| 00:01.14 | caloi | not sure about zillions of dollars; we are looking to supplement our SMB hosted product with a small call center application. The app will service 20 sessions X 20 tenants |
| 00:03.59 | tessier__ | 400 simultaneous calls? Asterisk can handle that on a good sized box. |
| 00:04.15 | tessier__ | I've run a few DS-3's off of one box. |
| 00:04.27 | tessier__ | I think I hit 8 DS-3's when I started having problems. |
| 00:04.51 | tessier__ | For that many calls definitely offload the PSTN stuff to some other gear though. |
| 00:05.31 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:05.35 | caloi | i saw an added benefit of SER/load balancing being the ability to have multiple 'matched' asterisk boxes behind it - if one asterisk box takes a dive, the other takes over |
| 00:06.02 | caloi | this scenario is all SIP; all the TDM -> SIP is done with our gateways |
| 00:06.07 | tessier__ | cool |
| 00:06.14 | tessier__ | And yes, that is an advantage to SER |
| 00:06.34 | tessier__ | That is pretty much the key to asterisk load balancing right there. Do it in SER. |
| 00:06.48 | tessier__ | Just be careful of any sort of state being saved on the individual asterisk boxes. |
| 00:07.03 | caloi | that's where i'm at now...... trying to figure out how :) |
| 00:08.10 | caloi | question though, since this app is going to be a call center app, and dependant on Asterisk queue app - is there a way to make that redundant? It seems if i lose an asterisk application server that the queue lives on, there's no way to recover from that.... right? |
| 00:08.21 | tessier__ | caloi: Unfortunately, yes. |
| 00:08.52 | caloi | so... in my scenario, i don't know if there is a benefit to SER before Asterisk, ya know what i mean? |
| 00:08.55 | tessier__ | caloi: You might want to set up a dedicated machine to just run the queue. And have another hot spare or another machine configured with the queue which can be made into the queue machine quickly |
| 00:09.02 | tessier__ | Right. |
| 00:09.19 | tessier__ | Really you only need SER if you are going to be handling so much volume that the media stream will overload the box. |
| 00:09.29 | tessier__ | Or if you want certain other kinds of redundancy. |
| 00:09.32 | tessier__ | Like call routing etc. |
| 00:09.34 | tessier__ | SER is good at that. |
| 00:10.05 | tessier__ | Just got phone to phone calling to work....sweet. If things keep going this easy it will be the easiest asterisk setup I have ever done. |
| 00:10.34 | tessier__ | Just need to figure out this PRI/zaptel/sangoma driver issue. |
| 00:10.39 | caloi | if i'm using asterisk real time for my queue and sip configs; and i offload the DB on a third box; if I lose my master asterisk will the second box maintain that queue state? like agent availability? |
| 00:10.58 | caloi | tessier: sorry i can't help ya with your pri foo |
| 00:11.06 | tessier__ | I don't know about that...I have never messsed with queues before. But the call center system I am setting up now will use queues. |
| 00:25.12 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 00:29.22 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 00:29.35 | *** join/#asterisk aigroine (n=ceb@paris.coldev.org) |
| 00:29.37 | aigroine | hoi |
| 00:30.56 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 00:34.30 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 00:34.42 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 00:45.25 | *** join/#asterisk finejava (n=12345@60.50.251.234) |
| 00:45.31 | finejava | hi guys |
| 00:45.46 | finejava | i've dot 1 question on dial application |
| 00:46.10 | finejava | in IAX i can dial(IAX2/test:test@host.com/12345) |
| 00:46.17 | finejava | how can i do that in SIP |
| 00:46.35 | finejava | i can't seems to get it working with the same format |
| 00:48.15 | *** join/#asterisk pbx1 (i=pbx1@netblock-66-245-194-170.dslextreme.com) |
| 00:48.20 | finejava | anyone out there can help me with the issue above? |
| 00:48.52 | aigroine | i think that it should work by replacing IAX2 by SIP |
| 00:49.00 | aigroine | but i may be wrong |
| 00:49.31 | finejava | i tried it |
| 00:49.39 | finejava | but it says no route to host |
| 00:49.57 | finejava | which means it's trying to find host.com in the sip.conf |
| 00:50.10 | aigroine | this problem seems to be related to network .. |
| 00:50.15 | Aurs | traceroute host.com |
| 00:50.30 | finejava | i can ping host.com |
| 00:50.38 | finejava | i'm using an internal DNS server |
| 00:50.48 | finejava | it works with IAX |
| 00:50.51 | Aurs | any help in "show application Dial" ? |
| 00:50.57 | finejava | nope |
| 00:51.14 | finejava | can't seems to find the dialing format guide |
| 00:52.08 | finejava | No such host: test.genme.com/12345 |
| 00:52.22 | finejava | this is the error that i'm getting |
| 00:52.29 | Aurs | http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+channels |
| 00:52.31 | finejava | from SIP |
| 00:54.30 | Aurs | what are you trying to call here? |
| 00:54.44 | Aurs | a sippeer on another box? |
| 00:56.20 | finejava | i'm trying to call asterisk to asterisk |
| 00:56.57 | finejava | Dial(SIP/8500@sip.com:9876) |
| 00:57.05 | finejava | where do i put the username and secret? |
| 00:57.43 | finejava | i wan the hostname to be in the dial application so that it can do a srv lookup |
| 00:57.52 | Aurs | Dial(SIP/${EXTEN}@host.com) |
| 00:59.28 | finejava | if i do that which means the B side must not set the password |
| 00:59.40 | finejava | or else it will request for authentication |
| 01:00.21 | finejava | the issue here is my B side is a dynamic IP |
| 01:00.26 | aigroine | finejava: to my mind .. you don't have to give a username/password to call somebody |
| 01:00.31 | aigroine | this would be insane |
| 01:00.50 | aigroine | you have a username/password to register on a server |
| 01:00.55 | Aurs | the person calling is probably a sip friend |
| 01:01.03 | finejava | this is my issue |
| 01:01.04 | Aurs | and he needs to have a password to be registered |
| 01:01.25 | finejava | i've * PABX on A side |
| 01:01.37 | finejava | and SIP GW in B side |
| 01:01.49 | finejava | B side has multiple user |
| 01:02.04 | aigroine | hmm |
| 01:02.13 | finejava | and A is running on dynamic DNS |
| 01:02.23 | finejava | which changes every 2 hours or so |
| 01:02.23 | aigroine | i think the solutions is to search in sip.conf |
| 01:02.35 | aigroine | something like defining a new sip peer |
| 01:02.40 | aigroine | let me check my configuration |
| 01:02.41 | finejava | unfortunately...if i set it as a peer |
| 01:02.51 | finejava | it will not lookup when the dns changes |
| 01:03.12 | aigroine | hmm |
| 01:03.33 | aigroine | as B is your gw with a static address, it doesn't matter |
| 01:03.33 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 01:03.49 | aigroine | you define a new section in sip.conf like [gw] |
| 01:04.03 | aigroine | with host=host.com |
| 01:04.07 | aigroine | username=user |
| 01:04.19 | aigroine | secret=password |
| 01:04.22 | finejava | how bout when B trying to call A |
| 01:04.26 | aigroine | type=peer |
| 01:04.38 | finejava | A to B is fine cos B is on static |
| 01:04.39 | aigroine | the same |
| 01:04.45 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 01:04.59 | aigroine | i don't know if resolution is at loading or at usage |
| 01:05.06 | finejava | unfortunately in B host=test.host.com |
| 01:05.16 | finejava | it cache the IP in asterisk |
| 01:05.20 | aigroine | hmm |
| 01:05.32 | Aurs | your ip is changed every 2 hours? |
| 01:05.37 | finejava | yeap |
| 01:05.54 | finejava | unstable connection |
| 01:06.09 | Aurs | sounds perfect for voip |
| 01:06.15 | aigroine | finejava: if A is registered on B, i think that B may be able to forward calls to the [client] in its sip.conf |
| 01:06.32 | finejava | LoL |
| 01:06.35 | Aurs | host=dynamic |
| 01:06.36 | aigroine | in A extensions.conf , you may call extension at gw |
| 01:06.38 | Aurs | in sip conf |
| 01:06.58 | aigroine | Aurs ;) |
| 01:07.08 | Aurs | when you send a registration, * will get the correct ip |
| 01:07.08 | finejava | tried it...didn't work |
| 01:07.17 | finejava | works 1 way... |
| 01:07.21 | finejava | but not the other |
| 01:07.43 | Aurs | your sip-gw is on a static ip, right? |
| 01:07.49 | *** join/#asterisk Crashsys (n=kumba@office.crashsys.com) |
| 01:07.50 | Aurs | and your asterisk-client is on a dynamic |
| 01:07.52 | aigroine | so the problem is perhaps in B extensions.conf ... |
| 01:08.19 | finejava | yes...it's in B |
| 01:08.23 | finejava | problem is in B |
| 01:08.29 | finejava | B can't reach A |
| 01:08.30 | aigroine | like Dial(SIP/5000@client) |
| 01:08.34 | finejava | A can reach B |
| 01:08.42 | Crashsys | Is a P3-500 w/ 256-megs of ram enough system for 2-pots and 6 extensions? (Just basic automatic-attendant with voicemail) |
| 01:08.49 | finejava | i can't use client which is in sip.conf |
| 01:08.54 | Aurs | register => 1234:password@mysipprovider.com |
| 01:08.54 | Crashsys | sip-extensions... |
| 01:09.00 | aigroine | where client is related to a [client] section with host=dynamic in sip.conf |
| 01:09.02 | finejava | it will not lookup |
| 01:09.05 | Aurs | do you have something like that in sip.conf on the * client? |
| 01:09.22 | aigroine | 19~/win 8 |
| 01:10.33 | finejava | yeap i have that |
| 01:10.41 | Aurs | now i think i understand |
| 01:11.03 | finejava | SIP/12345@test.host.com:5060 |
| 01:11.29 | Aurs | but the lookup of test.host.com fails |
| 01:11.35 | Aurs | that is your problem, right |
| 01:11.41 | finejava | nope |
| 01:11.50 | finejava | lookup works with IAX |
| 01:12.08 | finejava | ok lookup works if i do this |
| 01:12.10 | Aurs | does test.host.com have a sip peer named 12345 then? |
| 01:12.21 | Aurs | and is that peer registered? |
| 01:12.31 | finejava | Dial(IAX2/test:test@test.host.com/12345 |
| 01:12.40 | finejava | 12345 is an extension |
| 01:12.46 | *** join/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 01:13.05 | finejava | in my test 12345 will just playback a voice to simulate the scenario |
| 01:13.15 | finejava | Dial(IAX2/test:test@test.host.com/12345) |
| 01:13.18 | finejava | it works fine |
| 01:13.20 | Aurs | but test.host.com is not just a host in this case |
| 01:13.52 | finejava | but when i do this |
| 01:13.59 | finejava | Dial(SIP/test:test@test.host.com/12345) |
| 01:14.06 | finejava | it doesn't works at all |
| 01:14.28 | aigroine | finejava: but to my mind, this is not the right syntax to DIAL(SIP/...) |
| 01:14.52 | aigroine | ${EXTEN} may looks like user@context |
| 01:14.55 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 01:14.58 | Aurs | Dial(SIP/12345@test.host.com) |
| 01:15.00 | Crashsys | Does anyone know if a PRI is less prone to red alarms then a CAS T1? (My provider is telling me that I have a clean line all the way to the smart jack, yet i'm still getting seemingly randomn alarms where all 24-channels will go down then come back up) |
| 01:15.12 | finejava | tried that... |
| 01:15.29 | finejava | but where do i put the username and scret? |
| 01:15.38 | finejava | cos it needs authentication |
