
| 14:16.09 | [TK]D-Fender | yziquel: I'll likely never know.. |
| 14:16.41 | [TK]D-Fender | yziquel: Sending to 192.168.23.101 : 5060 (NAT) <-- This phone is LOCAL, isn't it? Still says NAT |
| 14:17.38 | yziquel | [TK]D-Fender: hummm.... |
| 14:17.41 | [TK]D-Fender | yziquel: Reliably Transmitting (NAT) to 91.121.167.75:5060: <-- and NO, nothing seems to have changed |
| 14:18.05 | yziquel | [TK]D-Fender: I should have nat=no everywhere? |
| 14:18.12 | [TK]D-Fender | yziquel: Use pastebin.com as well. it NUMBERS the lines for easier commenting |
| 14:18.40 | [TK]D-Fender | yziquel: On all ITSP entries, and local phones. Basically NAT should only be for remote NAT'd phones. |
| 14:18.48 | yziquel | [TK]D-Fender: for the outbound sip server, i use nat=yes. Shouldn't I? |
| 14:18.57 | [TK]D-Fender | yziquel: NO |
| 14:19.05 | [TK]D-Fender | yziquel: How many more times to I need to say this? |
| 14:19.12 | yziquel | ah! sorry. |
| 14:19.33 | [TK]D-Fender | [09:32]<[TK]D-Fender>leifmadsen: Of for phones its usually OK, its the ITSP side that kills. |
| 14:19.57 | [TK]D-Fender | [09:29]<[TK]D-Fender>leifmadsen: because when you tell * NAT it also overrides the IP for RTP to the signalling source and many ITSP's carry that on other servers. |
| 14:21.46 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:391a:13ea:4177:c2c3) |
| 14:22.52 | tzafrir_laptop | http://lwn.net/Articles/345357/ |
| 14:23.11 | tzafrir_laptop | and no: this is not about a Bug Tracking System |
| 14:23.19 | *** join/#asterisk b3nw (n=ben@unaffiliated/b3nw) |
| 14:23.23 | *** join/#asterisk willsey (n=mikew@79-74-194-25.dynamic.dsl.as9105.com) |
| 14:24.17 | *** join/#asterisk s14ck (n=s14ck@200.62.25.156) |
| 14:25.24 | willsey | hi everyone..can anyone give any assistance with setting up an AEX410 card on freebx V2.5.1. pls |
| 14:25.34 | [TK]D-Fender | willsey: Wrong channel. |
| 14:25.38 | [TK]D-Fender | ~freepbx |
| 14:25.39 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:25.45 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:25.56 | willsey | ok thnx |
| 14:26.21 | yziquel | [TK]D-Fender: so nat=no for the outbound proxy.... |
| 14:26.25 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:26.30 | *** part/#asterisk willsey (n=mikew@79-74-194-25.dynamic.dsl.as9105.com) |
| 14:26.34 | *** join/#asterisk manxpower (n=eric@69.73.94.162) |
| 14:27.57 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:27.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:28.07 | *** join/#asterisk dajhorn (n=dajhorn@sueno.dreamhost.com) |
| 14:28.43 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 14:29.08 | ceegee | re |
| 14:29.34 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 14:30.05 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 14:30.16 | viraptor | I've got an attended transfer problem... let's say that I've got N asterisks with many different domains (too many to list, so asterisk just allows any domains) I also load-balance traffic between those asterisks - is there any sane way to make attended transfers work in that scenario? with a mixture of huntgroups, queues and sip channels I always end up with 2 channels that I want to join on different hosts :/ |
| 14:30.34 | *** join/#asterisk Defraz (n=T0tal@c72co-edge-router.fuzecore.com) |
| 14:31.22 | Psychobilly | does sending text messages over sip work in asterisk? |
| 14:31.27 | Psychobilly | im trying somehting very basic using ekiga |
| 14:31.34 | ceegee | in my sip.conf my usernames are real names and not numbers, e.g. [user1], now I have to read out the username in my dialplan to do some db stuff, when I user ${SIPCHANINFO(peername)} I alway get something like SIP/user1-0b8eme38 and not only SIP/user1, this is very anoying, how can I read out only the username without these numbers |
| 14:31.43 | Psychobilly | and i get this: [Aug 5 15:21:09] WARNING[10015]: chan_sip.c:10207 receive_message: Received message to <sip:410@192.168.1.210> from "Z" <sip:ox2@192.168.1.210>;tag=b2230e20-2880-de11-8cff-000fb0773100, dropped it... |
| 14:31.49 | Psychobilly | <PROTECTED> |
| 14:31.55 | [TK]D-Fender | Psychobilly: No. |
| 14:31.55 | Psychobilly | <PROTECTED> |
| 14:32.04 | [TK]D-Fender | Psychobilly: IM != telephony |
| 14:32.25 | Psychobilly | sure, but since there is support for messages in sip.... |
| 14:32.30 | [TK]D-Fender | ceegee: "core show function CUT" |
| 14:32.37 | Psychobilly | i also tried sendtext() in * |
| 14:32.42 | Psychobilly | but nothing happened |
| 14:32.49 | [TK]D-Fender | Psychobilly: And we all know how well * supports SIP, now don't we? |
| 14:32.56 | *** join/#asterisk Jabka (n=jabka@hspot.sce.ac.il) |
| 14:33.05 | Psychobilly | lol |
| 14:33.11 | Psychobilly | ok u won :P |
| 14:33.14 | ceegee | [TK]D-Fender: I already tried with no luck |
| 14:33.17 | [TK]D-Fender | Psychobilly: And jsut because your car has a radio doesn't mean that it should be able to mount a big-screen TV in it |
| 14:33.36 | [TK]D-Fender | ceegee: You don't need "luck", you just need to do it right |
| 14:33.56 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 14:34.03 | [TK]D-Fender | ceegee: So show us a valid attempt and we will advise on how to fix it |
| 14:34.35 | Jabka | sorry for my newb question but how do i get the svn tree (i tried "svn co http://svn.digium.com/svn/asterisk/ foo " and got "svn: Server sent unexpected return value (400 Bad Request) in response to PROPFIND request for '/svn/asterisk'" ? |
| 14:36.05 | *** join/#asterisk coppice (n=chatzill@110.202.17.210.dyn.pacific.net.hk) |
| 14:36.28 | viraptor | Jabka: maybe you've got a stupid proxy in the way? svn ls http://svn.digium.com/svn/asterisk/ works for me |
| 14:36.44 | Jabka | no proxys afaik |
| 14:36.57 | Jabka | so i guess i have some strange net issues |
| 14:37.54 | ceegee | [TK]D-Fender: I tried it with exten => 10,1,Set(username=${CUT(${SIPCHANINFO(peername)},-,1)}) |
| 14:38.21 | ceegee | but this puts out: |
| 14:38.22 | ceegee | <PROTECTED> |
| 14:38.25 | ceegee | <PROTECTED> |
| 14:38.35 | ceegee | does not look like what I want |
| 14:39.11 | [TK]D-Fender | ceegee: because you do not pass CTU just 'text'. You pass it a VARIABLE name. Read the instructions again |
| 14:39.15 | [TK]D-Fender | CUT* |
| 14:39.44 | ceegee | [TK]D-Fender: this was an example I found |
| 14:40.35 | [TK]D-Fender | ceegee: Well I guess you should go read the instructions like I posted the CLI command for you to do and try again |
| 14:41.21 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:41.36 | *** join/#asterisk ingenius (n=alektro@host215.200-45-165.telecom.net.ar) |
| 14:41.56 | yziquel | Jabka: checkout if it is only the asterisk svn that sends back this, or any svn repo. |
| 14:42.33 | Jabka | only with digum server any other server worked |
| 14:43.19 | ceegee | [TK]D-Fender: this is the example I found: exten => 123,n,Set(var=${CUT(var,,1-3&5)}) |
| 14:43.26 | ceegee | looks like mine I think |
| 14:43.46 | [TK]D-Fender | ceegee: You think? You don't actually know if it looks like yours? Have you seen yours lately? |
| 14:44.03 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 14:44.07 | [TK]D-Fender | posts a "missing" poster... |
| 14:44.25 | *** join/#asterisk xa0z (n=Interex@75-129-230-28.dhcp.mtvr.il.charter.com) |
| 14:44.53 | xa0z | Can anyone here tell me a good VoIP/SIP Provider with unlimited plans, and multiple channels that I can use my own device? |
| 14:45.26 | ceegee | [TK]D-Fender: for me it looks identical, I only replaced var by the variable I want to split |
| 14:45.28 | [TK]D-Fender | xa0z: Most unlimited's are limited to 2 channels |
| 14:45.42 | xa0z | But simotaniously? |
| 14:45.48 | [TK]D-Fender | ceegee: No, in your first one, you pass it TEXT, not a VARIABLE |
| 14:45.55 | [TK]D-Fender | xa0z: Usually simultaneous. |
| 14:46.13 | [TK]D-Fender | ceegee: ${SIPCHANINFO(peername)} <- text |
| 14:46.16 | viraptor | xa0z: which country do you mean? |
| 14:46.36 | [TK]D-Fender | ceegee: ${var1} <- text |
| 14:46.43 | [TK]D-Fender | ceegee: var1 <- VARIABLE |
| 14:46.43 | xa0z | USA |
| 14:46.57 | [TK]D-Fender | xa0z: here, shop around : |
| 14:47.00 | [TK]D-Fender | ~itsplist-us |
| 14:47.01 | infobot | hmm... itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 14:48.07 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 14:50.08 | xa0z | Ok. |
| 14:50.11 | ceegee | [TK]D-Fender: ok, now I have |
| 14:50.12 | ceegee | exten => 10,1,Set(split=${SIPCHANINFO(peername)}) |
| 14:50.12 | ceegee | exten => 10,2,Set(username=${CUT(split,-,1)}) |
| 14:50.12 | ceegee | exten => 10,3,NoOp(${username}) |
| 14:50.49 | [TK]D-Fender | ceegee: PAStebiN <- |
| 14:50.50 | [TK]D-Fender | ~pb |
| 14:50.51 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 14:50.54 | ceegee | sorry |
| 14:51.34 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 14:52.05 | brah | Say i used exten => _88xxx, how could I tell what that xxx is? |
| 14:52.25 | viraptor | brah: ${EXTEN:2} |
| 14:52.32 | viraptor | or something like that... |
| 14:52.45 | ceegee | exten => 10,1,Set(split=${SIPCHANINFO(peername)}) |
| 14:52.46 | ceegee | exten => 10,2,Set(username=${CUT(split,-,1)}) |
| 14:52.46 | ceegee | exten => 10,3,NoOp(${username}) |
| 14:52.47 | Chainsaw | [TK]D-Fender: rafb.net is discontinued, would you mind removing that? |
| 14:52.48 | ceegee | ups |
| 14:52.55 | ceegee | wrong key, my fault |
| 14:53.16 | [TK]D-Fender | Chainsaw: Sucked anyway ;) |
| 14:54.18 | [TK]D-Fender | ~pb |
| 14:54.19 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:54.22 | [TK]D-Fender | Chainsaw: fixed |
| 14:54.27 | *** join/#asterisk af_ (n=getsmart@88-149-241-49.dynamic.ngi.it) |
| 14:54.39 | Chainsaw | [TK]D-Fender: Thanks :) |
| 14:55.14 | viraptor | does anyone at all use remote attended transfers in a cluster?... |
| 14:55.45 | guax | how cluster are we talking about? |