| 01:15.42 | _DAW | Hey everyone. Can someone tell me if during a call forward, does * send a diversion header in the invite if it does not recieve one in the 302 from the phone? |
| 01:15.56 | Aurs | why do you need to auth incoming calls again? |
| 01:16.34 | finejava | security reasons |
| 01:16.59 | finejava | B -> A(PABX) -> user |
| 01:17.09 | Aurs | so this call is going to be "forwarded" to PSTN or something? |
| 01:17.32 | *** join/#asterisk Fender22211 (n=fender21@cpe-70-125-138-128.satx.res.rr.com) |
| 01:17.33 | finejava | i'll pass the calls to * PABX which will then forward it to voip user |
| 01:17.36 | aigroine | finejava: you put in in a context section in sip.conf |
| 01:17.50 | finejava | meaning? |
| 01:18.36 | Fender22211 | anyone had any success implmenting the Voicemail ODBC Storage? |
| 01:19.01 | Aurs | finejava: then you need a [client] in A's sip.conf |
| 01:19.03 | aigroine | the context section in sip.conf is something like [gateway] username=toto secret=1234 host=test.com |
| 01:19.04 | Aurs | with type=friend |
| 01:19.15 | finejava | problem is |
| 01:19.23 | finejava | if i uses context |
| 01:19.32 | finejava | it will not lookup on the DNS |
| 01:19.34 | Aurs | and in B's sip.conf: register => client:pass@A |
| 01:19.54 | aigroine | finejava: as A is registering on B, only A has to do some lookup |
| 01:20.12 | finejava | A is on dnydns |
| 01:20.22 | finejava | cos it's on dynamic IP |
| 01:20.27 | aigroine | finejava: yes , we understood |
| 01:20.31 | finejava | if u check out sip.conf |
| 01:20.38 | finejava | srvlookup=yes |
| 01:20.58 | finejava | it will only take effect if FQDN is uses in the Dial application |
| 01:21.20 | finejava | i've manage to get it working with IAX |
| 01:21.23 | finejava | but not SIP |
| 01:21.41 | aigroine | hmm .. |
| 01:22.09 | aigroine | cannot help further .. have never worked with srvlookup and IAX |
| 01:22.18 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:22.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:22.41 | finejava | IAX i got it working |
| 01:22.46 | finejava | just that SIP |
| 01:22.49 | adelas | hey guys, can you tell me whats teh default ports to forward for sip to asterisk? |
| 01:22.53 | adelas | i know 5060 udp |
| 01:22.57 | adelas | whats the other ones? |
| 01:23.08 | Aurs | 10000-20000 for rtp traffic |
| 01:23.09 | finejava | same format doesn't work with SIP |
| 01:23.18 | aigroine | adelas: RTP port are defined in rtp.conf |
| 01:23.34 | adelas | well, my thing ins't free asterisk :\ |
| 01:23.50 | aigroine | adelas: read the doc |
| 01:23.51 | adelas | it uses sql webinterface, dosn't use the normal asterisk |
| 01:24.04 | adelas | this is like a properitery asterisk thing |
| 01:24.08 | aigroine | hm |
| 01:24.16 | Fender22211 | unless you can open all those RTP ports or set your box to DMZ, stick with IAX.. |
| 01:24.20 | adelas | it dosn't use the normal extension/sip.cfg ect files |
| 01:24.24 | aigroine | you could try to tcpdump but 10000-20000 may be a good start for a quick setup |
| 01:24.39 | adelas | well i did a 6000-max for udp port |
| 01:24.45 | adelas | i just wanted to narrow it down in a couple of days |
| 01:24.52 | adelas | the 10000-10100 didn't work |
| 01:24.53 | aigroine | isn't rtp in tcp ? |
| 01:25.04 | adelas | umm, also it was using udp |
| 01:25.06 | adelas | tcp didn't work |
| 01:25.32 | Fender22211 | SIP will use the complete range in the ports 10000-20000 so yo literally have to open them all for SIP. |
| 01:25.41 | adelas | bascially right now i got 5060 udp + 6000-max udp |
| 01:26.07 | adelas | heh 10k in open ports |
| 01:26.09 | adelas | okay |
| 01:26.18 | adelas | i'll give that a try tomrrow to see how it goes |
| 01:26.41 | adelas | better then everything i'm doing heh |
| 01:26.41 | Fender22211 | yeah, that's why IAX is nice.. everything goes through the one port ;-) |
| 01:26.47 | adelas | cisco phones are sip only :* |
| 01:26.48 | adelas | :( |
| 01:27.03 | Fender22211 | true.. |
| 01:27.07 | Fender22211 | Good luck Adela |
| 01:27.12 | adelas | okay thanks |
| 01:27.14 | Aurs | finejava: google around for "sip trunk" |
| 01:28.19 | Fender22211 | I'm trying to programatically (PHP) build something that will allow me to pull a voicemail per the CDR record. Does anyone have any ideas how to accomplish this? |
| 01:29.20 | aigroine | Fender22211: what do you mean "per the CDR record" ? |
| 01:30.23 | Fender22211 | well we are trying to build a custom app that will pull up the CDR record in a php web app and give the option to listen to the voicemail (if any). I'm struggling to find a programatically link between a call and a voicemail |
| 01:30.45 | Fender22211 | my first thought was to use the VOICEMAIL ODBC STORAGE method but that ended up seg faulting my test box. |
| 01:31.31 | Fender22211 | my second thought was to use SetCDRUserfield($VM_MESSAGE) which in theory should store the path in the CDR userfield but I have yet to get that to work. |
| 01:32.20 | aigroine | informations related to calls are stored in a .txt files having the same name that the message .wav |
| 01:32.54 | Fender22211 | right, but I haven't been able to see a link (i.e. UNIQUEID) between the CDR record and the voicemail? |
| 01:33.51 | Fender22211 | i know it's possible because the voicemail web app does this.. I'd rather not parse those .txt files if given the choice |
| 01:38.12 | finejava | thx guys...will do more rnd on it |
| 01:38.17 | finejava | appreciate it |
| 01:38.24 | aigroine | Fender22211: don't know |
| 01:39.00 | Fender22211 | me either :-) I'll keep searching, thanks aigroine |
| 01:39.37 | *** join/#asterisk xtr-II (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 01:41.56 | aigroine | np :) |
| 01:42.40 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-59.dslextreme.com) |
| 01:47.11 | *** part/#asterisk Grizzy (i=Generic@adsl-68-124-190-164.dsl.pltn13.pacbell.net) |
| 01:50.40 | *** part/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 01:57.49 | _DAW | Anyone here know if during a call forward, does * send a diversion header in the invite if it does not recieve one in the 302 from the phone? |
| 01:58.32 | file | ...what? |
| 01:59.04 | _DAW | what for me? |
| 01:59.34 | file | I'll just answer no |
| 02:00.39 | _DAW | fine |
| 02:01.27 | file | as in to your original question, Diversion header is never sent... and it's only parsed under one circumstance, which doesn't apply to call forwards - I think, I'd have to follow the code path to be sure |
| 02:05.16 | _DAW | I am speaking about diversion as specified in draft-levy-sip-diversion-04.txt . I read up in the * bug 0005484 but am having some issues. |
| 02:05.36 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:06.51 | hacked`` | guys |
| 02:06.54 | hacked`` | lets say i want to accept 4 incoming calls at the same time |
| 02:07.00 | hacked`` | whats a good provider to go with |
| 02:07.05 | hacked`` | know what i mean ? |
| 02:07.17 | Strom_C | voicepulse connect? |
| 02:07.17 | hacked`` | that isnt expensive, and is reliable |
| 02:07.25 | hacked`` | is that a question or a statement |
| 02:07.26 | file | _DAW: ah, well trunk doesn't support it... and can't say I have ever used that branch |
| 02:07.30 | Druken | from where? |
| 02:07.33 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:07.35 | Strom_C | hacked``, thats a half-suggestion |
| 02:07.42 | file | hacked``: you want it cheap and reliable? |
| 02:07.47 | _DAW | fiel: I guess that was my question. Thanks. |
| 02:07.51 | _DAW | er file |
| 02:07.54 | Corydon76-home | hacked``: your primary concern is your own ISP's bandwidth, not the voip provider |
| 02:08.12 | hacked`` | file, never said cheap, i said reasonable |
| 02:08.18 | Corydon76-home | Personally, I'd go with NuFone |
| 02:08.23 | hacked`` | Corydon, cable 10mbit |
| 02:08.35 | Fender22211 | exgn.net/vitelity.net has been great for me.. |
| 02:08.51 | Corydon76-home | hacked``: specifically, the problem with an ISP is going to be your upload limit |
| 02:09.46 | hacked`` | cory, what do i need for 4 incoming |
| 02:10.24 | Corydon76-home | It depends upon the codec and protocol you're using |
| 02:10.42 | Fender22211 | hacked: check out asteriskguru.com for their bandwidth calculator |
| 02:13.58 | Druken | anyone got the freeware g729 working? |
| 02:14.22 | aigroine | hmm .. |
| 02:14.44 | aigroine | I'm trying to get a music on hold depending on extension reached |
| 02:14.59 | aigroine | is the only way to setup a class by extension ? |
| 02:15.07 | aigroine | it seems so without patching asterisk |
| 02:15.45 | Druken | aigroine: wtf are you talking about? |
| 02:16.13 | _DAW | aigroine: you can set it by sip peer |
| 02:17.29 | aigroine | _DAW: don't think i can do this as my clients are created with autocreatepeers=yes |
| 02:18.38 | aigroine | Druken: what i want is asterisk playing different mp3 depending on phone extension reached when the person press the hold button |
| 02:19.09 | Druken | ok, so setmusiconhold before you ring the extension |
| 02:19.39 | aigroine | Druken: it is what i do ... but it means i must have a class by extension right ? |
| 02:19.53 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 02:20.16 | Druken | you'd need a diffrent context, yes |
| 02:20.47 | aigroine | hmm ok .. it sux ... having as many mpg123 as context running ... |
| 02:21.39 | Druken | why would you want that? |
| 02:22.10 | _DAW | dont use mpg123.. just use native |
| 02:22.50 | aigroine | _DAW: with format_mp3 ? |
| 02:23.02 | aigroine | I've tried to setup it .. but it seems it's not loaded |
| 02:23.19 | aigroine | so i put this in to todo queue for the moment |
| 02:23.51 | aigroine | Druken: we have several clients and we may like to have one different music on hold file per client |
| 02:25.48 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 02:27.02 | Fender22211 | anyone here using the VOICEMAIL ODBC STORAGE method? |
| 02:27.44 | Druken | someone needs to send me a working g729 codec module :) |
| 02:29.41 | Strom_C | Druken, just spend the $10 |
| 02:30.15 | Druken | i did, put the order in today, hours ago, but the fuckers are taking forever to process it |
| 02:30.24 | Druken | purchased 3 licenses |
| 02:30.26 | Qwell | it IS Saturday... |
| 02:30.29 | Strom_C | Druken, because today is Saturday |
| 02:30.32 | Strom_C | and the business is CLOSED |
| 02:30.37 | Druken | wtf is your point? |
| 02:30.42 | Druken | i work on saturdays |
| 02:30.49 | Druken | so should everyone else :) |
| 02:30.58 | Qwell | Druken: pay me, and I will :P |
| 02:31.55 | Druken | got nothing for ya to do at the moment..... sorry Qwell |
| 02:32.08 | Qwell | no, no, you misunderstood |
| 02:32.19 | Qwell | Pay me, and I'll work on Saturdays - but not for you. :P |
| 02:32.28 | file | lol |
| 02:32.36 | Druken | pfft |
| 02:32.38 | Druken | hehe |
| 02:32.41 | Qwell | ;) |
| 02:33.28 | Fender22211 | I'll pay someone to install this damn VOICEMAIL ODBC stuff.. this documentation is shotty at best |
| 02:33.32 | Druken | i'm just annoyed at the moment, cause my god damn carrier will only accept g729, so i can't make any calls till i get it |
| 02:33.55 | Strom_C | Druken, what kind of stupid carrier will only accept g729? |
| 02:34.21 | Druken | exactly that... hehe |
| 02:37.04 | coppice | Druken, what kind of stupid user signs up for a carrier that is incompatible? :-) |
| 02:37.16 | hmmhesays | bah I can't figure out wtf is going on with this |
| 02:37.18 | Qwell | No comment |
| 02:37.32 | hmmhesays | : channel.c:2706 ast_channel_make_compatible: No path to translate |
| 02:37.40 | file | 256 to 4? |
| 02:37.58 | Qwell | 4 to 12 |
| 02:38.04 | file | nooo not 12 |
| 02:38.07 | Qwell | uh huh |
| 02:38.22 | hmmhesays | file what is wrong with this box |
| 02:38.39 | file | hmmhesays: paste me the complete one and I can tell you what... |
| 02:38.52 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:39.32 | hmmhesays | http://pastebin.ca/150558 |
| 02:39.54 | file | one side negotiated at g729, the other at ulaw |
| 02:40.07 | hmmhesays | yet I don't understand why |
| 02:40.26 | hmmhesays | and how'd you get that out of that small log |
| 02:40.28 | file | sip debug the somalia_trunk |
| 02:40.34 | file | the numbers beside each name |
| 02:40.36 | Qwell | hmmhesays: format numbers.. 256 and 4 |
| 02:40.46 | file | 256 = g729, 4 = ulaw |
| 02:40.55 | Qwell | 12 == format_qwell |
| 02:41.01 | file | Qwell: ! |
| 02:41.05 | hmmhesays | so the codec negotiation is broken |
| 02:41.12 | Strom_C | 800km == format_file |
| 02:41.16 | Qwell | I'm like ulaw + alaw |
| 02:41.30 | Qwell | hmmhesays: What codecs is the device allowing? |
| 02:41.38 | file | do a sip debug |
| 02:41.39 | Qwell | the somalia trunk |
| 02:41.40 | file | on the somalia_trunk |
| 02:41.45 | file | and I can tell you what it is doing. |
| 02:41.51 | hmmhesays | the device is allowing g723 g729 and g711 |
| 02:42.07 | Qwell | and in sip.conf, you allow ulaw? |
| 02:42.22 | hmmhesays | ulaw g729 and ilbc in that order |
| 02:42.28 | aigroine | _DAW: is the native mp3 format for a* 1.2.x is supported by the format_mp3 pluggin from asterisk-addons ? |
| 02:42.51 | Qwell | aigroine: yes, format_mp3 will let you play mp3s from asterisk |
| 02:42.58 | aigroine | hmm ok |
| 02:43.04 | _DAW | like Qwell said.. |
| 02:43.10 | aigroine | will check if it can help |
| 02:45.56 | hmmhesays | ok something is farked up here |
| 02:46.37 | file | The Guten Tag Hop-Clop! |
| 02:48.50 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 02:51.29 | hmmhesays | so found description format G729 is there |
| 02:51.37 | hmmhesays | and i'm still having the problem |
| 02:52.26 | file | can't analyze it if I don't see it |
| 02:53.01 | hmmhesays | you want to see the sip debug after dial? |
| 02:53.17 | file | ...sure |
| 02:53.21 | *** part/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
| 02:54.34 | hmmhesays | http://pastebin.ca/150565 |
| 02:55.29 | file | it only has G729 in the outgoing SDP |
| 02:55.59 | hmmhesays | so it should be working |
| 02:56.14 | file | no, your other leg negotiated ulaw |
| 02:57.15 | hmmhesays | Peer audio RTP is at port 192.168.1.151:51004 |
| 02:57.15 | hmmhesays | Found description format PCMU |
| 02:57.15 | hmmhesays | Found description format G729 |
| 02:57.53 | file | like I said, one side negotiated at ULAW and the other only appears to have G729 |
| 02:58.00 | file | so thus why this is popping up |
| 02:58.10 | file | plus your trunk doesn't appear to be responding, but that's different |
| 02:58.34 | hmmhesays | yeah, i'm way too tired for this right now |
| 02:59.42 | hmmhesays | bah I had ulaw before g729 in general section of sip.conf |
| 03:18.09 | *** join/#asterisk AJaymn (n=boiwonde@156-77.dsl.scc.net) |
| 03:20.20 | *** join/#asterisk tiab (i=216389@bud.cc.swin.edu.au) |
| 03:25.05 | tiab | \? |
| 03:25.08 | tiab | ? |
| 03:26.48 | *** join/#asterisk Avalone (n=Avalone_@dial-448.vl-cen-as3.avtlg.ru) |
| 03:27.39 | tiab | hey all |
| 03:28.50 | tiab | anyone here able familiar with asterisk + odbc + postgres? |
| 03:29.40 | tiab | I;m having some trouble getting asterisk to store voicemail in the voicemessages table in a postgres db |
| 03:30.22 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 03:36.32 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.108) |
| 03:46.25 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 03:55.49 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 03:55.59 | Rahail | hi any one here used a2billing ? |
| 03:59.02 | Rahail | knock knock |
| 04:00.21 | *** join/#asterisk MGSsancho (n=sancho@adsl-67-126-128-134.dsl.irvnca.pacbell.net) |
| 04:03.52 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 04:09.20 | tiab | nope rahail |
| 04:10.12 | tiab | don't spose you've used asterisk with a postgres db connection via odbc? ;) |
| 04:10.52 | tiab | or anyone else watching for that matter.. |
| 04:13.12 | Rahail | thanx |
| 04:13.37 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 04:14.04 | novafirst | inside sip.conf can I use this callerid=${CALLERIDNUM} |
| 04:15.09 | novafirst | anyone? |
| 04:16.00 | Strom_C | format is: |
| 04:16.10 | Strom_C | callerid="Name Here"<5552368> |
| 04:20.52 | *** join/#asterisk atapi2 (n=virgill4@c-69-180-119-156.hsd1.fl.comcast.net) |
| 04:32.52 | novafirst | can "Name Here" be set automaticaly ? |
| 04:36.00 | Rahail | Any one here who can make a billing interface when my client they reach there Prepaid amount it desabel there account |
| 04:42.50 | *** join/#asterisk tessier (n=treed@75.5.99.178) |
| 04:43.15 | lowlevel | Rahail: yes. |
| 04:43.24 | Rahail | do you think you can do something like that |
| 04:43.36 | lowlevel | Rahail: I could definately. |
| 04:43.50 | lowlevel | Rahail: kind of like pre-paid calling cards ? |
| 04:43.53 | Rahail | yeah |
| 04:43.56 | lowlevel | sure. |
| 04:44.03 | Rahail | can we talk on private |
| 04:44.07 | lowlevel | why? |
| 04:44.08 | Rahail | if you preffer |
| 04:44.14 | lowlevel | heh |
| 04:44.20 | Rahail | i dont like pasting big lines here |
| 04:44.35 | lowlevel | well, I'm not going to do it .. you just asked if I could ;) |
| 04:45.37 | Rahail | OK |
| 04:45.53 | Rahail | however if you do it I will realy appricate it |
| 04:45.57 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.108) |
| 04:51.50 | *** join/#asterisk Tommmo (n=tps@203.62.181.52) |
| 04:51.51 | Tommmo | hi |
| 04:52.06 | Tommmo | in asterisk realtime extensions, how do i insert an 'include' statement into the database |
| 04:52.07 | Tommmo | ? |
| 05:03.20 | Tommmo | is there any way to have different call parking configurations for different contexts? |
| 05:06.16 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 05:07.39 | *** join/#asterisk BugKham (i=CKGLOB@61.47.100.233) |
| 05:08.31 | BugKham | anyone using PlayDTMF in 1.2.11? |
| 05:09.16 | BugKham | or knows if it's available in 1.2.11? |
| 05:10.32 | Rahail | Any one here who can make a billing interface when my client they reach there Prepaid amount it desabel there account |
| 05:16.27 | *** join/#asterisk BlepsoaF (n=pbaker@ool-457805b1.dyn.optonline.net) |
| 05:16.40 | BlepsoaF | hello all, does ztcfg have to be ran each time at boot? |
| 05:19.10 | Corydon76-home | Yes |
| 05:20.17 | Corydon76-home | Although, if you use it according to how the Makefile installs the modules, the ztcfg is done automatically when the module is loaded |
| 05:20.50 | *** join/#asterisk AJay-mn (n=boiwonde@156-77.dsl.scc.net) |
| 05:23.39 | BlepsoaF | hmm |
| 05:23.47 | Rahail | man |
| 05:23.58 | Rahail | any one used a2billing |
| 05:24.08 | BlepsoaF | Corydon-w: I autoload the module at boot time |
| 05:28.45 | BlepsoaF | Corydon-w: can you think of a reason why its not working then? I compiled it from source |
| 05:32.39 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 05:34.06 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 05:42.03 | *** join/#asterisk hatamen (n=hatamen@222.183.23.72) |
| 05:43.11 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 05:48.06 | Rahail | can some one help me to configure a2billing |
| 05:48.09 | Rahail | knock knock wake up |
| 05:48.15 | EyeCue | whos there?! |
| 05:48.19 | EyeCue | oh |
| 05:48.21 | Rahail | no one |
| 05:48.24 | EyeCue | *goes back to sunbaking on balcony* |
| 05:48.32 | Rahail | lol |
| 05:51.15 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 05:52.46 | BlepsoaF | figured it out, must be a bug in the Makefile |
| 05:53.02 | BlepsoaF | it doesnt generate the correct modules.d/zaptel for a wct4xxp |
| 06:01.20 | *** join/#asterisk afrosheen (n=test@c-24-0-138-247.hsd1.tx.comcast.net) |
| 06:01.36 | afrosheen | someone please tell me you know of a mirror for spandsp, their stupid website is down |
| 06:03.27 | EyeCue | http://www.freshports.org/comms/spandsp/ |
| 06:03.32 | EyeCue | check out master_sites section |
| 06:03.48 | EyeCue | should be able to pull the sources from anyone of those |
| 06:04.22 | afrosheen | will it matter than I'm not using freebsd but this is listed as a port? |
| 06:04.37 | EyeCue | shouldnt do |
| 06:04.41 | EyeCue | hang |
| 06:05.33 | EyeCue | ftp://ftp.uk.freebsd.org/pub/FreeBSD/ports/local-distfiles/pav/spandsp-0.0.2pre18.tar.gz |
| 06:06.29 | EyeCue | ftp://ftp.freebsd.org/pub/FreeBSD/ports/distfiles/spandsp-0.0.2pre26.tar.gz |
| 06:06.30 | afrosheen | cool..now I'm missing the app_fax and other .c files |
| 06:06.31 | EyeCue | dar joo go |
| 06:06.38 | EyeCue | later version on another mirror |
| 06:07.11 | afrosheen | app_rxfax and app_txfax |
| 06:07.18 | afrosheen | plus the patch |
| 06:07.28 | afrosheen | gah I hate it when people can't maintain a simple website or encourage mirroring |
| 06:07.46 | EyeCue | blows donut |
| 06:08.05 | afrosheen | seriously how hard is it? you can have a 486 under a pile of newspapers running a website |
| 06:08.12 | mishehu | ugh. speex takes a shit on opteron x86_64 builds :-/ |
| 06:08.32 | afrosheen | mishehu: I don't trust native 64bit stuff yet :/ |
| 06:12.39 | mishehu | afrosheen: it's really the only thing other than non-x86_64 asm code (i.e. zaptel MMX opt code) that craps out on me. |
| 06:12.59 | afrosheen | mishehu: I guess 64bit stuff has come a long way in the last year |
| 06:13.07 | mishehu | afrosheen: *nod* |
| 06:13.19 | afrosheen | I wouldn't touch it with a yardstick before |
| 06:13.53 | afrosheen | mishehu: do you see any real benefit from using it vs. 32bit? |
| 06:13.57 | mishehu | afrosheen: yeah, I run slamd64 for over a year, been doing some dev work on it, even playing kwak 4. |
| 06:14.42 | afrosheen | lol kwak 4 |
| 06:15.00 | mishehu | afrosheen: yeah, with transcoding media files especially, and with working with big numbers (> 32bit integers, for example) |
| 06:15.20 | afrosheen | so a 64bit linux is faster at ripping dvds maybe? |
| 06:15.39 | mishehu | afrosheen: I don't know about *ripping* them, but at transcoding them yes. |
| 06:15.49 | afrosheen | well transcoding is part of the process |
| 06:15.53 | afrosheen | that's cool then |
| 06:16.04 | mishehu | nah, ripping is just copying it to disk. |
| 06:16.11 | mishehu | i.e. dvdbackup.c |
| 06:16.19 | afrosheen | well when most people say 'ripping this or that' they mean rip + reencode |
| 06:16.30 | mishehu | I know, I'm just being technical ;-) |