| 14:55.49 | PMantis | [TK]D-Fender: Well, my conference call ended, and I upgraded to 1.6.1.2... the dnsmgr lines are gone now. Cool! |
| 14:56.18 | *** join/#asterisk Puma1337 (i=Puma1337@ool-44c66019.dyn.optonline.net) |
| 14:56.25 | [TK]D-Fender | PMantis: \o/ |
| 14:56.44 | viraptor | guax: all hosts having the same dialplan, handling the same users, incoming calls load-balanced |
| 14:57.02 | xa0z | No one seems to offer "Unlimited" really with what I want. |
| 14:57.26 | guax | viraptor, usually i made a hard division in load, but no a real cluster. no problemas at all in transfer |
| 14:57.29 | PMantis | [TK]D-Fender: Cool, thanks. |
| 14:57.32 | *** part/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 14:57.38 | guax | not a real* |
| 14:57.56 | Puma1337 | can anyone help me with this: http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/#comments |
| 14:58.29 | Puma1337 | I'm really confused about the macro-stdexten thing |
| 14:58.31 | viraptor | guax: so you never had problems with users wanting to join calls that go out via different asterisks? |
| 14:59.00 | guax | nope |
| 14:59.01 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 14:59.49 | guax | viraptor, i have a 3 machines pbx making a load share between pstn cards and fxs banks and sips, they transfer to one each other |
| 14:59.54 | [TK]D-Fender | [10:56]<xa0z>No one seems to offer "Unlimited" really with what I want. <- Oh you mean, 5$ and spam the universe with 10,000 channels at one? |
| 14:59.55 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 14:59.57 | [TK]D-Fender | once* |
| 15:00.01 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.46 | viraptor | guax: ok, thanks... that doens't look like my scenario |
| 15:00.51 | yziquel | [TK]D-Fender: For the 101 extension that was local and that was saying (NAT), sip show peer 101 gives Nat: RFC3581. |
| 15:01.18 | [TK]D-Fender | Puma1337: Verify the PErMISSIONS on the script and specify the full path to it and look out for how quote chars may need to be ESCAPED |
| 15:02.23 | Puma1337 | [TK]D-Fender: I am confused where to put the macro-stdexten |
| 15:02.24 | *** join/#asterisk many (n=many@dslb-188-098-007-013.pools.arcor-ip.net) |
| 15:02.39 | Puma1337 | When calls come in, they are routed to two different ring groups |
| 15:02.52 | Puma1337 | And I am not sure how to customize the example there |
| 15:02.55 | Puma1337 | or where to put it |
| 15:02.56 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 15:03.39 | Psychobilly | ser on the other hand handles sip text mesages just fine..., lets see how hard will be to configure * as a backend for ser :P |
| 15:03.40 | [TK]D-Fender | Puma1337: My guess would be AFTEr your dial. |
| 15:03.42 | yziquel | [TK]D-Fender: so nat=rfc3581 or nat=never? |
| 15:03.53 | [TK]D-Fender | yziquel: Just set NO in your configs |
| 15:04.25 | yziquel | it's set to no everywhere, and I still get the (NAT) line... |
| 15:04.30 | Puma1337 | [TK]D-Fender: right, but where is that? |
| 15:05.04 | [TK]D-Fender | Puma1337: this is your dialplan.. you tell ME |
| 15:05.15 | [TK]D-Fender | yziquel: Look again. |
| 15:05.37 | Puma1337 | [TK]D-Fender: I am using FreePBX, so I'm not exactly sure. |
| 15:05.40 | [TK]D-Fender | Psychobilly: * as a back-end server for SER is a common large scale solution |
| 15:05.47 | [TK]D-Fender | Puma1337: OH, well then : |
| 15:05.50 | [TK]D-Fender | ~freepbx |
| 15:05.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:05.56 | [TK]D-Fender | Puma1337: GUI's are NOT supported here |
| 15:06.01 | Puma1337 | lol |
| 15:06.03 | Puma1337 | ok |
| 15:06.27 | Puma1337 | Thanks anyways. |
| 15:06.53 | Psychobilly | [TK]D-Fender i know, i just never done that before |
| 15:09.18 | viraptor | does anyone know a scenario where asterisk uses the same callid for 2 different outgoing calls? (parallel forking? queue agents?) |
| 15:09.42 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 15:11.50 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 15:12.30 | [TK]D-Fender | viraptor: Should never happen |
| 15:12.42 | [TK]D-Fender | viraptor: Every call is unique |
| 15:13.17 | yziquel | [TK]D-Fender: got these two lines 'Sending to 192.168.23.101 : 5060 (NAT)' and 'Transmitting (no NAT) to 192.168.23.101:5060:' just one after another. Aren't they contradicting themselves? |
| 15:13.35 | [TK]D-Fender | yziquel: Focus on the ITsp side |
| 15:13.52 | yziquel | [TK]D-Fender: ok. thanks for being helpful. |
| 15:13.58 | [TK]D-Fender | yziquel: And the former looks like a possible localnet issue |
| 15:15.28 | viraptor | [TK]D-Fender: ok, thanks... I know some pbxes do parallel forking just by changing "Record-Route: ... ftag=???" on that hop, so I wanted to make sure |
| 15:15.47 | yziquel | [TK]D-Fender: a localnet issue? I've got sip.conf: 'localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks' |
| 15:16.47 | [TK]D-Fender | yziquel: :/ |
| 15:16.56 | [TK]D-Fender | yziquel: Keep looking on the other side then |
| 15:17.57 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 15:19.08 | xa0z | I need a provider that offers an unlimited plan... multiple channels... and will let me use FreeSwitch |
| 15:19.18 | xa0z | and most of the ones listed, and checked don't offer all the above. |
| 15:19.46 | xa0z | Broadvoice isn't clear about their plans, it's $23 for USA Unlimited, but how much extra for BYOD? *sigh* |
| 15:20.37 | xa0z | They don't specific channels information either. |
| 15:20.43 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:22.11 | [TK]D-Fender | xaobecause unlimited is almost always 1-2 ONLY |
| 15:22.32 | [TK]D-Fender | xa0z: What part of "this isn't a free lunch" don't you get? this shit still costs something |
| 15:25.07 | xa0z | I know this... |
| 15:25.07 | xa0z | But that isn't the problem. |
| 15:25.37 | [TK]D-Fender | xa0z: And there is no extra cost with BV for BYOD. they just won't support it for you. |
| 15:27.22 | Defraz | What is a good device to convert sip to pri |
| 15:27.51 | Defraz | We have a phone system at a remote location that accepts PRI and the people in charge don't want to spend money for new phoens and so on and training. |
| 15:28.21 | Defraz | So we still want to integrate with them and have them use our trunks on our system and I think I can do that with some type of SIP to PRI gateway. |
| 15:28.37 | Defraz | Does anyone have any suggestions of where to look or good brands? |
| 15:28.51 | [TK]D-Fender | Defraz: AudioCodes Mediant |
| 15:29.05 | [TK]D-Fender | defor drop an * box there. |
| 15:29.08 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 15:29.11 | [TK]D-Fender | Defraz: or drop an * box there. |
| 15:29.12 | xa0z | [TK]D-Fender, what I'm saying about BroadVoice that isn't clear is that their "BYOD Plan" say's it's $11.42 and you then choose any of their "Service Plans" |
| 15:29.38 | [TK]D-Fender | xa0z: Call them if you have questions. |
| 15:30.45 | WHYS | could someone please look at this sip trace. I am simply trying to register a Ekiga softphone, and It appears to try to register on multiple interfaces, but not eth0. |
| 15:30.48 | WHYS | http://pastebin.com/d4f1affff |
| 15:31.09 | brah | (1.4.26.1) Why is my var ${EXTENSION} being set as global with Set(Extension=foo)? |
| 15:32.12 | [TK]D-Fender | brah: PASTERBIN <- |
| 15:32.14 | [TK]D-Fender | ~pb |
| 15:32.14 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:32.50 | *** join/#asterisk cryptanthus (n=newview@wsip-72-214-233-12.om.om.cox.net) |
| 15:33.29 | yziquel | [TK]D-Fender: I've got 'Sending to 91.121.167.75 : 5060 (non-NAT)' and 'Reliably Transmitting (no NAT) to 91.121.167.75:5060:' on the remote SIP peer server. |
| 15:33.39 | Stese | WHYS> can you PB your ifconfig? |
| 15:34.07 | yziquel | [TK]D-Fender: I'll be checking the firewall. You said that I should try so that nothing related to SIP is managed by the firewall. |
| 15:34.48 | WHYS | ifconfig: http://pastebin.com/d1f191f2b |
| 15:34.51 | phr3ak | openvox could detect the fax voice every time or only in first several seconds? |
| 15:35.28 | yziquel | [TK]D-Fender: contacted the firewall guys. They said they activated http://www.howtonetworking.com/Routers/ssg1.htm |
| 15:35.35 | yziquel | [TK]D-Fender: deactivating it... |
| 15:36.25 | *** join/#asterisk ketema (n=ketema@71.43.207.50) |
| 15:36.31 | [TK]D-Fender | WHYS: Call seems to arrive on HAM1... |
| 15:36.42 | *** join/#asterisk karlag (n=iseit@host-190-15-166-65.movilmax.com) |
| 15:37.02 | WHYS | just registering, but I don't want it to register on that interface. |
| 15:37.18 | WHYS | It looks like its trying several interfaces |
| 15:37.43 | karlag | im having serious trouble with my asterisk.. when im in the console and i do a reload it is very slow.. |
| 15:37.59 | *** part/#asterisk ketema (n=ketema@71.43.207.50) |
| 15:38.01 | spck | hi guys, i'm trying to get call parking to work correctly, here is a pb of my log: http://pastebin.com/d4efbfbeb, i'm getting the feeling i need to setup user context's better or something |
| 15:38.04 | karlag | the proccess are normal, also the cpu and memory |
| 15:38.05 | WHYS | slow reload could be DNS |
| 15:38.18 | spck | basically my parked call hangs up after 45 seconds. |
| 15:38.21 | karlag | how can that be? |
| 15:38.31 | WHYS | Check your FQDNs to be sure they are correct and more importantly answering |
| 15:38.36 | cryptanthus | I am a new Asterisk user. I'm running Fedora 11. When I run service asterisk start, the system says [ok]. When I try to connect asterisk -vvvr it says that it can't connect. When I try service asterisk status, the response is asterisk dead but subsys locked. |
| 15:38.38 | grandpapadot | spck: pastebin your /etc/aseterisk/features.conf |
| 15:38.52 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 15:39.02 | grandpapadot | cryptanthus: check permissions on /var/run/asterisk* |