| 06:16.32 | afrosheen | like, you rip an audio cd, it usually ends up a nice stack of mp3s |
| 06:16.33 | afrosheen | ;) |
| 06:16.45 | afrosheen | or oggs |
| 06:18.11 | mishehu | oddly enough,, when I lame some cd's, it's slower than if I ogg/vorbis encode them |
| 06:19.01 | x86 | lame uses a constant bit rate, which requires more CPU, and vorbis is usually variable |
| 06:19.26 | mishehu | x86: I thought that lame can do both cbr and vbr |
| 06:20.20 | hads|home | It can |
| 06:20.27 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 06:22.35 | SaTLaN32 | hello |
| 06:22.39 | SaTLaN32 | i need some help |
| 06:22.44 | SaTLaN32 | i have a 4 fxo card, everything is working, but when i hang up either sides of the call, asterisk still hold both of them, and not being disconnected |
| 06:22.58 | SaTLaN32 | the dial string i\m using is: ZAP/g1/*43${cldid},60 |
| 06:23.35 | *** join/#asterisk bugz (n=will@cpe-70-123-122-41.houston.res.rr.com) |
| 06:25.27 | *** join/#asterisk vlt (n=daniel@dslb-088-073-202-029.pools.arcor-ip.net) |
| 06:26.27 | x86 | hads|home: last i knew lame could only do CBR |
| 06:30.58 | hads|home | x86: Dunno, I thought it did VBR. Doesn't matter though, not really Asterisk talk :) |
| 06:33.52 | *** join/#asterisk firewired (n=firewire@124.104.11.51) |
| 06:34.06 | SaTLaN32 | anyone here can help |
| 06:42.51 | tzafrir | SaTLaN32, soft hangup helps to disconnet? |
| 06:43.09 | tzafrir | Also: any chance busydetect will help here? |
| 06:43.54 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 06:59.22 | *** part/#asterisk hatamen (n=hatamen@222.183.23.72) |
| 06:59.28 | *** part/#asterisk BlepsoaF (n=pbaker@ool-457805b1.dyn.optonline.net) |
| 07:04.19 | *** join/#asterisk mikeeeeee (n=fdsfs@091.pth0504.pth.iprimus.net.au) |
| 07:04.31 | mikeeeeee | hi, has anyone got h.264 running in asterisk? |
| 07:04.39 | *** join/#asterisk lopt (i=komodo@204-9-8-245.inetlink.ca) |
| 07:05.15 | mikeeeeee | i need it for some grandstream gvx3000 phones to run in asterisk and they only support h.264? i kind of compiled the newest asterisk from the trunk sources, but it never worked properly. How can i get this working as its quite mission critical i get it up asap |
| 07:06.02 | lopt | I'm having a pri problem. My Zaptel and Zapata configs have not changed in over a year but now I'm getting this No D-channels available! Using Primary channel 24 as D-channel anyway any thoughts? |
| 07:06.10 | mikeeeeee | was running trixbox, got the latest zaptel, asterisk and asterisk-addons, zaptel and asteisk compile fine, but not the addons |
| 07:07.26 | mikeeeeee | i can actually start asteisk and get into the cli, but i go into freepbx, and when i goto the extensions, it says asterisk manager not started. |
| 07:09.04 | bugz | anyone have experience with the grandstream GXV-3000 |
| 07:09.16 | mikeeeeee | yeh i had the running |
| 07:09.26 | bugz | hows the performance? |
| 07:09.36 | mikeeeeee | somehow, dont ask me, but asterisk never worked properly, like coudl get shit from freepbx but the cli would nolonger run |
| 07:09.39 | mikeeeeee | umm excellent |
| 07:09.51 | mikeeeeee | on lan, im yet to try them over net |
| 07:10.00 | mikeeeeee | will do as soon as i get this asterisk working properly |
| 07:10.06 | bugz | i cant believe how inexpensive it is. |
| 07:10.11 | mikeeeeee | yeh |
| 07:10.17 | mikeeeeee | i payed 1300 AUD for 3 |
| 07:10.35 | bugz | ive found them on the net for around $260 US |
| 07:10.35 | mikeeeeee | can you help me with h.264 in * :)?? |
| 07:10.39 | mikeeeeee | nice |
| 07:10.53 | bugz | havent done that yet but i can try |
| 07:11.07 | bugz | ive built a couple pbx's though ;) |
| 07:11.14 | mikeeeeee | cool |
| 07:11.28 | mikeeeeee | ok i managed to compile it all, had to miss some modules in addons |
| 07:11.36 | mikeeeeee | but when i goto freepbx and then the extensions bit |
| 07:11.46 | mikeeeeee | it says cannot connect to the asterisk manager |
| 07:12.05 | mikeeeeee | i can start asterisk with asterisk -c and then get into the CLI |
| 07:12.09 | mikeeeeee | and it is running |
| 07:12.14 | mikeeeeee | any idea what im doin wrong? |
| 07:12.41 | mikeeeeee | i installed latest trixbox, then went off and compiled new zaptel asterisk and addons |
| 07:13.09 | *** join/#asterisk JonZombie (n=JonZombi@71-8-63-16.dhcp.leds.al.charter.com) |
| 07:13.16 | bugz | manager.conf is configured? |
| 07:13.25 | JonZombie | Hi |
| 07:13.32 | bugz | i'd assume the corresponding codec/module file would need to be configured to log into AMI |
| 07:14.19 | JonZombie | I need some help with my Zaptel TDM card. |
| 07:14.36 | bugz | JonZombie: RMA it... |
| 07:14.41 | bugz | best advice i can give |
| 07:14.58 | JonZombie | why? |
| 07:15.31 | bugz | sangoma has a superior product if you can get it working |
| 07:15.55 | JonZombie | I got it installed, and configured |
| 07:16.07 | JonZombie | problem is, it holds the line offhook. |
| 07:17.47 | JonZombie | Am I missing something. |
| 07:18.05 | JonZombie | With 252 people, I'm surprised no one has anything to say. :) |
| 07:19.11 | bugz | there is a reason for that |
| 07:19.14 | bugz | replace the card |
| 07:19.29 | mikeeeeee | what needs config in manager.cong |
| 07:19.30 | mikeeeeee | what needs config in manager.conf |
| 07:19.32 | mikeeeeee | ? |
| 07:19.38 | bugz | mikeeeeee: a login |
| 07:19.56 | mikeeeeee | mmm ok thought it woudl be there |
| 07:22.09 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 07:23.44 | *** join/#asterisk backblue (n=moo@87-196-67-13.net.novis.pt) |
| 07:30.40 | hads|home | JonZombie: There's nothing wrong with the TDM400, bugz appears to be single minded. |
| 07:31.41 | hads|home | JonZombie: Usually the problem with holding lines open comes down to not having disconnect supervision from the telco. |
| 07:33.35 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 07:44.21 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:45.07 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:46.23 | JonZombie | Is there anything I can do about it? |
| 07:46.39 | JonZombie | As soon as I plug a phone line into the card, it goes offhook |
| 07:46.51 | hads|home | Ah, different story. |
| 07:47.14 | hads|home | All the lines/modules? |
| 07:47.24 | SaTLaN32 | i have a strange problem. |
| 07:47.31 | JonZombie | I only have one module (fxo) |
| 07:47.47 | hads|home | OK, tried a different cable? |
| 07:47.47 | *** join/#asterisk marta (n=a@host100-5.pool8711.interbusiness.it) |
| 07:47.55 | JonZombie | I have tried putting it into all 4 slots, same results |
| 07:47.57 | marta | hello |
| 07:48.03 | JonZombie | I have used 2 different cables |
| 07:48.12 | *** join/#asterisk fafnir (n=notfaf@unaffiliated/fafnir) |
| 07:48.21 | SaTLaN32 | when i use Read() and dial the dtmf's, everything is ok, beside the time i dial at least 2 fast "0" which hangs the call |
| 07:48.38 | SaTLaN32 | everything else (1111,4444,etc.) is doing ok. |
| 07:49.48 | hads|home | JonZombie: Interesting. I can't think of anything that would do that off the top of my head. It could be a faulty module/card. |
| 07:50.03 | JonZombie | That's the conclusion I have come to. |
| 07:50.25 | JonZombie | I bought it on Ebay... I am guessing the guy wanted to dump a dead card on someone. |
| 07:50.39 | JonZombie | I went back and looked again, his return policy is 3 days. |
| 07:50.57 | JonZombie | I suspect he knew it was bad. |
| 07:51.22 | hads|home | Bummer |
| 07:51.29 | *** join/#asterisk infoaddict (n=infoaddi@user-0c9h7dt.cable.mindspring.com) |
| 07:52.11 | JonZombie | next time I will buy one from a vendor so I can get a warranty on it. |
| 07:53.09 | Qwell | JonZombie: What color is the module? |
| 07:53.15 | JonZombie | red |
| 07:53.32 | Qwell | yeah, it's an fxo.. |
| 07:54.20 | JonZombie | I had a X100P that was working fine... till some kids broke into my house and stole the computer. |
| 07:54.36 | JonZombie | Sad thing is that the computer video taped them breaking in. |
| 07:54.41 | hads|home | Not having a good run then. |
| 07:54.47 | Qwell | JonR800: nice... |
| 07:54.50 | Qwell | erm, JonZombie |
| 07:55.49 | JonZombie | I never thought anyone would go into the closet and still a computer. |
| 07:56.08 | JonZombie | I thought theives knew computers were worthless at pawn shops. |
| 07:57.32 | JonZombie | They stole 2 IP phones, and the Asterisk Server. |
| 07:58.08 | JonZombie | So, now I am building a new one. |
| 07:58.26 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 07:58.29 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 08:05.50 | *** part/#asterisk JonZombie (n=JonZombi@71-8-63-16.dhcp.leds.al.charter.com) |
| 08:16.52 | *** part/#asterisk firewired (n=firewire@124.104.11.51) |
| 08:23.42 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
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| 08:31.26 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 08:33.00 | Newbie___ | hi all, when i make outbound voip calls. i get background noise. is there a way to eliminate it ? |
| 08:33.53 | *** join/#asterisk garuse (n=stelu@80.97.71.230) |
| 08:34.19 | sxpert | Newbie___: you need some echo-cancellation fu |
| 08:35.30 | Newbie___ | sxpert: tks but is sip-sip calls. where do i put the echo cancellation ? in sip.conf? |
| 08:35.56 | sxpert | possibly. |
| 08:37.00 | Newbie___ | sxpert: hrm all along i though echo cancellation can only be done when TDM is use |
| 08:37.08 | X-Rob | Echo cancellation is done in zaptel |
| 08:37.12 | X-Rob | (and only zaptel) |
| 08:37.24 | X-Rob | if you're not using zaptel, you don't need echo cancellation |
| 08:37.26 | X-Rob | well |
| 08:37.30 | X-Rob | can't have it |
| 08:37.42 | hads|home | SIP echo is possibly related to handsets. |
| 08:38.09 | Newbie___ | hads|home: is not echo. is background static noise |
| 08:38.38 | hads|home | OK. That could be handsets too. |
| 08:38.53 | X-Rob | SaTLaN32, sounds like you're matching XNNNN rather than X. |
| 08:39.03 | Newbie___ | ok |
| 08:39.07 | hads|home | Well, when you say outbound calls, are you making calls out a provider. |
| 08:39.10 | X-Rob | (ps, DTMF detection is done in software, nothing to do with hardware) |
| 08:39.37 | sxpert | I have issues with the meetme app. it doesn't do what it says on the box :D |
| 08:40.01 | hads|home | Hmm, questions look odd when you forget the question mark. |
| 08:40.05 | Newbie___ | hads|home: yes SIP-voipbuster |
| 08:40.31 | hads|home | So there is quite possibly PSTN involved. |
| 08:40.34 | sxpert | at some point in my dialplan, I have "exten => 8600,n,Meetme(,1qdEMp)" |
| 08:40.46 | sxpert | well. despite the 'q', I get a ton of prompts |
| 08:40.51 | Newbie___ | rather asterisk-voipbuster using sip protocol |
| 08:41.11 | sxpert | like "you are entering conference blah" |
| 08:51.41 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
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| 10:01.26 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 10:02.02 | EmleyMoor | I'm getting "No route to destination" when I try to make a call over FWD |
| 10:02.58 | EmleyMoor | I can still call internal numbers, but FWD is not working |