| 15:39.14 | karlag | thats weird.. because is only with reload and restart.. |
| 15:39.25 | cryptanthus | grandpapadot: What should the permissions be? |
| 15:39.37 | spck | grandpapadot: http://pastebin.com/d4f0fcb4 |
| 15:39.52 | WHYS | Karlag: I've just went through that problem - or similar. |
| 15:40.13 | karlag | ok, ill check that and let you know... |
| 15:40.28 | WHYS | turned out to be a trunk that was not listening on the correct port |
| 15:40.56 | grandpapadot | spck: See where it says parkingtime => 45 ; ... default is 45 seconds? uncomment and change that to a higher number, restart asterisk (reload won't do it) |
| 15:41.17 | spck | ya i get that but it's still going to hang up on people |
| 15:41.35 | spck | i guess what i don't get is why it's going back to context,s,1 |
| 15:41.42 | spck | instead of just ringing the extension back |
| 15:41.44 | cryptanthus | grandpapadot: I have 755 /var/run/asterisk, and the user and group are asterisk. |
| 15:41.46 | WHYS | [TK]D-Fender: any ideas whys Ekiga would try many interfaces, but not eth0? |
| 15:42.10 | grandpapadot | cryptanthus: the user connecting has to have write permissions on the asterisk.ctl file |
| 15:42.16 | [TK]D-Fender | cryptanthus: And who are you logged in as? |
| 15:42.38 | cryptanthus | [TK]D-Fender: root |
| 15:42.54 | grandpapadot | spck: it will send the call back to the context it came from, afaik, can you pastebin the console output (verbosity set to 3) |
| 15:43.01 | [TK]D-Fender | cryptanthus: Just because * service starts doesn't mean its working. it could be crashing in circles |
| 15:43.11 | [TK]D-Fender | cryptanthus: Do calls process normally? |
| 15:43.38 | cryptanthus | [TK]D-Fender: I haven't even got it to start yet. |
| 15:44.21 | spck | http://pastebin.com/d4efbfbeb i think that has more then verbosity 3 |
| 15:44.38 | [TK]D-Fender | cryptanthus: Then kill it and start it manually and see what happens |
| 15:45.05 | grandpapadot | spck: pastebin your "default" context |
| 15:46.12 | spck | ok here's something different i changed features.conf to say comebacktoorgin=yes, now here's the console: http://pastebin.com/d1effeb68 |
| 15:46.24 | spck | basically it's trying to call 830|30|tk |
| 15:46.44 | spck | it's using the old format with the |'s |
| 15:46.59 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.29 | cryptanthus | [TK]D-Fender: It's not running. However I will delete the asterisk.ctl and asterisk.pid files then restart asterisk using /usr/sbin/asterisk Is this what your recommending? |
| 15:48.00 | spck | ideally i want the parked call to ring back to the person who parked |
| 15:48.03 | [TK]D-Fender | cryptanthus: No. I'm recommending you kill the script thats trying to start *. SU to asterisk. and start it MANUALLY |
| 15:48.07 | spck | but i was running into the above error beofre |
| 15:48.59 | brah | Is this a good way to approach the problem or is there a better one? If($ [${EXISTS(${FOO})} = 1] ?1:3) |
| 15:49.32 | cryptanthus | [TK]D-Fender: I don't understand. What do you mean kill the script that's trying to start Asterisk. I was using the Fedora service start asterisk. Could you help me with the procedure of what your recommending. |
| 15:51.09 | *** join/#asterisk acxty (n=acxty@201.220.136.117) |
| 15:51.21 | spck | grandpapadot: there isn't really anything in my default context |
| 15:52.13 | grandpapadot | spck: k, that may be why it's failing to send the call to default,s |
| 15:52.41 | [TK]D-Fender | cryptanthus: "service asterisk stop". Verify that its gone. "ps -A" |
| 15:54.22 | cryptanthus | [TK]D-Fender: service asterisk stop ---> [FAILED] , ps -A | grep asterisk ---> no output |
| 15:54.49 | spck | grandpapadot: is this one failing because its trying to use the old format? http://pastebin.com/d1effeb68 |
| 15:55.00 | spck | i.e. the |'s and not ,'s |
| 15:55.10 | spck | nm |
| 15:55.44 | [TK]D-Fender | cryptanthus: Stop grep-ing, and look at EVERYTHING |
| 15:56.41 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:56.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:57.19 | cryptanthus | [TK]D-Fender: Alright. I inspected the output. I don't see anything unusual. |
| 15:57.23 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 15:57.59 | [TK]D-Fender | cryptanthus: Neither do I |
| 15:59.24 | cryptanthus | [TK]D-Fender: Alright. So whats the proper procedure for restarting asterisk manually. Don't I just run /usr/sbin/asterisk ? |
| 15:59.45 | [TK]D-Fender | cryptanthus: asterisk -gvvvvvvvvvvc |
| 16:00.01 | cryptanthus | [TK]D-Fender: as root or as su asterisk |
| 16:00.05 | bpgoldsb | Anyone using Asterisk Realtime for SIP information and want to answer a few questions? |
| 16:00.25 | [TK]D-Fender | cryptanthus: I'd confirm what your init was trying and do that first |
| 16:05.46 | spck | am i correct in thinking that this shouldn't work in * 1.6: -- Executing [SIP0830@park-dial:1] Dial("SIP/1140-09564c70", "SIP/830|30|tk") in new stack |
| 16:05.46 | spck | ? |
| 16:06.44 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:07.34 | cryptanthus | [TK]-D-Fender: I looked at the init script, It appears to run asterisk as usr: asterisk grp:asterisk. However, I could not su asterisk. So I decided to run it as root using the command asterisk -gvvvvvvvvvvc as you recommended. When I did this my terminal turned grey output a bunch of text and ended with Asterisk ready. I went to another terminal and typed (as root) service asterisk status, the output was [running]. This seems to |
| 16:07.34 | cryptanthus | <PROTECTED> |
| 16:08.24 | Qwell | cryptanthus: man asterisk - look at -U and -G |
| 16:09.27 | Qwell | bpgoldsb: only if you ask a question |
| 16:10.28 | bpgoldsb | Qwell, I'm trying to move my existing sip.conf into using realtime. I'm confused as to if I need to do anything special to my sip.conf aside from removing peer information |
| 16:10.59 | cryptanthus | [TK]D-Fender: I accidentally typed your name wrong above. You may not have seen the previous note highlighted. ... This could also be a permission problem because root would have complete access. |
| 16:11.32 | bpgoldsb | I've put some sip peers in the database, but it doesn't appear to recognize them. However, I verified it is connecting to the database just fine. |
| 16:11.35 | frieze | I there any particularly good site to check for reviews of wireless SIP phones? especially with a view towards use with asterisk? |
| 16:12.32 | Qwell | frieze: general consensus - they all suck |
| 16:12.46 | Qwell | most people would recommend getting a SIP DECT phone |
| 16:13.00 | Qwell | (or $10 wireless phone on an ATA or something) |
| 16:13.30 | ariel_ | has anyone had issues that hey can't do a kill (PID#) on asterisk? |
| 16:13.40 | clemahieu | When I started reading: (or $10 |
| 16:13.44 | clemahieu | I thought you were writing code. |
| 16:13.58 | clemahieu | Coding brain-fry. |
| 16:14.48 | frieze | Qwell: yeah, thought I was missing something. Can I use DECT here in the US? |
| 16:15.36 | ariel_ | For wireless phones I have been happy and using the Polycom 8020 and 8030, they work great with there SVP server. |
| 16:16.39 | Micc_ | Is there a way to get aastra phones to use a meetme conference room when they use the conf key? So if the aastra phone hangs up the other two can still talk? |
| 16:17.22 | *** part/#asterisk sack (n=sack@231.Red-81-32-166.dynamicIP.rima-tde.net) |
| 16:17.33 | frieze | ariel_: for a small deployment is the server really necessary? |
| 16:17.38 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 16:17.51 | asteriskmonkey | is there a way of clearing call-limit counters in 1.6? |
| 16:18.05 | Stese | Qwell > Do you have any examples of SIP DECT phones I could look at? |
| 16:18.10 | Qwell | Stese: nope |
| 16:18.23 | [TK]D-Fender | Stese: www.polycom.com |
| 16:18.30 | Stese | thanks :) |
| 16:19.19 | *** join/#asterisk yziquel (i=55da5c63@gateway/web/freenode/session) |
| 16:19.32 | frieze | ariel_: I used to have a weird problem with the write permissions where asterisk was storing its PID so the service wouldn't die. I could still kill manually though |
| 16:21.31 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 16:22.55 | cryptanthus | [TK]D-Fender: Alright. Now I ran as root: asterisk -U asterisk -G asterisk -gvvvvvvvvvvc the output is error logger.c unable to create event log Permission Denied. |
| 16:23.15 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 16:23.25 | *** join/#asterisk ta^3 (n=tacvbo@67.201.69.2) |
| 16:23.36 | [TK]D-Fender | ~asterisk-non-root |
| 16:23.37 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 16:23.39 | [TK]D-Fender | cryptanthus: ^^^^^^^ |
| 16:25.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:27.11 | cryptanthus | [TK]D-Fender: I changed the user and group to asterisk on /var/log/asterisk/event_log , messages, queue_log ... asterisk started fine. |
| 16:27.45 | *** join/#asterisk eliyahud (n=eliyahu@77.126.64.188) |
| 16:28.09 | cryptanthus | [TK]D-Fender: Thanks for your help. I appreciate it. |
| 16:29.16 | Maxxed | is there a way to log an agent in via the CLI or manager interface? |
| 16:29.27 | Maxxed | i see agent logoff |
| 16:29.34 | Maxxed | but havent figured the login part yet |
| 16:30.41 | [TK]D-Fender | Maxxed: How do agents "log in"? |
| 16:30.53 | Maxxed | dial an extention on the hand set |
| 16:31.00 | [TK]D-Fender | Maxxed: that does WHAT? |
| 16:31.08 | Maxxed | puts them in a callqueue |
| 16:31.12 | [TK]D-Fender | .... |
| 16:31.18 | [TK]D-Fender | Maxxed: HOW? |
| 16:31.43 | Maxxed | <PROTECTED> |
| 16:32.18 | [TK]D-Fender | maxBetter. go mak another exten that uses variables instead of CALLERID(), and Originate a call to it |
| 16:33.27 | Maxxed | ey? |