| 10:04.55 | EmleyMoor | How do I trace why that would be? I've tried restarting asterisk to no avail |
| 10:11.43 | EmleyMoor | It worked last night, up to a point |
| 10:11.48 | *** join/#asterisk littleball (n=littleba@cm82.epsilon172.maxonline.com.sg) |
| 10:13.23 | littleball | hello, i am looking for a asterisk box hosting ISP provider because i need to be able to connect to E1 easily and also be able to get DDI number for this E1 line. who can recommend? I need super voice quality for my special application |
| 10:17.45 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 10:26.40 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
| 10:27.35 | EmleyMoor | Aug 27 11:27:24 NOTICE[17566]: chan_iax2.c:7875 iax2_poke_noanswer: Peer 'iaxfwd' is now UNREACHABLE! Time: 0 |
| 10:27.52 | EmleyMoor | Been getting that since sometime last night - it worked before and I didn't change anything |
| 10:33.14 | sxpert | ah HA... my super-quiet option works ;D |
| 10:33.35 | KDan | is there an option/command to validate whether a jar is a valid jar? |
| 10:33.54 | KDan | woops |
| 10:33.56 | KDan | wrong channel |
| 10:34.01 | sxpert | lol |
| 10:42.03 | EmleyMoor | I suspect fwd may have a fault - is it fairly easy to set up two-way SIP with them in asterisk? (my asterisk server is on a public IP) |
| 10:42.26 | *** join/#asterisk Lefels (n=vector@dsl-146-99-57.telkomadsl.co.za) |
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| 10:53.28 | EmleyMoor | Is there some other service I can sign up to for free to do some testing, if I am NOT in the USA? |
| 10:55.36 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:57.13 | kumamoto | voipdiscount & sipdiscount |
| 10:57.31 | kumamoto | they are the same thing |
| 10:58.00 | EmleyMoor | I'm trying to register for fwd's forums to report the fault - but it's taking forever to send me a confirmation# |
| 11:02.02 | EmleyMoor | This is ridiculous |
| 11:05.19 | *** join/#asterisk alawguy (n=mike@85-124-36-191.dynamic.xdsl-line.inode.at) [NETSPLIT VICTIM] |
| 11:05.39 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 11:08.55 | EmleyMoor | OK - suppose I do sign up for sipdiscount - is there a sample configuration entry for asterisk for it? |
| 11:12.03 | kumamoto | http://www.invisible.ca/space/voip-projects/asterisk-with-sipdiscount-howto |
| 11:14.36 | aigroine | hi ppl |
| 11:15.30 | *** join/#asterisk xblx (n=mbe@ip51cf8e39.direct-adsl.nl) |
| 11:24.22 | *** part/#asterisk xblx (n=mbe@ip51cf8e39.direct-adsl.nl) |
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| 11:55.22 | jhiver | hi all |
| 11:55.45 | jhiver | i must be blind or something but I have a strange issue |
| 11:55.58 | jhiver | in my sip.conf, i have a peer called [gecko] |
| 11:56.06 | jhiver | and on the asterisk CLI, I see: |
| 11:56.20 | jhiver | <PROTECTED> |
| 11:56.21 | jhiver | <PROTECTED> |
| 11:56.21 | jhiver | Aug 27 13:42:24 WARNING[16714]: chan_sip.c:1980 create_addr: No such host: gecko |
| 11:56.26 | jhiver | what's going on? |
| 12:10.34 | tzafrir | (next time use a pastebin) |
| 12:12.53 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 12:15.19 | *** join/#asterisk warthog (n=warthog@16.56.233.220.exetel.com.au) |
| 12:15.56 | warthog | anyone know how to deal with rr_delay=xx, iax2 protocol failure |
| 12:16.20 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:16.24 | warthog | is see very little about this on google or asterisk knowledgebase |
| 12:17.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:17.58 | puzzled | hi |
| 12:18.52 | warthog | hello |
| 12:28.09 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 12:28.31 | areq | QQ |
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| 13:00.45 | BugKham | does anyone have "Action: PlayDTMF" present in 1.2.11 |
| 13:01.57 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:04.30 | russellb | did it ever exist? |
| 13:14.35 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 13:14.41 | jhiver | so lads |
| 13:15.12 | jhiver | why am i getting this 'chan_sip.c:1980 create_addr: No such host: mediant |
| 13:15.12 | jhiver | ' error |
| 13:15.27 | jhiver | when i _do_, in fact, have a [mediant] in sip.conf? or do I, |
| 13:15.29 | jhiver | ? |
| 13:15.49 | Strom_C | did you put a host= line in [mediant] ? |
| 13:16.03 | RoyK | perhaps, just perhaps, you have added host=mediant? |
| 13:16.21 | Strom_C | that too |
| 13:16.48 | jhiver | no it looks like the conf is ok |
| 13:16.56 | jhiver | strange... |
| 13:16.58 | RoyK | jhiver: i beleive it isn't :P |
| 13:17.02 | RoyK | jhiver: pastebin the lot |
| 13:17.08 | BugKham | russellb: 1.2.8 |
| 13:17.18 | BugKham | russellb: that's I saw from the wiki |
| 13:17.40 | *** join/#asterisk saftsack (n=saftsack@p54A7E0C6.dip.t-dialin.net) |
| 13:17.48 | BugKham | russellb: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+PlayDTMF |
| 13:17.55 | jhiver | actually something kind of wierd |
| 13:18.08 | jhiver | when i do sip show peers, mediant doesn't appear in the list of peers |
| 13:18.24 | Druken | did you reload since adding it? |
| 13:18.34 | jhiver | yes |
| 13:18.40 | jhiver | i'll paste bin the stuff |
| 13:18.43 | Druken | then you fucked something up... |
| 13:19.59 | jhiver | http://pastebin.ca/150894 |
| 13:20.35 | jhiver | can you see something wrong in the conf? |
| 13:20.50 | Druken | i don't see a TYPE |
| 13:21.02 | jhiver | aaah ;) |
| 13:21.08 | jhiver | i was being blind :) |
| 13:21.12 | jhiver | thanks man :) |
| 13:23.42 | BugKham | russellb: I never use it either but just wanna make use of it now |
| 13:25.29 | *** join/#asterisk hatamen (n=hatamen@222.183.23.72) |
| 13:26.21 | jhiver | Druken, thanks so much man, it's good to have an extra pair of eyeballs sometimes when you're stuck on something that silly |
| 13:27.38 | Strom_C | jhiver, the official term is "sanity check" :D |
| 13:28.54 | jhiver | lol |
| 13:30.26 | jhiver | or CRC : Completely Ridiculously and some word that would start with C making this a funny line |
| 13:30.27 | jhiver | :) |
| 13:30.47 | Strom_C | http://catb.org/~esr/jargon/html/S/sanity-check.html |
| 13:31.59 | BugKham | anyway to interrupt Playback in a particular channel uisnf manager API? |
| 13:32.09 | Druken | completely riciculous calamity? |
| 13:32.12 | BugKham | I can't find anything about this |
| 13:32.42 | BugKham | Druken: ? |
| 13:32.52 | Strom_C | Druken, i prefer "completely ridiculous cock-up" |
| 13:33.51 | BugKham | Druken: are you using manager api at all? |
| 13:34.11 | Druken | newp |
| 13:34.17 | *** join/#asterisk littleball (n=littleba@cm82.epsilon172.maxonline.com.sg) |
| 13:34.47 | BugKham | just wanna know if anyone has tried "Action: PlayDTMF" |
| 13:35.09 | BugKham | it doesn't seem to exist in 1.2.11 |
| 13:35.18 | Druken | ask in the mailing list... ? |
| 13:36.00 | BugKham | it's strange that russellb hasn't heard about it |
| 13:36.32 | BugKham | Druken: yeah, probably that's a better way |
| 13:37.52 | BugKham | Druken: and do you know how to process other things in the same channel while in "Playback" or "Background" ? |
| 13:38.44 | BugKham | Druken: from the wiki, this doesn't seem possible |
| 13:39.06 | Druken | you want to do what ? |
| 13:39.36 | BugKham | Druken: ok |
| 13:39.56 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
| 13:40.00 | gambolputty | Hi. I want a called party to hear something before the calling and called parties actually talk together. Is this possible? I have tried the G option of the Dial command and it doesn't do this so far. |
| 13:40.12 | BugKham | I want the callers to listen to music while waiting for the "http post" outcome |
| 13:40.54 | Druken | your using agi of course, i belive there's a thing in agi that allows that.... |
| 13:41.07 | BugKham | I'm using agi |
| 13:41.22 | Druken | todo processing while moh or playback or something... |
| 13:41.22 | BugKham | but still can't see a way |
| 13:41.59 | BugKham | so I will use stream_file() |
| 13:42.10 | BugKham | plus Action: PlayDTMF |
| 13:42.16 | BugKham | to make it possible |
| 13:42.56 | BugKham | u know where to find that info, processing with moh or playback |
| 13:44.03 | Druken | been a long time since i've seen it... i tend not to play with agi... it frusterates me :) |
| 13:44.48 | BugKham | agi is really good to implement complicated IVR systems |
| 13:45.37 | BugKham | I only see this on the wiki "Background: Play a sound file while processing other commands " |
| 13:45.57 | brimstone | BugKham, try calling background, then working whatever you need to process the http post |
| 13:46.03 | BugKham | but actually Background cannot do it |
| 13:46.48 | BugKham | brimstone: process at the next priority? |
| 13:47.12 | brimstone | nah, process as in whatever your script needs to do |
| 13:47.36 | brimstone | then call a playback or noop or something with the results |
| 13:47.45 | brimstone | man, if i had time, i'd play with that now |
| 13:48.55 | BugKham | brimstone: ok, i'll give it a try |
| 13:49.11 | brimstone | snazzy, let me know how it turns out |
| 13:49.15 | brimstone | i'm curious too |
| 13:55.41 | BugKham | i'm looking for a large file for Background =) |
| 13:56.56 | brimstone | demo-congrats? |
| 13:58.13 | Strom_C | spam.gsm? |
| 13:58.43 | BugKham | brimstone: doesn't work |
| 14:00.57 | BugKham | brimstone: I did exec('Background','demo-instruct'); |
| 14:00.59 | brimstone | ahh, well, not sure right off then, sorry |
| 14:01.22 | BugKham | then wait_for_digit(10000); |
| 14:01.27 | BugKham | and exit; |
| 14:01.43 | BugKham | the file is longer than 10 secs |
| 14:02.04 | BugKham | if it works it should stop at 10th sec, right? |
| 14:02.50 | brimstone | not sure, since you're asking asterisk to pause |
| 14:03.00 | brimstone | try a 10sec pause in your agi language |
| 14:03.16 | brimstone | then exit |
| 14:05.33 | brimstone | let me know how it works out BugKham |
| 14:19.09 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.191) |
| 14:25.36 | *** part/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
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| 14:54.51 | *** join/#asterisk lordbaron (n=redbaron@216.133.155.50) |
| 14:55.54 | lordbaron | what is the proper way to config the TE412P for PRI_NET? I am connecting 2 for stress test. I need to know how to config the timing on the NET/CPE sides |
| 14:56.18 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:56.57 | zoa | put 1 on 0 and 1 on 1 |
| 14:57.49 | lordbaron | so..pri_net side: span1=1,1,0,esf,b8zs; span2=1,2,0,esf,b8zs, etc? |
| 14:57.49 | *** join/#asterisk doolph (n=doolph@200.124.28.155) |
| 14:58.12 | lordbaron | err span2=2,2,0,esf,b8zs |
| 14:58.23 | *** join/#asterisk krausen (n=krausen@cpe-24-170-62-63.stx.res.rr.com) |
| 14:58.35 | lordbaron | and of course...span=2,2,0,esf,b8zs |
| 14:58.37 | lordbaron | can't type today |
| 14:59.07 | krausen | question: Is it possible to have an asterisk release installed and asterisk from SVN, and be able to switch inbetween them on a whim? |
| 14:59.19 | lordbaron | and then on the cpe side, second position is 0 for all? |
| 14:59.24 | lordbaron | makes sense |
| 15:01.34 | krausen | anyone else messing with asterisk/googletalk in svn? |
| 15:03.38 | lordbaron | <krausen> I do this for testing, but it requires that you remove all your modules, etc. |
| 15:05.27 | krausen | so might be easier to just cron the SVN pull, and re-build/install if I want to monkey with it, then do "make install" again in the release directories if I want to toggle back? |
| 15:10.46 | tzafrir | krausen, get yourself a nice little chroot jail. Build and install Asterisk there, until you're happy with it |
| 15:10.55 | tzafrir | (in debian: use debootstrap) |
| 15:11.25 | krausen | thanks |
| 15:12.22 | tzafrir | However this will not be as effective if you want the program to listen on some ports, unless you have a space IP on that system |
| 15:13.26 | krausen | talking about concurrently with another asterisk running? |
| 15:16.05 | ctaloi | hey guys - do you know if issuing a 'reload' impacts calls waiting in queue? |
| 15:21.47 | *** join/#asterisk litnimax (n=chatzill@host-86-106-219-208.moldtelecom.md) |
| 15:23.05 | litnimax | hello folks. I am trying to use ExternalIVR . I don't get H event (H: the channel was hung up by the connected party). Anyone can advise? |
| 15:24.34 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 15:28.26 | *** join/#asterisk mtgco (n=Techie@static-71-125-10-2.nycmny.fios.verizon.net) |
| 15:29.53 | mtgco | greetings having trouble compiling asterisk-addons seem to have all of the packages but getting this error: app_addon_sql_mysql.c:78: warning: data definition has no type or storage class |
| 15:31.01 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 15:31.02 | blitzrage | initial guess is that you're missing the mysql-devel package |
| 15:31.40 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 15:33.00 | mtgco | thought that two but a check of packages in web min shows its there mysql-devel 4.1.20-1.RHEL4.1 also did a yum install mysql-devel and it informed me that there was nothing to do???? |
| 15:34.48 | mtgco | did a pretty good google search and that didnt turn up anything either |
| 15:35.18 | *** part/#asterisk krausen (n=krausen@cpe-24-170-62-63.stx.res.rr.com) |
| 15:35.59 | blitzrage | hrmmmm... not really sure. Compiling should be fairly straight forward... |
| 15:36.13 | blitzrage | although I've never tried to compile that package, so it might be expecting something else |
| 15:36.45 | mtgco | odly enough i would have thought the ./confugure would have flaged an error if i didnt have the correct complier or packages and that ran clean |
| 15:38.16 | mtgco | any way to find out what packages asterisk-addons requires to compile? |
| 15:39.47 | tzafrir | mtgco, try to build it. It should fail pretty early on (by rpm) |
| 15:40.11 | tzafrir | Not sure if there's anything better. My rpm experince is a bit rusty |
| 15:40.53 | hacked`` | guys |
| 15:40.56 | hacked`` | lets say i want to accept 4 incoming calls at the same time |
| 15:41.00 | mtgco | i am building asterisk-addons from svn tzafrir that is what is failing |
| 15:41.01 | hacked`` | whats a good provider to go with |
| 15:42.15 | tzafrir | mtgco, a missing mysql dev package? |
| 15:42.51 | tzafrir | *mysql*-devel . Shouldn't be too many of those in your distro |
| 15:43.09 | mtgco | yeah checked that have msql-devel and yum says no more action heres the error again:app_addon_sql_mysql.c:78: warning: data definition has no type or storage class |
| 15:44.53 | mtgco | app_addon_sql_mysql.c:78: warning: data definition has no type or storage class |
| 15:44.53 | mtgco | app_addon_sql_mysql.c: In function `unload_module': |
| 15:44.53 | mtgco | app_addon_sql_mysql.c:420: error: `STANDARD_HANGUP_LOCALUSERS' undeclared (first use in this function) |
| 15:44.53 | mtgco | app_addon_sql_mysql.c:420: error: (Each undeclared identifier is reported only once |
| 15:44.53 | mtgco | app_addon_sql_mysql.c:420: error: for each function it appears in.) |
| 15:44.54 | mtgco | app_addon_sql_mysql.c: At top level: |
| 15:44.56 | mtgco | app_addon_sql_mysql.c:441: warning: data definition has no type or storage class |
| 15:44.58 | mtgco | make[1]: *** [app_addon_sql_mysql.o] Error 1 |
| 15:45.00 | mtgco | make[1]: Leaving directory `/usr/src/asterisk-addons' |
| 15:45.02 | mtgco | make: *** [all] Error 2 |
| 15:45.16 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:45.16 | Strom_C | holy christ, man |
| 15:45.23 | Strom_C | have you not heard of pastebin? |
| 15:45.27 | Strom_C | ~pb |
| 15:45.30 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 15:45.52 | mtgco | sorry all was trying to paste that |
| 15:48.43 | *** join/#asterisk saftsack (n=saftsack@p54A7E0C6.dip.t-dialin.net) |
| 15:51.31 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:52.10 | mtgco | here is the full error that i get when i do the make of asterisk-addons: http://pastebin.com/777261 |
| 15:53.08 | *** join/#asterisk FTexcom (n=strapse@169.Red-88-2-81.staticIP.rima-tde.net) |
| 15:53.28 | *** join/#asterisk bmg505 (n=leon@dsl-146-17-228.telkomadsl.co.za) |
| 15:55.25 | *** join/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 15:55.42 | mmurdock | Howdy everyone. |
| 15:56.39 | hacked`` | guys |
| 15:56.44 | hacked`` | is there some sort of review site |
| 15:56.47 | hacked`` | for voip providers |
| 15:56.52 | hacked`` | dont know who to choose |
| 15:57.01 | hacked`` | for like 3 incoming lines, and 2-3 outgoing |
| 15:57.21 | mtgco | check out http://nervittles.com he has a few reviewed |
| 15:57.30 | hacked`` | im looking there now |
| 15:57.37 | hacked`` | cant find any that are good |
| 15:58.00 | mmurdock | I've got a TDM2400 with the echo cancellation on it and when I have that daugter card plugged in I can't hear the phone ringing or the caller, but they can hear me. When I take it out, everthing seems to work normally. Any suggestions? |
| 15:58.30 | FTexcom | mmurdock try playing with the rxgain and txgain on the zapata.conf |
| 15:58.32 | russellb | mmurdock: have you contacted digium support? |
| 15:58.33 | Strom_C | mmurdock, FXO or FXS modules? |
| 15:58.51 | blitzrage | hacked``: www.asteriskguru.com has some too |
| 15:58.57 | hacked`` | like im interested in telasip, but the thing is, whats the difference between residential and business plans, the residential plans look more attractive |
| 15:59.00 | mmurdock | Strom_C: I have both. |
| 15:59.15 | Strom_C | mmurdock, have you contacted digium support? |
| 15:59.30 | mmurdock | Strom_C: I have not. Thought I would try here first. |
| 15:59.46 | Strom_C | mmurdock, well, ok...wait till tomorrow and contact them :) |
| 15:59.54 | mmurdock | Strom_C: Will do. |
| 16:00.56 | hacked`` | anyone here use telasip ? |
| 16:09.03 | *** join/#asterisk s0lid (n=s0lid@210.213.242.179) |
| 16:19.20 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:20.30 | mmurdock | is rxgain the only way to increase the volume of a call other then the volume control on the phone? |
| 16:21.06 | Strom_C | mmurdock, you use rxgain to compensate for attenuation on analog trunk circuits |
| 16:22.12 | Strom_C | mmurdock, where are you located? |
| 16:22.29 | mmurdock | Strom_C: k, by the way I got that card to work. I had to turn off echotraining in the zapata file. I'm in Utah |
| 16:22.36 | lordbaron | I have a pots line that will not use KewlStart on my TDM400P..does this indicate a line problem, or just an incompatibity with the telco? |
| 16:22.52 | Strom_C | lordbaron, what do you mean, exactly |
| 16:23.17 | Strom_C | mmurdock, find the number to your switch's local milliwatt test and use that to tune the rxgain on your FXO lines |
| 16:23.17 | lordbaron | strom_c: when I use KS, and dial out, it dials, and then hangs up. Status shows ANSWERED |
| 16:23.38 | Strom_C | lordbaron, what kind of switch is your line served out of? |
| 16:23.41 | lordbaron | if I use LS, then dials like normal, but I do not get disconnect supervision |
| 16:24.04 | lordbaron | SWBell - Dallas, TX. I don't know much more, but I did confirm that they placed disconnect supervision on it. |
| 16:24.10 | lordbaron | What more should I ask the telco? |
| 16:24.45 | Strom_C | lordbaron, is your line doing something like polarity reversal upon answer supervision? |
| 16:24.54 | mmurdock | Strom_C: I will do that when I put this machine into service. Right now I am testing it using my Vonage phone line. |
| 16:25.03 | Strom_C | mmurdock, EWWWW |
| 16:25.04 | lordbaron | I do see those messages if I start the mod with a debug=1 |
| 16:25.13 | mmurdock | Strom_C: :) |
| 16:25.14 | Strom_C | lordbaron, which messages |
| 16:25.23 | lordbaron | POLARITY REVERS 0 => 1 |
| 16:25.37 | Strom_C | do you have a way of testing line polarity? |
| 16:25.55 | lordbaron | I probably have the tools, and just lack the knowledge :> |
| 16:25.57 | mmurdock | Strom_C: Just a side note, I volunteered to setup this Asterisk box in a new school. We will be running 30+ phones off it. |
| 16:26.17 | Strom_C | mmurdock, 30+ analog phones? |
| 16:26.40 | mmurdock | Strom_C: Nope, 30- Snom 300's and 3 Snom 320's |
| 16:26.44 | Strom_C | ok, good :) |
| 16:26.59 | lordbaron | I have a buttset, with a red light. I swapped the tip/ring, and the red light went away |
| 16:27.01 | mmurdock | I really like how the Snom phones have worked out. |
| 16:27.12 | Strom_C | lordbaron, ok, do this |
| 16:27.30 | Strom_C | lordbaron, hook up the buttset correctly and then dial a local call |
| 16:27.35 | *** part/#asterisk mtgco (n=Techie@static-71-125-10-2.nycmny.fios.verizon.net) |
| 16:27.41 | Strom_C | lordbaron, tell me whether the red light goes away after that call answers |
| 16:27.54 | lordbaron | yes |
| 16:28.13 | Strom_C | ok, then you have answer supervision on your line, not disconnect-only supervision |
| 16:28.19 | lordbaron | so...use reversepolarityonanswer=yes? |
| 16:28.26 | lordbaron | oic |
| 16:28.26 | *** join/#asterisk Splas (n=jwb@gate.lga2.us.voxel.net) |
| 16:28.42 | Strom_C | i think you use that option |
| 16:28.44 | lordbaron | is that a problem for KS? |
| 16:28.48 | Strom_C | it's not a KS line :) |
| 16:28.55 | Strom_C | it's an LS line with answer supervision |
| 16:29.02 | lordbaron | ic, so I need to use LS..gotcha |
| 16:29.07 | *** part/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 16:29.31 | lordbaron | if I enable that, will that help fix the disconnect problem? Rightnow, it is not working. |
| 16:29.46 | Strom_C | reversepolarityonanswer=yes in zapata.conf? |
| 16:30.00 | lordbaron | Yes |
| 16:30.25 | Strom_C | give it a shot. I don't have a line that supervises like that, so I can't test it |
| 16:30.30 | lordbaron | I have not tested that part yet, To test, I grab a zap line, dial direct to a cell phone, answer, and then hangup the cell phone After the telco 'blips', I get a fastbusy, but b |
| 16:30.41 | lordbaron | busydetect nevers detects this fast busy |
| 16:30.43 | Strom_C | fast busy == reorder |
| 16:30.46 | lordbaron | it is 10ms on, 20 of |
| 16:30.53 | Strom_C | fast busy is not a busy signal |
| 16:31.12 | lordbaron | oic..so busydetect will never get that huh? |
| 16:31.29 | Strom_C | look, youve GOT answer supervision on the line. you don't need busydetect :) |
| 16:31.39 | lordbaron | ok, makes sense |