| 16:35.11 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 16:36.03 | Maxxed | eh fuck it, il figure it out :p |
| 16:36.32 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 16:36.54 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 16:37.28 | timeshell_atwork | [TK]D-Fender According to http://www.voipstore.com/2009/06/trixbox-ce-28-released/ Trixbox doesn't support HUDLite |
| 16:37.39 | viraptor | Maxxed: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+AgentCallBackLogin ? |
| 16:37.41 | timeshell_atwork | Nonetheless, I'm trying to make it work anyway |
| 16:37.43 | timeshell_atwork | :p |
| 16:38.06 | timeshell_atwork | However, there's no option in hudlites context.xml file to specify authentication info to connect to asterisk 5038 |
| 16:39.14 | timeshell_atwork | Any idea what it is? Apparently previous versions of hudlite supported 5038 for authentication |
| 16:39.17 | ScarEye | [TK- |
| 16:39.24 | *** join/#asterisk sjobeck (n=Adium@69-30-99-139.dv1sn.easystreet.com) |
| 16:39.35 | ScarEye | [TK]D-Fender: You got a minute bro? |
| 16:40.16 | *** part/#asterisk sjobeck (n=Adium@69-30-99-139.dv1sn.easystreet.com) |
| 16:40.42 | timeshell_atwork | Maxxed AgentCallBackLogin is deprecated |
| 16:40.51 | timeshell_atwork | YOu shoulduse the the new method. |
| 16:41.56 | Maxxed | whats the new method? |
| 16:42.18 | ScarEye | Anyone know of any good *CHEAP* cordless handsets ? |
| 16:42.19 | timeshell_atwork | Maxxed http://leifmadsen.wordpress.com/tag/addqueuemember/ |
| 16:42.32 | timeshell_atwork | ScarEye LMAO!! |
| 16:42.35 | ScarEye | =) |
| 16:42.41 | Maxxed | bitchin |
| 16:42.45 | timeshell_atwork | None are good and none are cheap |
| 16:42.49 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:43.02 | ScarEye | really |
| 16:43.08 | ScarEye | none are good? |
| 16:43.15 | timeshell_atwork | I use SNOM M3 right now |
| 16:43.17 | timeshell_atwork | It's ok... |
| 16:43.23 | ScarEye | what about this. http://www.voip-info.org/wiki/view/480i+CT+Cordless |
| 16:44.13 | timeshell_atwork | ScarEye http://www.snom.com/en/products/snom-m3-voip-phone/ |
| 16:44.38 | timeshell_atwork | Not really cheap though |
| 16:44.45 | ScarEye | how much? |
| 16:44.48 | ScarEye | roughly? |
| 16:47.07 | *** join/#asterisk eliyahud (n=eliyahu@77.126.64.188) |
| 16:48.58 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 16:50.48 | Defraz | can you use wildcards with the database command or regexp? |
| 16:51.16 | ScarEye | Anyone know of any good voip companies that offer at least 2 channels and BYOD ? |
| 16:51.29 | ScarEye | for a business? |
| 16:51.31 | Defraz | I want to delete the cidname where number = null. |
| 16:52.44 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:53.29 | thehar | Defraz: no you cannot |
| 16:53.56 | ScarEye | dude, this is great. http://www.callcentric.com/dids/office_unlimited |
| 16:54.24 | ScarEye | anyone use callcentric here? |
| 16:54.54 | [TK]D-Fender | [12:35]<Maxxed>eh fuck it, il figure it out :p <- I just handed you a completely functional idea |
| 16:55.51 | [TK]D-Fender | ScarEye: Aastra handset is tied to the base phone. It will never be a "distinct" identity and will ring on the bas as well |
| 16:56.08 | *** join/#asterisk n3td3v (n=n3td3v@42.pool85-56-142.dynamic.orange.es) |
| 16:56.17 | n3td3v | Hey! |
| 16:56.22 | [TK]D-Fender | ScarEye: For places you really need wireless either go analog + ATA, or something like a Polycom DECT base |
| 16:56.42 | n3td3v | Hey! |
| 16:57.08 | ScarEye | DECT base will work with analog handset that is DECT compatible? |
| 16:59.02 | *** join/#asterisk nightrid3r (n=borgirc-@78-20-232-172.access.telenet.be) |
| 17:00.35 | *** join/#asterisk Psychobilly (n=moi@adsl176-124.kln.forthnet.gr) |
| 17:01.14 | acxty | HI guys, I will but some polycom phones. I have asterisk working. The idea is to have the phones connected to a separate patch panel -> switch -> pc(centos running asterisk). Do I need to buy special kind of patch panels or switch. Centos machines also have a dhcpd server running on it + sharing internet connection |
| 17:01.20 | acxty | *uy |
| 17:01.25 | acxty | *buy sorry |
| 17:02.11 | *** part/#asterisk xa0z (n=Interex@75-129-230-28.dhcp.mtvr.il.charter.com) |
| 17:03.13 | *** join/#asterisk old_monk (n=borgirc-@78-20-232-172.access.telenet.be) |
| 17:03.37 | acxty | I have a 24 ports patch panel + a linksys sr224 10/100 24 ports, that I used for the old lan |
| 17:03.47 | ScarEye | acxty: You should be good. |
| 17:03.51 | acxty | will the polycom phones work with that one? |
| 17:04.38 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
| 17:04.40 | acxty | ok, thanks |
| 17:05.34 | shimi | hey all. I am trying to understand: isn't exten => _.#,n,NoOp(Number ends with a hash sign) supposed to work? |
| 17:06.54 | Defraz | I think I might have a corrupt asterisk db |
| 17:07.02 | Qwell | shimi: no, nothing can be after . |
| 17:07.03 | Defraz | is there a way to rebuilt it ? |
| 17:07.25 | shimi | that explains it. so how do I match "all numbers ending with pound" ? |
| 17:08.23 | aurax | is there a way i can know on which channel my provider send signaling on E1? |
| 17:08.41 | shimi | aurax, isn't it quite standard ? |
| 17:09.09 | shimi | span=1,1,0,ccs,hdb3,crc4 |
| 17:09.09 | shimi | bchan=1-15,17-31 |
| 17:09.09 | shimi | dchan=16 |
| 17:09.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:09.35 | shimi | this setting works for bezeq, hot, orange and cellcom (and i tried it personally ;)) |
| 17:11.31 | *** join/#asterisk anthm][ (n=anthm@67.201.69.2) |
| 17:12.16 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:12.50 | shimi | Qwell, would exten => _X# be correct, then ? |
| 17:12.59 | Qwell | sure, if it's 1 digit |
| 17:13.38 | shimi | I need an unknown number of digits. basically i want to strip off # from the end of numbers |
| 17:13.52 | shimi | exten => _X!# maybe ? |
| 17:13.57 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 17:14.01 | Qwell | # shouldn't ever really get into your dialplan.. |
| 17:14.08 | Qwell | shimi: no, nothing can go after ! either |
| 17:14.16 | shimi | wonderful :) |
| 17:14.31 | shimi | any wildcard out there CAN have anything else after it? :) |
| 17:14.42 | Qwell | no |
| 17:14.54 | shimi | and what do you mean shouldn't ever really get into my dialplan? |
| 17:15.00 | shimi | how else can I handle this key ? |
| 17:16.26 | *** join/#asterisk errotan (n=errotan@5403E455.catv.pool.telekom.hu) |
| 17:16.30 | shimi | so different lines of: _[*0-9]# and _[*0-9][*0-9]# and _[*0-9][*0-9][*0-9]# (continue 20 repeats...) would do? |
| 17:18.05 | Juggie | something very very silly? :) |
| 17:18.42 | Juggie | sounds like the attempt was to trap any digit or * up to 20 long, terminated by a # |
| 17:18.53 | shimi | that's correct |
| 17:19.01 | shimi | if you have a more clever way, i am happy to hear :) |
| 17:19.12 | Qwell | Why is # getting into your dialplan in the first place? |
| 17:19.28 | timeshell_atwork | When you add a new user to manager.conf, do you need to shut down and restart asterisk for it take effect? |
| 17:19.31 | shimi | because a dumb app which is a black box that I do not control, appends it to its dials. |
| 17:19.36 | Juggie | timeshell_atwork, no. |
| 17:19.39 | shimi | timeshell_atwork, reload manager |
| 17:19.41 | [TK]D-Fender | shimi: What is the call coming in as? or is this for input from an IVR? |
| 17:19.52 | Juggie | [TK]D-Fender, i'd bet ivr input |
| 17:19.55 | shimi | it's dialed inside a DISA app |
| 17:20.31 | Juggie | shimi, there are a number of better ways you could do it. |
| 17:20.39 | Puma1337 | can anyone tell me why I would get this error [2009-08-05 12:12:44] WARNING[13989] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 17:20.44 | [TK]D-Fender | shimi: Either do your 20 pattern method, or use Read() instead and feed it a dialtone recording. |
| 17:20.50 | Puma1337 | [2009-08-05 12:12:44] VERBOSE[13989] logger.c: == Everyone is busy/congested at this time (3:0/0/3) |
| 17:21.05 | [TK]D-Fender | Puma1337: Because * has no idea how to reach the device you are calling. |
| 17:21.07 | shimi | Puma1337, the SIP ext you're dialing to is probably not registered |
| 17:21.12 | [TK]D-Fender | ^^^ |
| 17:21.38 | shimi | I could also match everything and use an if to check the last digit |
| 17:21.39 | *** join/#asterisk propellerhead (n=yogurt2u@host144.190-30-188.telecom.net.ar) |
| 17:21.45 | Puma1337 | Even though it was ringing 2 minutes before? |
| 17:21.54 | shimi | Puma1337, perhaps it's behind a NAT ? |
| 17:22.36 | shimi | I don't know how to check "what is the last digit", though. I know how to take the 1st, 2nd, 4th-7th, but I don't know the syntax to get the last digit from ${EXTEN} ... |
| 17:22.39 | Juggie | [TK]D-Fender, what about just doing a wide open pattern and then checking it in logic/regex afterwards to ensure its valid |
| 17:23.25 | [TK]D-Fender | Juggie: Depends |
| 17:23.38 | [TK]D-Fender | shimi: And if # isn't the last digit? |
| 17:23.53 | shimi | continue as usual |
| 17:23.58 | shimi | gotoif... |
| 17:24.08 | Puma1337 | shimi, if it is behind a nat what do you suggest? |
| 17:24.33 | *** join/#asterisk subl (i=sublime@sublime.xmission.com) |
| 17:24.37 | shimi | Puma1337, so your NAT router may have forgot the session between your phone and your asterisk |
| 17:24.43 | [TK]D-Fender | shimi: Continue where? Doing what? |
| 17:24.59 | shimi | [TK]D-Fender, ok, I want to put this inside the context that currently processes the DISA |
| 17:25.10 | shimi | [TK]D-Fender, right now, it just makes the call as if it was from an internal phone, it works ok |
| 17:25.20 | [TK]D-Fender | Puma1337: see in #freepbx |
| 17:25.31 | *** part/#asterisk subl (i=sublime@sublime.xmission.com) |
| 17:25.40 | Juggie | shimi, if ${EXTEN} = 1234# to get # you would do ${EXTEN:-1:1) |
| 17:25.44 | shimi | [TK]D-Fender, the problem is, that with the automated system, a # is appended to the dialed number, and then, of course, it won't dial |
| 17:25.52 | [TK]D-Fender | shimi: Just use READ, or make your own IVR to collect digits. |
| 17:25.59 | shimi | but it's not an IVR |
| 17:26.01 | shimi | it's a DISA |
| 17:26.09 | [TK]D-Fender | shiMAKE ONE INSTEAD |
| 17:26.16 | [TK]D-Fender | shimi: DISA = useless anyway |
| 17:26.24 | Juggie | ya, just use an ivr disa is lame. |
| 17:26.42 | shimi | it is working ok for me :) |
| 17:26.47 | shimi | and I don't see the difference |
| 17:26.58 | Juggie | shimi, if you have 12345# and you want to cut off the # |
| 17:27.02 | [TK]D-Fender | shimi: DISA is a IVR with no failover, less control and only makes dialtone "easy" (whoopteedo.... you can make your own in no time flat...) |
| 17:27.16 | shimi | I really just need to strip the # |
| 17:27.23 | shimi | everything else works perfect |
| 17:27.26 | shimi | let see. |
| 17:27.42 | [TK]D-Fender | shimi: Feel free to do your 20 length iterations then |
| 17:27.59 | shimi | but I can use the if |
| 17:28.06 | Juggie | you'll want to do something like ${EXTEN:0:${LEN(${EXTEN})}-1} or something like that. thats from memory :) |
| 17:28.12 | shimi | right, that's my plan |
| 17:28.16 | shimi | or to use CUT() |
| 17:28.29 | *** join/#asterisk jtodd (i=p6lyud36@ns.fox-den.com) |
| 17:28.29 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:28.34 | shimi | another q, if i may |
| 17:28.34 | Juggie | jtodd! |
| 17:29.13 | shimi | once an exten => is matched. does all the dialplan needs to be with the same pattern ? or will every line that continues to match, even if the pattern is different, will match and execute ? |
| 17:29.26 | Juggie | it needs to be the same pattern |
| 17:29.54 | shimi | so that's the catch :D |
| 17:30.42 | Juggie | you can break out of it w/ a goto or whatever |
| 17:30.51 | Juggie | but you are really better off not matching 20 seperate patterns |
| 17:31.14 | *** join/#asterisk lizone (n=zenst@user-0ccejib.cable.mindspring.com) |
| 17:32.13 | n3td3v | hi! |
| 17:36.09 | *** join/#asterisk subl (n=sublime@2001:470:1f0f:da:224:1dff:fe1e:15c) |
| 17:36.34 | frieze | has anyone written a web frontend for ARI that will work with polycom's xhtml on a small screen? |
| 17:37.16 | *** join/#asterisk hammerzone45 (n=hammer15@c-71-229-108-12.hsd1.fl.comcast.net) |
| 17:38.04 | hammerzone45 | hi have a problem with my queues login people out automatically ... is asterisk going to log out a member that reporst status = CHANUNAVAIL? |
| 17:40.24 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 17:42.07 | Defraz | if I delete the astdb file and restart asterisk will it recreate that file? |
| 17:42.10 | Defraz | from the config files? |
| 17:43.25 | *** join/#asterisk Puma1337 (i=Puma1337@ool-44c66019.dyn.optonline.net) |
| 17:43.45 | [TK]D-Fender | Defraz: It will make a new blank one |
| 17:43.52 | [TK]D-Fender | defit does not come from config files |
| 17:44.00 | Puma1337 | [TK]D-Fender, i posted a link to pastebin in #freepbx for you |
| 17:45.59 | Defraz | so asterisk doesn't create that after it reloads the config files? |
| 17:46.05 | Defraz | How would I generate a new one? |
| 17:48.23 | n3td3v | hi!! |
| 17:49.41 | shimi | how do I cut the last digit in ${EXTEN:<whatever>} format out ? |
| 17:50.19 | Corydon76-dig | depends upon what version you're using |
| 17:50.27 | [TK]D-Fender | defNothing in there COMES from configs. |
| 17:50.36 | shimi | unfortunately 1.2 right now |
| 17:50.36 | Corydon76-dig | If you're using 1.6.x, ${EXTEN:whatever:-1} |
| 17:52.10 | Corydon76-dig | then you have to go with ${EXTEN:whatever:$[${LEN(${EXTEN})} - 1]} |
| 17:52.11 | Corydon76-dig | or no... |
| 17:52.11 | Corydon76-dig | then you have to go with ${EXTEN:whatever:$[${LEN(${EXTEN})} - 1 - whatever]} |
| 17:52.20 | Corydon76-dig | See why 1.6.x is so much better? |
| 17:52.25 | shimi | :) |
| 17:53.09 | shimi | what is "whatever" ? |
| 17:53.48 | Corydon76-dig | A number, based upon how much you want to trim from the front |
| 17:54.02 | shimi | so 0 if i want the whole number |
| 17:54.07 | shimi | besides the last digit |
| 17:54.27 | Corydon76-dig | Right |
| 17:57.55 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:01.03 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
| 18:01.08 | shimi | damn router :) |
| 18:02.25 | *** join/#asterisk QaDeS (n=mklaus@p4FC72D49.dip0.t-ipconnect.de) |
| 18:02.38 | *** part/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 18:02.59 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:04.44 | *** join/#asterisk vector_xyz (n=vecy@unaffiliated/t3rminator) |
| 18:05.54 | vector_xyz | hey guys i know this is not an phone only channel but i dont know where else to ask for help - i ordered a landline etc... i have a modem and it supports 'Caller ID' however any software i try 'Caller Id Software' to see on my screen who is calling does not seem to work; one of them does but it doesnt show any numbers and shows once in a while ... im confused what am i doing wrong, everything is hooked up right. |
| 18:08.40 | vector_xyz | or what application can i use to determine truly the capabilities of the modem... like does it support TAPI |
| 18:08.48 | *** join/#asterisk alphanet (n=ircuser@shakotay.alphanet.ch) |
| 18:09.54 | alphanet | is it possible to use Skype with Asterisk (I heard about a skype channel, but is it really functionnal?) |
| 18:10.32 | Qwell | ~skypeforasterisk |
| 18:10.33 | infobot | [~skypeforasterisk] is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.astricon.net/skype for beta details. |
| 18:10.46 | Qwell | hmm, need to update that url |
| 18:10.53 | [TK]D-Fender | vector_xyz: Your modem is worthless with * |
| 18:11.57 | Qwell | infobot: skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://store.digium.com/productview.php?product_code=804-00019 for details on the open beta. |
| 18:11.58 | infobot | ...but skypeforasterisk is already something else... |
| 18:12.02 | Qwell | infobot: no, skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://store.digium.com/productview.php?product_code=804-00019 for details on the open beta. |
| 18:12.03 | infobot | Qwell: okay |
| 18:12.06 | *** join/#asterisk errotan (n=errotan@5403E455.catv.pool.telekom.hu) |
| 18:13.02 | asteriskmonkey | anyone know of how to clear a sip user max call-limit variable |
| 18:13.21 | asteriskmonkey | im running into issues in 1.6.x where its not being cleared |
| 18:14.19 | Qwell | what do you mean not being cleared? |
| 18:14.39 | Qwell | the call-limit isn't going to change while Asterisk is running unless you change the config and reload |
| 18:16.23 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 18:18.40 | vector_xyz | so no one knows how i can figure out what my modem is capable of |
| 18:19.06 | asteriskmonkey | Qwell: i have issues when i set call-limit=1 sometimes it sticks at 1 on some clients even when there is no usuage |
| 18:19.17 | asteriskmonkey | there has to be a better way of clearing out stale limits |
| 18:19.46 | shimi | Corydon76-dig, thanks a lot dude |
| 18:19.56 | shimi | Aug 5 21:19:22 VERBOSE[3719] logger.c: -- Executing NoOp("SIP/gsmgateway-09dcb620", "Number ends with #| going back to from-internal with ext: 7777") in new stack |
| 18:19.57 | shimi | Aug 5 21:19:22 VERBOSE[3719] logger.c: -- Executing Goto("SIP/gsmgateway-09dcb620", "from-internal|7777|1") in new stack |
| 18:20.01 | shimi | :) |
| 18:20.21 | [TK]D-Fender | vector_xyz: It is not supported by * |
| 18:21.02 | coppice | I remember when manuals were a good way to find what your stuff was capable of |
| 18:21.23 | asteriskmonkey | vector_xyz: us at commands |
| 18:21.33 | asteriskmonkey | vector_xyz: http://www.computerhope.com/help/modem.htm |
| 18:22.04 | asteriskmonkey | but yes.. modems not generally supported in asterisk unless you hack one into an x100p.. at that point though your really just wasting time |
| 18:22.15 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 18:22.29 | *** join/#asterisk Gnarfy (n=mirc@213.144.244.115) |
| 18:28.27 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 18:29.54 | n3td3v | hi |
| 18:30.47 | bpgoldsb | I'm setting up Asterisk Realtime for my SIP and probably voicemail. Is there a community written web frontend for this or do I need to look at rolling my own? |
| 18:31.09 | bpgoldsb | For adding/updating/removing sip peers/users, that is |
| 18:31.55 | n3td3v | hi |
| 18:31.56 | n3td3v | hi |
| 18:35.31 | citywok | when we make an outbound call, we have a button that lets an agent leave a pre-recorded voicemail on a persons voicemail box. to do this i'm using the redirect command which does mostly what i want, except that when you redirect a call you are unable to send a variable to it, and all variables from teh pre-existing channel disappear. any ideas? |
| 18:38.31 | *** join/#asterisk brezular (n=brezular@adsl-dyn149.91-127-22.t-com.sk) |
| 18:43.07 | citywok | how do people handle CDR's for calls that get transferred? wiping the variables from the call kinda breaks everything lol |
| 18:44.34 | *** join/#asterisk moy (n=moy@67.201.69.2) |
| 18:44.49 | *** join/#asterisk ta^3 (n=tacvbo@67.201.69.2) |
| 18:46.57 | asteriskmonkey | ResetCDR :P |
| 18:47.16 | asteriskmonkey | what version of asterisk? cdrs are handle different by different versions |
| 18:47.26 | citywok | i'm in 1.4 testing, but moving to 1.6 |
| 18:47.40 | asteriskmonkey | start with 1.6 |
| 18:47.47 | asteriskmonkey | drop testing with 1.4 |
| 18:47.55 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:48.17 | citywok | yea, i ran into a few issues and am finishing getting my 1.6 environment going atm |