| 16:31.40 | lordbaron | thanks |
| 16:31.46 | Strom_C | but please test that |
| 16:31.49 | Strom_C | and let me know if it works |
| 16:31.54 | lordbaron | will do now..thanks |
| 16:32.29 | Strom_C | lordbaron, where is the reversepolarityonanswer option documented? |
| 16:34.55 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 16:36.09 | UForgotten | how would I configure a zap channel to wait 6 rings before answering? |
| 16:36.32 | Strom_C | figure out how many seconds 6 rings is, and then issue a Wait() before the Answer() |
| 16:36.57 | Strom_C | but why would you want to do that? |
| 16:37.21 | Dovid | morning |
| 16:37.26 | UForgotten | because my wife is getting pissed that the pbx is answering before she can get to it, then she can't shut it off when it does |
| 16:37.26 | zoa | evening |
| 16:37.41 | Dovid | the variable chanisavail only checks to see if asterisk is using that channel |
| 16:37.42 | Dovid | ? |
| 16:37.44 | UForgotten | If she answers first, meaning we're home, I don't want it to answer at all, so the wait won't work |
| 16:37.59 | zoa | so that woulld work |
| 16:38.07 | Strom_C | UForgotten, asterisk is not designed to function as an adjunct to an existing home telephone network |
| 16:38.11 | zoa | maybe its easier to find another wife :) |
| 16:38.15 | Dovid | i want to have both asterisk and a fax machine plugged into a pots line and before asterisk tries to make a call to see if the fax machine is using that channel |
| 16:38.19 | UForgotten | lol, I had considered that |
| 16:38.38 | Strom_C | UForgotten, the assumption is that any analog lines will be trunk lines between a telco and the pbx |
| 16:38.42 | Dovid | anyone ? |
| 16:38.46 | Strom_C | Dovid, that is not how chanisavail works at all |
| 16:38.46 | zoa | ~secondhandwifes |
| 16:39.03 | zoa | hmm no bot |
| 16:39.03 | Dovid | Srom_C: its only for internal like for asteirsk calls ? |
| 16:39.04 | Strom_C | Dovid, asterisk cannot check to see if the pots line is busy before picking it up |
| 16:39.08 | Dovid | hmm |
| 16:39.09 | *** part/#asterisk clive- (n=pirch@dsl-145-13-144.telkomadsl.co.za) |
| 16:39.29 | Dovid | there has to be a way. I know that my $30.00 cordless can see if the line is in use or not |
| 16:39.45 | Strom_C | Dovid, im answering roughly the same question for UForgotten |
| 16:39.57 | Strom_C | asterisk is not designed to function as an adjunct to an existing home telephone network |
| 16:40.05 | Strom_C | the assumption is that any analog lines will be trunk lines between a telco and the pbx |
| 16:40.13 | Dovid | Strom_C: can it be patched to check the line ? |
| 16:40.16 | Strom_C | and therefore will not have anything else connected to them |
| 16:40.16 | Dovid | i am willing to pay |
| 16:40.25 | Strom_C | Dovid, run your fax machine through the pbx |
| 16:40.32 | Strom_C | fxs port, fxo port |
| 16:40.37 | Dovid | Strom_C: i did that but now they dont go thru |
| 16:40.38 | Dovid | did that |
| 16:40.52 | Strom_C | did you configure your rxgain and txgain correctly? |
| 16:40.57 | Dovid | hm |
| 16:41.04 | Dovid | they are good for voice calls |
| 16:41.07 | UForgotten | also mus use ulaw or alaw |
| 16:41.08 | Dovid | how do i ajust them ? |
| 16:41.18 | UForgotten | ~echo |
| 16:41.19 | jbot | i guess echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, 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, or of ... |
| 16:41.19 | Dovid | how do i force them to use ulaw ? |
| 16:41.29 | Strom_C | Dovid, you find your telco's milliwatt test and use that to adjust rxgain and txgain |
| 16:41.41 | Dovid | i am still learning |
| 16:41.47 | Dovid | whats a milliwatt test ? |
| 16:41.49 | UForgotten | Strom_c: I wish it were that easy - you have to practically TORTURE a lineman for them to give you any info |
| 16:41.59 | Strom_C | UForgotten, thats why you dont ask the linemen |
| 16:42.12 | Strom_C | you use the AT&T test line directory :) |
| 16:42.14 | *** join/#asterisk lordbaron2 (n=redbaron@host55-226.rancor.birch.net) |
| 16:42.27 | Dovid | Storm_C: how do i do that ? |
| 16:42.36 | Strom_C | Dovid, 1004hz tone at 0dbm |
| 16:43.02 | Strom_C | Dovid, who is your telephone company? |
| 16:43.02 | lordbaron2 | storm_c: Sorry--got bumped. I was wrong. answeronpolarityswitch is the proper |
| 16:43.04 | Dovid | Strom_C: Chineese to me. i am linux guy. not telco. still leanring |
| 16:43.06 | UForgotten | google has been unhelpful on that |
| 16:43.07 | Dovid | Verizon |
| 16:43.15 | UForgotten | I'm on BellSouth ac 352 |
| 16:43.50 | Strom_C | UForgotten, i dont know where bellsouth maps their test numbers |
| 16:44.04 | Strom_C | but search for their CLEC website |
| 16:44.23 | UForgotten | I've searched their clec website, you have to have a login/password for the good stuff |
| 16:44.44 | Dovid | Strom_C: how do i find out verizon's ? |
| 16:44.46 | Strom_C | UForgotten, so wait till AT&T finishes the acquisition :) |
| 16:44.52 | Strom_C | Dovid, which state? |
| 16:45.04 | Dovid | NJ |
| 16:45.11 | Strom_C | bell atlantic...hmmmm |
| 16:45.20 | UForgotten | ok, I'll keep futzing around. ttyl |
| 16:45.39 | Dovid | when i googled verizon milliwat i got thsi |
| 16:45.39 | Dovid | http://www.pticom.com/tariffs/fcc-4.pdf#search=%22verizon%20milliwatt%22 |
| 16:45.56 | Strom_C | Dovid, the key is finding the milliwatt test on your local switch |
| 16:46.01 | Strom_C | it can't come from anywhere but your local switch |
| 16:46.13 | Strom_C | trunks may incur additional transmission loss |
| 16:46.25 | Dovid | local switch meaning the verizon switch that my home connects to ? |
| 16:46.31 | Strom_C | yes |
| 16:46.40 | Dovid | ok so how do i find that out ? i need to call verizon ? |
| 16:46.47 | Strom_C | yes |
| 16:46.52 | Dovid | or is there an app that i can run ? |
| 16:46.57 | Strom_C | ...an app? |
| 16:47.06 | Dovid | well i dont even know what it is |
| 16:47.19 | Strom_C | you lost me |
| 16:47.20 | Dovid | if i play with the tx and rx gain i should get it working ? |
| 16:47.22 | Dovid | lol |
| 16:47.23 | Dovid | ok |
| 16:47.25 | Dovid | if i play with the tx and rx gain i should get it working ? |
| 16:47.25 | lordbaron2 | strom_c: I put those options in place, but still does not detect disconect. Should I be using callprogress? |
| 16:47.35 | blitzrage | Dovid: if it works, then yes, if not, then no |
| 16:47.41 | Strom_C | lordbaron2, wait till tomorrow and contact digium support |
| 16:47.44 | Dovid | ok |
| 16:47.55 | Dovid | tx gain is for ??? and rx gain is for ?? |
| 16:47.59 | lordbaron2 | k,thx |
| 16:48.06 | Dovid | rx is in and tx is out ? |
| 16:48.09 | *** part/#asterisk lordbaron2 (n=redbaron@host55-226.rancor.birch.net) |
| 16:48.11 | Strom_C | yes |
| 16:48.37 | Dovid | right now i have |
| 16:48.40 | Dovid | rxgain=0.0 |
| 16:48.40 | Dovid | txgain=-2.0 |
| 16:48.50 | Strom_C | why do you have that? |
| 16:48.52 | Dovid | what do u think i need to change ? |
| 16:49.03 | Dovid | just coppied my last configs that a friend did and it worked great |
| 16:49.17 | Strom_C | how many feet long is your copper loop? |
| 16:49.20 | Dovid | till the faxing... |
| 16:49.29 | Dovid | from me to the telco ? no clue |
| 16:50.04 | Dovid | also relaxed dtmf is on should it be off ? |
| 16:50.08 | Strom_C | Dovid, find out what your local milliwatt test number is, and get back to me |
| 16:50.23 | Strom_C | for faxing, you bet your ass it should be off |
| 16:50.34 | Dovid | that means calling verizon which never picks up so i will never get backt o u :( |
| 16:50.49 | Strom_C | that just means you're not persistent enough |
| 16:50.51 | Dovid | oops |
| 16:50.52 | Strom_C | call repair service |
| 16:50.57 | Dovid | relaxeddtmf=yes |
| 16:51.00 | Dovid | it should be no correct ? |
| 16:51.08 | Strom_C | that's what I just said |
| 16:51.40 | Dovid | brb |
| 16:52.14 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:56.46 | zoa | :) |
| 17:05.28 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
| 17:07.05 | rg1_ | in a dialplan - is there a "defined" variable I can use to print the current date/time? |
| 17:07.17 | rg1_ | print/use |
| 17:07.29 | rg1_ | like ${CURRENT_TIME} ? |
| 17:07.36 | rg1_ | that would reflect the system date and time? |
| 17:09.05 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 17:09.17 | rg1_ | test - anyone see me here? |
| 17:09.24 | Strom_C | yes |
| 17:09.32 | Strom_C | try ${EPOCH} |
| 17:09.41 | rg1_ | ah |
| 17:09.42 | rg1_ | ok, thx |
| 17:12.18 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 17:18.59 | rg1_ | Strom_C-ok, i got epoch, but in form: |
| 17:19.05 | rg1_ | [1156699014] |
| 17:19.36 | Strom_C | yes |
| 17:19.42 | rg1_ | is there any (easy) way to get that formatted into something human readable? (i.e. 08 Aug 2006 12:20) |
| 17:19.43 | rg1_ | ? |
| 17:19.52 | rg1_ | within the dialplan? |
| 17:19.54 | Strom_C | the number of seconds that have elapsed since midnight, st january 1970 |
| 17:20.03 | Strom_C | s/st/1st/ |
| 17:20.12 | rg1_ | right |
| 17:20.21 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 17:20.28 | Strom_C | perhaps ${DATETIME} but I think that has been deprecated |
| 17:20.30 | rg1_ | but is there a function or something to make that a tad easier |
| 17:20.35 | rg1_ | let me try that |
| 17:20.38 | EmleyMoor | fwd back :-) |
| 17:20.43 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 17:20.52 | russellb | in trunk, there is STRFTIME ... |
| 17:21.29 | russellb | file: ! |
| 17:21.42 | novafirst | how can I asign language setting per sip user? |
| 17:21.44 | FTexcom | how can I make that if I press a number I can pickup a ringing extension? |
| 17:21.59 | Strom_C | FTexcom, pickup groups |
| 17:22.01 | florz | Strom_C: Actually, it's the number of days that have elapsed since then plus the number of seconds since last midnight in UTC - and even that isn't completely exact yet ... =:-) |
| 17:22.06 | florz | err |
| 17:22.14 | florz | Strom_C: Actually, it's the number of days that have elapsed since then times 86400 plus the number of seconds since last midnight in UTC - and even that isn't completely exact yet ... =:-) |
| 17:22.22 | file | russellb: ? |
| 17:22.31 | russellb | file: i was blaming you for something |
| 17:22.32 | Strom_C | close enough |
| 17:22.41 | file | oic |
| 17:23.06 | florz | Strom_C: Well, depends on the application - but the UTC part is important to mention ... |
| 17:23.47 | Strom_C | florz, I was giving the short short version |
| 17:24.26 | EmleyMoor | However, I can't receive calls on it... |
| 17:24.36 | EmleyMoor | I get this when a call comes in: |
| 17:24.55 | EmleyMoor | Aug 27 18:23:09 WARNING[32422]: channel.c:506 ast_best_codec: Don't know any of 0xf800 formats |
| 17:25.07 | EmleyMoor | Aug 27 18:23:09 ERROR[32422]: chan_iax2.c:7383 socket_read: No best format in 0xf800?? |
| 17:25.25 | EmleyMoor | Aug 27 18:23:09 NOTICE[32422]: chan_iax2.c:7388 socket_read: Rejected connect attempt from 192.246.69.186, requested/capability 0x4/0xf804 incompatible with our capability 0xff03. |
| 17:25.38 | EmleyMoor | Why do I get this, and what do I have to do to fix it? |
| 17:29.52 | EmleyMoor | I can't proceed any further until I can receive calls :-( |