| 18:48.45 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 18:49.20 | citywok | unforatunetly, i execute the redirect from the AMI, and by then the cdr is already wiped |
| 18:49.40 | citywok | would i execute forkcdr or resetcdr from teh ami, then do my redirect? |
| 18:52.36 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-22e2e921f12eb219) |
| 19:02.15 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 19:03.43 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 19:09.19 | lesouvage | Is there a silence detection available so a message can be played by asterisk after X seconds of silence on the callee side so a message to be delivered will be completely recorded and start playing before the "beep" so part of the message is not recorded. |
| 19:09.45 | lesouvage | and start= and not start |
| 19:10.25 | *** join/#asterisk yziquel (i=55da5c63@gateway/web/freenode/x-e7897db3d020ce09) |
| 19:11.12 | [TK]D-Fender | lesouvage: "core show application waitforsilence" <_ ? |
| 19:12.11 | lesouvage | [TK]D-Fender: thanks! |
| 19:13.42 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:13.43 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:16.38 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:20.22 | hammerzone45 | does asterisk v1.4.26 still works with zaptel? |
| 19:20.30 | Qwell | hammerzone45: yes |
| 19:20.40 | Qwell | though it's recommended to upgrade to dahdi |
| 19:23.04 | *** join/#asterisk jtodd (i=l3jafar8@ns.fox-den.com) |
| 19:23.04 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:24.41 | hammerzone45 | Qwell: thanks |
| 19:25.34 | hammerzone45 | if i want to upgrade/downgrade asterisk do i have to also recomplie zaptel or just asterisk? |
| 19:26.19 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 19:26.19 | [TK]D-Fender | hammerzone45: Just * |
| 19:27.07 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 19:28.18 | hammerzone45 | D-Fender: tnx |
| 19:29.00 | errr | anyone used any USB phones with Bria Pro? |
| 19:29.32 | citywok | errr i'm playing with bria pro & a non usb headset -- what did you want to know? |
| 19:30.27 | errr | citywok: I wanted to know about bria pro with usb phones. Im using the same thing as you currently and I have users who will not use a headset so Im trying to find out how well using bria pro with a usb phone works |
| 19:30.53 | citywok | oh, like a usb PHONE, not usb headphones. lol, interesting. no idea |
| 19:31.17 | errr | yeah their admin guide says they support it but I have never heard of doing it like that |
| 19:31.19 | citywok | i mean i can understand why you would want to give up the convenience, that makes lots of sense! hah. |
| 19:31.44 | citywok | buy them some real sip phones if what they want is a desk phone |
| 19:31.51 | mmattice | does anybody have t-mobile's smsc working? |
| 19:31.52 | errr | we have aastra 55i |
| 19:32.21 | errr | citywok: the thing is times are tough and phones arent cheap, so Im trying to find more ways to save us money |
| 19:32.30 | *** part/#asterisk hammerzone45 (n=hammer15@c-71-229-108-12.hsd1.fl.comcast.net) |
| 19:33.00 | errr | Im 100% happy making eveyone use a headset and bria pro, but some of the women and even a couple of the other guys in IT flat refused |
| 19:33.56 | citywok | personally i love headsets, and so does our entire company, but we are a call center heh :) |
| 19:34.27 | errr | I love the headset too |
| 19:34.45 | errr | I wouldnt go back to a hardphone unless they gave me a raise for doing it |
| 19:35.48 | errr | citywok: do yall use AD there too and take advantage of the LDAP directory tools in bria pro? |
| 19:36.30 | citywok | i'm still just testing it, i dont think we're going to purchase bria pro for our uses |
| 19:36.38 | errr | ah |
| 19:36.48 | citywok | i was going to until i found this gem: http://www.zoiper.com/activex.php |
| 19:37.07 | errr | I didnt like zoiper, I tried it before I tried xlite |
| 19:37.23 | citywok | for our agents, it's perfect |
| 19:37.34 | citywok | they are workign with me to fix the auto-answer bug that i can't seem to get resolved right now |
| 19:37.49 | citywok | but for $900 we get an unlimited use license for it and can run it instead of bria pro at $45/lic |
| 19:39.48 | *** join/#asterisk moy (n=moy@c-98-193-24-21.hsd1.il.comcast.net) |
| 19:39.58 | [TK]D-Fender | err.... |
| 19:40.01 | [TK]D-Fender | ~savemoney |
| 19:40.02 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 19:41.24 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-142-149-133.stat.centurytel.net) |
| 19:41.28 | jeff_phillips | hello |
| 19:41.29 | citywok | [TK]D-Fender: i can also embed it into our web-driven agent interfaces, and not have to install it at any computers, have it autoconfigure itself, and serve every feature i need |
| 19:41.53 | cryptanthus | I'm trying to get my first Asterisk system running. I have everything working and can call in and get the sample greeting success message. I'm working on sip.conf. I have a Aastra 6757i phone. It won't register with Asterisk. Netstat shows that there isn't anything connected to port 5060. |
| 19:42.11 | citywok | which means if i have an agent that needs to work from home, i dont actually have to do anything other than get them a headset, they can do everything from the web :) |
| 19:42.30 | citywok | staff softphones will probably be bria or bria pro though :) |
| 19:42.40 | [TK]D-Fender | cryptanthus: Show us your "netstat" attempt.... |
| 19:42.42 | [TK]D-Fender | ~pb |
| 19:42.43 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 19:42.44 | [TK]D-Fender | ^^^^^6 |
| 19:42.50 | jeff_phillips | I'm trying to get Asterisk Timeclock working. Followed instructions on http://www.asterisktimeclock.org/node/8 |
| 19:43.02 | [TK]D-Fender | cryptanthus: Then go to * CLI, do "sip set debug", and restart the phone |
| 19:43.36 | jeff_phillips | the php timeclock web interface is working fine. the extension 9990 that their file created just goes to a fast busy signal though. can't seem to figure out why |
| 19:44.28 | Qwell | jeff_phillips: without seeing dialplan output, there isn't much we can do to help.. |
| 19:44.32 | Qwell | ~pastebin |
| 19:44.33 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 19:45.04 | citywok | with ResetCDR, it isn't a manager command so i can't execute it from the console, how would i do it from the AMI on a specific channel? or how would i trigger ResetCDR when doing a Command: Redirect from the AMI? |
| 19:45.16 | jeff_phillips | qwell: http://www.pastebin.ca/index.php |
| 19:45.20 | jeff_phillips | oops |
| 19:45.20 | [TK]D-Fender | lol |
| 19:45.24 | cryptanthus | [TK]D-Fender: http://pastebin.com/m2f1207a |
| 19:45.27 | jeff_phillips | http://pastebin.ca/1519598 |
| 19:45.38 | jeff_phillips | there that's more like it |
| 19:45.56 | [TK]D-Fender | cryptanthus: ALL of it, including the command called |
| 19:46.09 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 19:46.10 | Qwell | Goto(s|1) ? |
| 19:46.17 | Qwell | it's...in a different context |
| 19:46.41 | lesouvage | What is the best application to use to detect an answering machine on an outbound call? AMD() , MachineDetect() or some other application? |
| 19:46.44 | [TK]D-Fender | Qwell: INCLUDE FAIL |
| 19:46.46 | [TK]D-Fender | :p |
| 19:47.12 | jeff_phillips | Qwell: that's the way they wrote the file on http://www.asterisktimeclock.org it seemed odd to me |
| 19:47.19 | Qwell | well it's wrong |
| 19:47.31 | jeff_phillips | that would explain why it doesn't work then |
| 19:47.41 | cryptanthus | [TK]D-Fender: To get the output I used netstat > netstat.txt then I copied and pasted into pastebin.com, CLI sip set debug on ---> SIP debugging enabled |
| 19:47.47 | [TK]D-Fender | jeffAnd does their site take FreePBX garbage into account? |
| 19:48.02 | [TK]D-Fender | cryptanthus: copy paste it from CLI |
| 19:48.03 | jeff_phillips | no, it wasn't written for freePBX |
| 19:48.34 | [TK]D-Fender | jeff_phillips: Well then you should try to know the first thing about clashing extens in included contexts <- |
| 19:48.52 | [TK]D-Fender | jeff_phillips: [from-internal] has it's own "s" |
| 19:49.08 | jeff_phillips | why would they use s in this file? |
| 19:49.26 | Qwell | because they don't understand dialplan either |
| 19:49.27 | [TK]D-Fender | jeff_phillips: So put all of that "s" stuff in ANOTHER context and do a complete Goto() to jump there |
| 19:49.41 | *** part/#asterisk pointer (n=pointer@aj.catt.com) |
| 19:50.00 | [TK]D-Fender | jeff_phillips: Because its legit when they do it because they don't assume any overlap. |
| 19:50.48 | [TK]D-Fender | lesouvage: Dial -> M() + AMD() |
| 19:50.55 | cryptanthus | [TK]D-Fender: Do you mean from the bash terminal or some other command from asterisk CLI? |
| 19:51.22 | [TK]D-Fender | cryptanthus: I mean show me the COMPLETE netstat call from OS CLI. And show me the failed reg attemp SIP DEBUG from * CLI |
| 19:52.27 | jeff_phillips | They've got this all broken up into [timeclock-app], [timeclock-app-eng], etc... So I just need to create an extension in [from-internal] and tell it to goto [timeclock-app]? |
| 19:52.51 | *** part/#asterisk wwalker (n=wwalker@72.249.1.66) |
| 19:52.59 | cryptanthus | [TK]D-Fender: http://pastebin.com/d242cfc44 .... I'm working on getting the failed reg attempt |
| 19:53.13 | [TK]D-Fender | jeff_phillips: Your original include concept is find for the NUMBERED exten. its the "s" OVERLAP that FUBAR's you. Move THAT into another context |
| 19:53.46 | [TK]D-Fender | cryptanthus: "netstat -an" |
| 19:53.51 | jeff_phillips | couldn't I just change all the S's to something else? |
| 19:53.59 | Qwell | [TK]D-Fender: it is in another context. the problem is his goto doesn't switch contexts |
| 19:54.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:54.29 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 19:54.30 | shimi | hey tzaf |
| 19:54.34 | lesouvage | [TK]D-Fender: thanks, do you think the default values of the parameters will do? |
| 19:55.00 | [TK]D-Fender | Qwell: I supposed he could target that same context and avoid the overlap, but that still leaves priorities merged into [from-internal] that shouldn't be. HORRIFIC |
| 19:55.28 | Qwell | Goto(timethingie,s,1) |
| 19:55.31 | Qwell | all that's needed.. |
| 19:55.35 | [TK]D-Fender | lesouvage: Never personally used it, just chatted aout it a few times with people who have. |
| 19:55.45 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 19:56.17 | Qwell | it *should* be a Dial() on a Local channel, but...who knows how FreePBX would handle that |
| 19:56.17 | ariel_ | would it not be easyer to just add exten => _99990,Goto(Context,s,1) |
| 19:56.41 | Qwell | ariel_: that's what I said :p |
| 19:57.15 | lesouvage | [TK]D-Fender: I will do some serieus testing in the coming days |
| 19:57.29 | cryptanthus | [TK]D-fender: http://pastebin.com/d74a916c9 --- I see at the top there's a udp connection at 5060. |
| 19:57.34 | ariel_ | yes you did, but I feel they did not catch that and it would have taken care if it |
| 19:58.01 | jeff_phillips | I'm trying to understand a little better .. at the moment I have #include exten.timeclockapp.conf in extensions_custom.conf just under the [from-internal-custom]. So I need to move the include line to be outside of from-internal-custom so it is still going to load the file but not make it part of that context? |
| 19:58.05 | cryptanthus | [TK]D-Fender: There was no output on the asterisk CLI when I restarted the phone. |
| 19:58.16 | [TK]D-Fender | cryptanthus: Getting warmer. UDP is **STATELESS**. there is no such thing as a "connection". But at least it means something is LISTENING |
| 19:58.29 | [TK]D-Fender | cryptanthus: Time to check your firewall |
| 19:58.46 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 19:59.09 | ariel_ | jeff_phillips: remove the include and just add to the from-internal-custom the exten => _99990,goto(context,s,1) |
| 19:59.56 | cryptanthus | [TK]D-Fender: The firewall was already temporarily shut off. |
| 20:00.18 | jeff_phillips | ariel: how would asterisk know to load the contents of exten.timeclockapp.conf at all then, if I remove the #include entirely? |
| 20:00.45 | jeff_phillips | _9990 is defined inside that file |
| 20:00.46 | ariel_ | you sending the call there |
| 20:01.07 | Qwell | jeff_phillips: he said nothing about the #include |
| 20:01.15 | [TK]D-Fender | cryptanthus: Any other SIP software running on that server? |
| 20:01.50 | jeff_phillips | Qwell: "<ariel_> jeff_phillips: remove the include and just add ..." |
| 20:02.50 | cryptanthus | [TK]D-Fender: Not that I know of. It's a fresh install of Fedora 11 with source compiled versions of dahdi, dahdi-tools, asterisk and wanpipe (Sangoma driver). |
| 20:03.17 | ariel_ | jeff_phillips: if your sending the call via the dial plan to another context you don't have to include that context. |
| 20:03.53 | jeff_phillips | ariel: but somewhere I need to give it the file name that has all this in it |
| 20:04.08 | jeff_phillips | if I take the include line out, it won't know to load the file at all |
| 20:04.09 | [TK]D-Fender | cryptanthus: Prove that the FW is empty, and describe the networking between the 2 |
| 20:04.27 | jeff_phillips | i can't really goto something inside of a file that wasn't loaded |
| 20:05.28 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 20:05.48 | grandpapadot | Hey guys, on Polycom phones, is a 603 "Decline" usually an indicator that Do-Not-Disturb is active? |
| 20:07.13 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 20:09.30 | grandpapadot | Or rather, does anyone know what would cause a Polycom phone that has otherwise been working properly send back a 603 "Decline" message? |
| 20:10.49 | DaveCanoe | I've got a challenging problem. I'm getting a complete failure of IAX under lightly loaded conditions (5 to 20 calls from two sources). |
| 20:11.01 | cryptanthus | [TK]D-Fender: I don't know how to "prove" it but I disabled the firewall with the GUI config tool as well as I shutdown iptables and ip6tables. The asterisk server is on a static IP of 192.168.0.14 and the phone has been dynamically assigned an IP of 192.168.0.201 from the DHCP server. |
| 20:11.46 | DaveCanoe | Right when the problem occurs, I get "chan_iax2.c: Auto-congesting call due to slow response" followed quickly by "chan_iax2.c: Peer 'dot' is now UNREACHABLE! Time: 19" |
| 20:12.21 | DaveCanoe | ... but this is demonstrably not the problem as one of the hosts is on the same lan and the other is on an empty 10meg metro fiber (less than 2ms away). |
| 20:13.27 | jeff_phillips | Okay I put exten => 182,goto(timeclock-app,s,1) in my extensions_custom.conf |
| 20:13.49 | jeff_phillips | (I would rather it be 182 than the default 9990.) Now I get a message that my call can't be completed as dialed when i dial 182 |
| 20:14.10 | DaveCanoe | I've searched the archives extensivly to find more evidence of this problem. Once this bug happens (1.4.25, BTW), no IAX peers are reachable (even though the peers are up and responsive) |
| 20:14.52 | DaveCanoe | Not only that, but the host with the problem stops listening on it's IAX port. |
| 20:15.10 | grandpapadot | Polycom experts - does anyone know what would cause a Polycom phone that has otherwise been working properly send back a 603 "Decline" message? |
| 20:16.37 | [TK]D-Fender | cryptanthus: Doesn't work that way. "iptables --list" |
| 20:17.07 | [TK]D-Fender | grandpapadot: PASTEBIN <- |
| 20:17.41 | ariel_ | 603 is Decline mainly due to 2 things on a polycom, 1 DND pressed or they pressed ignor on the call when it was ringing. |
| 20:17.48 | cryptanthus | [TK]D-Fender: It shows empty. :) |
| 20:18.01 | [TK]D-Fender | cryptanthus: not that I see. |
| 20:18.15 | cryptanthus | [TK]D-Fender: ? |
| 20:18.22 | [TK]D-Fender | cryptanthus: PASTEBIN |
| 20:19.12 | *** join/#asterisk Elwell (i=elwell@freenode/staff/elwell) |
| 20:19.42 | cryptanthus | [TK]D-Fender: http://pastebin.com/d78d3ce09 |
| 20:20.19 | grandpapadot | TK: http://pastebin.com/m21d110a9 |
| 20:20.28 | [TK]D-Fender | cryptanthus: OK, well no packets arriving at * with SIP DEBUG enabled means your phone isn't even reaching the box |
| 20:20.41 | grandpapadot | TK: It's not anything sip.conf releated, it's something with the phone. Just lost on it. |
| 20:21.16 | [TK]D-Fender | grandpapadot: Do I really have to say it? |
| 20:21.58 | grandpapadot | TK, lol, what? Am I missing something obvious? |
| 20:22.09 | [TK]D-Fender | grandpapadot: F-ING SIP DEBUG |
| 20:22.36 | cryptanthus | [TK]Defender: I have the extensions.conf file as shown on page 73 of the book which shows an internal extension of 500. When I dial 500 on the Aastra phone... output does show on the asterisk CLI |
| 20:22.48 | grandpapadot | TK: One sec. |
| 20:23.51 | *** part/#asterisk lipek (i=lipek@lipek.pl) |
| 20:24.35 | citywok | any idea how to trigger resetcdr when doing a redirect from AMI? |
| 20:25.07 | DaveCanoe | Now... in reading chan_iax2.c ... it seems that this is a "can't get there from here" type of error --- ie possible stack corruption or something else evil. |
| 20:26.04 | jeff_phillips | well i give up with this stupid thing for now |
| 20:26.46 | *** join/#asterisk niros (n=niros@bzq-79-178-48-226.red.bezeqint.net) |
| 20:27.01 | jeff_phillips | thanks qwell & ariel, i think I have a better idea of what is wrong now |
| 20:27.16 | jeff_phillips | i'll come back to it later. ttyl |
| 20:27.24 | *** part/#asterisk jeff_phillips (n=jeff_phi@209-142-149-133.stat.centurytel.net) |
| 20:27.36 | *** join/#asterisk hammerzone45 (n=hammer15@c-71-229-108-12.hsd1.fl.comcast.net) |
| 20:28.18 | hammerzone45 | problem with queues kicking out people automatically .. anyone wants to help? |
| 20:29.20 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 20:29.39 | *** join/#asterisk p3nguin (i=BuhPjX1J@i.got.my.network.from.a2infotech.com) |
| 20:29.49 | [TK]D-Fender | checkout time, BBIAB |
| 20:31.01 | ariel_ | hammerzone45: queues only do what there told to do, more info on what is happening and what you can give us from your cli when it happens via pastebin would help. |
| 20:33.33 | *** join/#asterisk scunizi (n=scunizi@adsl-99-40-45-60.dsl.sndg02.sbcglobal.net) |
| 20:35.12 | hammerzone45 | ariel: i have set up settings in queues.conf and agents.conf to prevent that |
| 20:36.02 | hammerzone45 | ariel: like autologoffunavail = no |
| 20:36.46 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:38.08 | hammerzone45 | ariel: paste bin --> http://pastebin.com/d42877178 |
| 20:38.13 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:38.58 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 20:42.20 | hammerzone45 | anyone that can take a look at this code and help out? http://pastebin.com/d42877178 |
| 20:46.56 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 20:49.53 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 20:50.02 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 20:50.24 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 20:50.25 | *** join/#asterisk [TK]D-Fender (n=zsirc@161.216.151.9) |
| 20:55.15 | *** part/#asterisk b3nw (n=ben@unaffiliated/b3nw) |
| 20:55.45 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:59.03 | lesouvage | [TK]D-Fender: I've just done some testing with AMD() and it seems to work fine with default settings. |
| 20:59.15 | *** join/#asterisk SoCal (n=ster@pool-70-109-24-51.atl.dsl-w.verizon.net) |
| 20:59.41 | [TK]D-Fender | lesouvage: excellent |
| 21:00.15 | SoCal | do I need any particular card to run pbx via vmware |
| 21:00.39 | hammerzone45 | D-Fender: do you mind to take a look at this code to see what might be wrong? http://pastebin.com/d364a07bb |
| 21:00.53 | *** join/#asterisk jtodd (i=jgf2mhb8@ns.fox-den.com) |
| 21:00.53 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:01.05 | DaveCanoe | is there any chance that ztdummy is the root of the cause for IAX completely faulting out? |
| 21:01.09 | [TK]D-Fender | SoCal:no, only if you need to support actual physical lines |
| 21:01.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:01.34 | [TK]D-Fender | hammerzone45: ask again in 20min |
| 21:01.51 | hammerzone45 | d-fender: will do. tnx |
| 21:08.32 | joako | SoCal: If you run Asterisk in VMWare you CAN'T use any hardware interface cards |
| 21:08.42 | SoCal | ah |
| 21:09.20 | [TK]D-Fender | ~book |
| 21:09.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:09.29 | [TK]D-Fender | ~jerjerguide |
| 21:09.30 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 21:09.34 | [TK]D-Fender | ^^^^^ |
| 21:10.41 | joako | Jeremy's a jerk, FWIW |
| 21:12.32 | [TK]D-Fender | joako:to you perhaps |
| 21:14.36 | [TK]D-Fender | joako: I have my off-days as well. opinion may vary :) |
| 21:16.47 | *** join/#asterisk ebroder (n=broder@DR-WILY.MIT.EDU) |
| 21:16.49 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 21:21.18 | *** join/#asterisk af_ (n=getsmart@88-149-240-180.dynamic.ngi.it) |
| 21:22.06 | *** join/#asterisk garymc (n=gar@host86-162-158-214.range86-162.btcentralplus.com) |
| 21:23.34 | svm_invictvs | Is there a way to get Asterisk to forward E/911 data? |
| 21:23.44 | svm_invictvs | I have it set up for hosted PBX at multiple locations. |
| 21:24.50 | *** join/#asterisk ddickenson (n=ddickens@67-198-0-5.static.grandenetworks.net) |
| 21:25.18 | ddickenson | [TK]D-Fender: got a sec? |
| 21:26.18 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:26.37 | [TK]D-Fender | 1 min |
| 21:29.39 | ebroder | I'm having a hard time getting call forwarding (with Dial()) to work |
| 21:29.58 | ebroder | When the forwarded-to number picks up, all of the sound goes dead |
| 21:30.01 | Elwell | anyone point me to a good howto for spa3102+*? |
| 21:30.05 | ebroder | But I can play recordings and stuff from Asterisk before that |
| 21:30.19 | *** part/#asterisk clemahieu (n=none@63.169.105.141) |
| 21:31.54 | [TK]D-Fender | brb |
| 21:32.35 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:32.54 | svm_invictvs | so running asterisk on a slice isn't so bad. |
| 21:33.55 | [TK]D-Fender | back |
| 21:34.16 | [TK]D-Fender | ddickenson: Go for it |
| 21:34.24 | hammerzone45 | d-fender: have time to take a loo at his now? http://pastebin.com/d364a07bb |
| 21:34.52 | ddickenson | http://pastebin.com/d78983296 |
| 21:35.51 | [TK]D-Fender | hammerzone45: Ok, news to me... |
| 21:35.52 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-29.twcny.res.rr.com) |
| 21:36.41 | ddickenson | The problem I'm having is the dial part. I think the rest of the macro is working fine but when you dial it is telling me http://pastebin.com/d379a02d1 |
| 21:36.55 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:38.28 | [TK]D-Fender | ddickenson: * probably has no idea how to reach them. |
| 21:38.34 | [TK]D-Fender | ddickenson: Go look at your peer |
| 21:38.49 | ddickenson | sip show peers? |
| 21:38.55 | *** join/#asterisk ingenius (n=alektro@190.247.156.247) |
| 21:39.57 | ddickenson | I only have 4 phones active right now because it's a new install but here's the outputhttp://pastebin.com/d2655116a |
| 21:41.57 | ddickenson | http://pastebin.com/d642433db |
| 21:42.36 | [TK]D-Fender | ddickenson: Addr->IP : (Unspecified) Port 5060 <-- not registered, and * has nowhere to send the call |
| 21:43.32 | lesouvage | I'm not sure AMD is going to work. In Holland we have the habbit of picking up the phone and say something like "goodmorning with John, what can I do for you". This sets the status in AMD to answering machine. |
| 21:43.33 | ddickenson | d-fender: that's weird because it has to know about the phone or it wouldn't be getting the dial command from it... |
| 21:44.01 | ddickenson | and it's pulling IP |
| 21:44.07 | ddickenson | 's from the asterisk server |
| 21:44.34 | *** join/#asterisk manxpower (n=eric@69.73.94.162) |
| 21:45.45 | ddickenson | d-fender: http://pastebin.com/d474af471 I screwed up and printed one that wasn't connected yet... |
| 21:46.58 | *** join/#asterisk hardwire (n=hardwire@216-67-99-228.static.acsalaska.net) |
| 21:48.16 | *** join/#asterisk pulpster (n=pulpster@p5.eregie.pub.ro) |
| 21:48.35 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 21:49.50 | *** join/#asterisk lizone (n=zenst@user-0ccejib.cable.mindspring.com) |
| 21:49.59 | [TK]D-Fender | ddickenson: New call attempt with SIP DEBUG enabled and a new peer dump |
| 21:53.10 | *** join/#asterisk ddickenson_ (n=android@m3d0536d0.tmodns.net) |
| 21:53.44 | ddickenson_ | d-fender: closed out accidentally. thx for the help. back to debugging |
| 22:19.59 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
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| 22:36.45 | *** join/#asterisk italorossi (n=italoros@189.90.5.34) |
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| 22:43.31 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 22:48.23 | italorossi | Does anyone know the possible reason to get a ALARM_SYNC with R2 ? |
| 22:49.43 | *** join/#asterisk Alfio (n=Amunoz@adsl-54-20.tricom.net) |
| 22:50.06 | moy | italorossi: alarms are not R2 specific, and I suspect you are using Digivoice, aren`t you? |
| 22:50.31 | italorossi | Hello moy, exactly! I'm using Digivoice |
| 22:51.14 | italorossi | I'm trying to connect an asterisk 1.4.26 with Philips D120 |
| 22:51.51 | moy | you should try contacting digivoice support, never had some of those boards |
| 22:53.22 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:55.56 | *** part/#asterisk italorossi (n=italoros@189.90.5.34) |
| 23:03.55 | *** join/#asterisk ddickenson_ (n=ddickens@67-198-0-5.static.grandenetworks.net) |
| 23:04.38 | ddickenson_ | anyone have experience with call queues? More specifically how to make the phones ring rather than just beep and automatically accept call, and no MOH for agents |
| 23:06.19 | [TK]D-Fender | ddickensondon't use AgentLogin. thats the onlyt hing that "beeps" and auto-answers (you're actually already SITTING in the call) |
| 23:07.20 | ddickenson_ | what should I use instead if I want a group of 5 phones and people to be able to log in at any phone they sit down at? |
| 23:08.21 | ddickenson_ | and track who's getting what ammt of calls |
| 23:13.31 | ddickenson_ | no ideas? |
| 23:14.28 | *** part/#asterisk ebroder (n=broder@DR-WILY.MIT.EDU) |
| 23:14.34 | *** join/#asterisk snowboarder04 (n=un@94-193-80-153.zone7.bethere.co.uk) |
| 23:15.06 | [TK]D-Fender | ddickenson : "core show application addqueumember" |
| 23:15.12 | ddickenson_ | k |
| 23:16.11 | *** join/#asterisk f0ner00t (i=f0ner00t@99-21-148-71.lightspeed.frokca.sbcglobal.net) |
| 23:16.11 | ddickenson_ | thanks |
| 23:16.12 | f0ner00t | Hello |
| 23:16.25 | ddickenson_ | I think that's exactly what I need |
| 23:16.49 | f0ner00t | Question if I am using MD5 it should be authtype= md5 than secret = md5 secretcode |
| 23:25.31 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 23:26.28 | buttons840 | my script is nearly complete :) |
| 23:28.15 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
| 23:28.24 | *** join/#asterisk Faiz (n=otakucon@c-98-221-51-177.hsd1.nj.comcast.net) |
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| 23:31.17 | *** part/#asterisk sjobeck (n=Adium@69-30-99-139.dv1sn.easystreet.com) |
| 23:34.07 | *** join/#asterisk nixer (n=nixer@78.154.216.86) |
| 23:36.01 | nixer | I installed PBX in a Flash v 1.4 with Asterisk 1.6 and I'm trying to dial out on my PSTN line but I keep getting "All circuits are busy now" -- I tried changing from-pstn to from-zaptel but it didn't work. I've been reading forum posts for the past 2 hours and got no where. Where should I look? |
| 23:36.38 | Alfio | nixer GUI are not supported here, you can go to freepbx channel |
| 23:36.47 | Alfio | ~freepbx |
| 23:36.48 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:37.13 | nixer | I'm not trying to work the GUI. I generated the config files though the dahdi_genconf tool |
| 23:37.25 | nixer | * work with the gui |
| 23:37.57 | nixer | I'll ask there anyway, though I think this is more config related rather than GUI related. |
| 23:40.54 | f0ner00t | Hmmp weird your getting an all circuits are busy. |
| 23:41.02 | f0ner00t | What kind of circuit is it? |
| 23:41.08 | nixer | Analog. |
| 23:41.42 | f0ner00t | Hmmp Analog is usually straigh foward |
| 23:41.52 | f0ner00t | OFHK DT |
| 23:41.53 | f0ner00t | lol |
| 23:42.04 | f0ner00t | Odd your getting an all circuits are busy. |
| 23:42.10 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 23:42.15 | f0ner00t | It muzst not be sending on eof the signals properly. |
| 23:42.27 | nixer | No clue :/ |
| 23:43.23 | f0ner00t | Umm me neither sorry. |
| 23:43.34 | nixer | Thank you anyway :) |
| 23:44.36 | svm_invictvs | So I have 4 offices connected together with a central hosted PBX. How do I give each location it's own Emergency location? |
| 23:57.20 | Faiz | I can upgrade Asterisk 1.6.1.1 to 1.6.1.2 by running the latest patch, correct? |
| 23:58.01 | nixer | You need to apply the patch on the source code then recompile. |
| 23:58.55 | Faiz | is there a document I could follow that would assist me? it's my first time |
| 23:59.09 | nixer | No clue. Never did that. |
| 23:59.23 | nixer | What is your asterisk setup? |
| 23:59.34 | Faiz | i have the asterisk-1.6.1.2-patch.gz, not sure how to "extract" it |
| 23:59.34 | nixer | Did you compile it yourself? or are you using a customized distro? |
| 23:59.52 | Faiz | i'm running asterisk 1.6.1.1 currently, compiled myself |
| 23:59.59 | Faiz | on centOS v 5.3 |