| 17:29.56 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 17:30.39 | Strom_C | EmleyMoor, show me your iax.conf |
| 17:30.46 | Strom_C | ~pb |
| 17:30.47 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 17:31.04 | EmleyMoor | OK - will pastebin shortly |
| 17:32.49 | rg1_ | exten => s,n,NoOp([DateTime()]-ALMSG-2006-08-27 |
| 17:33.18 | rg1_ | I was hoping above would print the date/time in the log - instead it just printed "[DateTime()].... |
| 17:33.27 | Strom_C | thats because you didnt listen to me, rg1_ |
| 17:33.30 | rg1_ | any idea how i get that to evalulate |
| 17:33.33 | Strom_C | i said ${DATETIME} |
| 17:33.43 | rg1_ | ah |
| 17:33.53 | rg1_ | one more try..... |
| 17:34.20 | EmleyMoor | http://pastebin.com/777319 |
| 17:34.58 | FTexcom | EmleyMoor the extensions.conf would be useful too |
| 17:35.09 | EmleyMoor | OK - will do that too... |
| 17:35.27 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:35.35 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:35.58 | Strom_C | EmleyMoor, add allow=ulaw to your user entry for iax |
| 17:36.04 | Strom_C | s/iax/fwd |
| 17:36.19 | *** join/#asterisk svenna_ (n=svenna@p548D3D23.dip0.t-ipconnect.de) |
| 17:36.25 | EmleyMoor | Strom_C: My entry? Which entry? |
| 17:36.38 | Strom_C | your FWD user entry in iax.conf |
| 17:36.42 | rg1_ | Strom - that evaluated to a blank |
| 17:36.44 | EmleyMoor | user - ah |
| 17:37.04 | Strom_C | rg1_, show me your noop statement |
| 17:37.05 | EmleyMoor | ttp://pastebin.com/777323 is my extensions.conf |
| 17:37.11 | EmleyMoor | http... even |
| 17:38.44 | EmleyMoor | Sorted - thanks Strom_C |
| 17:38.49 | Strom_C | sorted? |
| 17:39.02 | rg1_ | <PROTECTED> |
| 17:39.19 | Strom_C | rg1_, perhaps try ${TIMESTAMP} |
| 17:39.21 | rg1_ | log showed: -ALMSG-2006-08-27-1214a-tqm_start) |
| 17:39.25 | EmleyMoor | Yes, Strom_C: I can now receive calls... |
| 17:39.27 | rg1_ | let me try that |
| 17:39.36 | Strom_C | EmleyMoor, good |
| 17:39.46 | Strom_C | EmleyMoor, you have bandwidth=low in your general section |
| 17:39.52 | EmleyMoor | Anyone want to give me a try on FWD 794933? |
| 17:39.58 | EmleyMoor | Is that a bad thing? |
| 17:40.00 | Strom_C | and bandwidth=low does not include ulaw |
| 17:40.06 | EmleyMoor | Ah! |
| 17:41.52 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 17:43.19 | EmleyMoor | Now I've solved that, I can read a bit more of the book and try and set up some other cool stuff |
| 17:43.32 | EmleyMoor | Calls to my FWD number are welcome, within reason |
| 17:46.35 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:52.18 | rg1_ | exten => s,n,NoOp([ALMSG-${STRFTIME(${EPOCH},,%m/%d/%Y-%H:%M:%S)}]- |
| 17:52.21 | rg1_ | that worked! |
| 17:52.43 | russellb | rg1_: nice :) |
| 17:52.57 | rg1_ | thanks strom and all. |
| 17:58.04 | novafirst | <PROTECTED> |
| 17:58.18 | EmleyMoor | Sound generated by my asterisk sounds awful - is it possible that that is because it's under-spec hardware? |
| 17:58.30 | zoa | normally no |
| 17:59.41 | FTexcom | EmleyMoor using zap? |
| 17:59.56 | EmleyMoor | FTexcom: No - entirely iax |
| 18:01.07 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 18:01.08 | FTexcom | novafirst add the line language=whatever inside the extension conf file |
| 18:02.08 | FTexcom | EmleyMoor strange, perhaps using wrong codec? |
| 18:02.39 | EmleyMoor | The sound is badly broken up, as if the machine is too slow (which, I suppose, it is) |
| 18:03.07 | EmleyMoor | I will be buying new hardware before making this a fully deployable solution but am hoping to confirm this first |
| 18:03.14 | *** join/#asterisk dieno2 (n=dienno2@124.29.194.150) |
| 18:04.39 | dieno2 | can n e 1 tell me how can i solve this prb |
| 18:04.42 | dieno2 | http://pastebin.com/777342 |
| 18:04.48 | dieno2 | plz |
| 18:04.54 | EmleyMoor | Local echo test sounds similar - remote echo test via FWD sounds OK |
| 18:05.39 | novafirst | FTexcom: but I don't want to set global language settings, instead only per user |
| 18:05.42 | dieno2 | iz this for me :P |
| 18:06.34 | FTexcom | novafirst inside the [extension]... |
| 18:06.45 | dieno2 | how can i solve it |
| 18:07.24 | novafirst | did anyone applied asterisk-1.2.11-patch ? |
| 18:07.29 | zoa | check with a sniffer to see if its normal there |
| 18:07.31 | zoa | maybe its your phone |
| 18:17.03 | EmleyMoor | Can someone please call me on FWD 794933 or offer to take a call from me, so that we can test how my setup works on a real call? |
| 18:18.49 | Dovid | for an FXS port on asteris |
| 18:18.51 | Dovid | asterisk* |
| 18:19.04 | Dovid | to transfer calls , dnd etc. where are these options set ? |
| 18:19.04 | *** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net) |
| 18:20.00 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 18:31.28 | dieno2 | can ne tell me best Quality of Cedec Please |
| 18:32.43 | *** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com) |
| 18:32.47 | harlequin516 | How can I get asterisk to output to the logs the same way that `asterisk -c -vvvvvvvvv -dddddddddd` does? |
| 18:33.15 | harlequin516 | I have this fgeeling there is no standard way to do this. |
| 18:35.18 | harlequin516 | Anyone here? |
| 18:35.46 | websae | nope |
| 18:36.13 | harlequin516 | There's like a billion people here, and no one is talking. |
| 18:38.55 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:40.45 | *** join/#asterisk Flauto (n=HP_Owner@adsl-75-3-138-135.dsl.chcgil.sbcglobal.net) |
| 18:41.29 | Flauto | hey |
| 18:41.54 | Flauto | what is the package called in debian for php-cli |
| 18:48.43 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.101) |
| 19:01.21 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-54.intrastar.net) |
| 19:02.07 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 19:03.07 | *** join/#asterisk GlobbeTotssz (i=GlobbeTo@200.122.149.57) |
| 19:03.20 | harlequin516 | Sorry, gentoo here |
| 19:07.52 | docelmo | Flauto its for running php from the CLI or Command Line Interface |
| 19:08.07 | docelmo | its not however a module for Apache etc.. |
| 19:08.16 | docelmo | Im assuming your new as linux |
| 19:09.41 | b4ka | docelmo: he asked how was the package called... |
| 19:10.21 | docelmo | ohh misunderstod |
| 19:10.23 | docelmo | tood |
| 19:10.25 | docelmo | my bad.. |
| 19:10.36 | docelmo | do a apt-get install php* |
| 19:10.51 | docelmo | that should install all php associated packages |
| 19:11.03 | b4ka | why would he do that? |
| 19:11.36 | zoa | hey ho docelmo |
| 19:20.52 | zoa | iiiits ooooh so quiet |
| 19:20.57 | zoa | iiiits oooooh soooo stilll |
| 19:22.09 | docelmo | yep |
| 19:22.27 | docelmo | simple way to find the cli package for php |
| 19:22.48 | docelmo | Why do poeple use vonage for their phone? People just do dumb things.. |
| 19:26.55 | Qwell | docelmo: That's their slogan, right? |
| 19:27.15 | Qwell | "People do stupid things." or whatever |
| 19:27.33 | docelmo | yep |
| 19:27.38 | *** join/#asterisk Fender22211 (n=fender21@cpe-70-125-138-128.satx.res.rr.com) |
| 19:28.04 | Fender22211 | anyone here have any experience with VOICEMAIL OBDC Storage? |
| 19:28.45 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:28.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:28.50 | Qwell | mog: !!! |
| 19:29.18 | mog | Qwell, ! |
| 19:29.38 | Deeewayne | ? |
| 19:29.50 | file | Qwell: do you want my black Digium shirt? it's obviously a medium despite what the label says |
| 19:30.10 | Qwell | file: yeah, they run big. the small is almost too big, heh |
| 19:30.25 | Qwell | or...hmm, that wouldn't make sense, would it? |
| 19:30.35 | Qwell | mine ran big, yours ran small? :P |
| 19:30.37 | mog | Deeewayne, !! |
| 19:30.38 | file | no, but anyway - my orange one is fine, but this black one... ungood |
| 19:30.43 | Qwell | weird |
| 19:30.47 | Qwell | umm, sure |
| 19:30.57 | Deeewayne | mog!! file!! qwell!! |
| 19:31.00 | Qwell | Deeewayne: ! |
| 19:31.49 | file | Deeewayne: !!! |
| 19:31.56 | Fender22211 | other than modifying the voicemail.c source, is there anyway to add the Caller ID Phone number only to the MSG***.txt file? |
| 19:32.12 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 19:32.14 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 19:36.53 | Qwell | Fender22211: It stores the callerid already, doesn't it? |
| 19:37.39 | harlequin516 | Can you specify an incoming context and an outgoing context in the iax.conf or sip.conf? |
| 19:38.19 | Strom_C | you dont need to specify an outgoing context |
| 19:38.38 | Strom_C | you call that entry using the dial() app |
| 19:38.46 | Qwell | You specify an incoming context for your provider, then an outgoing context on your device |
| 19:39.16 | Strom_C | hi, btw :)_ |
| 19:39.17 | Strom_C | er :) |
| 19:39.22 | Qwell | ~hi |
| 19:39.23 | jbot | hello, qwell |
| 19:39.33 | Strom_C | ~hi bob |
| 19:39.34 | jbot | Many greetings, bob, most strange traveller, to this IRCdom of plenty. |
| 19:39.39 | Strom_C | hah! thats how it works |
| 19:39.43 | Qwell | ~hi randomperson |
| 19:39.44 | jbot | Many greetings, randomperson, most strange traveller, to this IRCdom of plenty. |
| 19:39.47 | *** join/#asterisk |ryan| (n=foo@c-24-23-17-75.hsd1.ca.comcast.net) |
| 19:40.19 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 19:40.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-96-211.red.bezeqint.net) |
| 19:40.51 | Strom_C | now that I'm rested, it feels amazing to be back home :) |
| 19:40.51 | file | Strom_C: Der Waffle Haus! |
| 19:41.06 | Strom_C | no tengo Der Waffle Haus in Los Angeles, unfortunately |
| 19:41.12 | Qwell | Strom_C: start one |
| 19:41.21 | Strom_C | Qwell, I looked into it |
| 19:41.24 | Qwell | ha |
| 19:41.33 | Strom_C | unfortunately it's incredibly difficult to become a Waffle House franchisee |
| 19:41.41 | file | you have to already be an empployee |
| 19:41.43 | file | er employee |
| 19:41.48 | Qwell | wtf |
| 19:41.59 | Qwell | You have to work there, before you can start one? |
| 19:42.08 | harlequin516 | hmm... I'm confused is iaxtel and freeworld dialup two different things? |
| 19:42.13 | Qwell | harlequin516: yes |
| 19:42.15 | file | harlequin516: yes |
| 19:42.22 | Strom_C | harlequin516: yes |
| 19:42.25 | Qwell | harlequin516: yes |
| 19:42.33 | harlequin516 | Okay I get itr! ;) |
| 19:42.35 | Strom_C | harlequin516: yes |
| 19:43.09 | *** join/#asterisk FullService (n=jiggaman@CPE0002724fc55e-CM000f211fd29c.cpe.net.cable.rogers.com) |
| 19:49.19 | Fender22211 | anyone familiar with editing voicemail.c? |
| 19:53.49 | *** join/#asterisk jphansen (n=jphansen@cpe.atm4-0-1301192.0x3ef3c23a.hgnxx3.customer.tele.dk) |
| 19:54.30 | *** join/#asterisk darby_t (n=tom@aapr208.neoplus.adsl.tpnet.pl) |
| 19:58.45 | pyrom | harlequin516, yes |
| 19:59.31 | Strom_C | hahha |
| 19:59.37 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:03.16 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 20:03.20 | Strom_C | I'm always horrified when people describe the mechanical ringer inside an older telephone set as having a "ring tone" |
| 20:04.49 | trelane | Strom_C, but it does... and it has to be tuned by adjusting the bell |
| 20:05.05 | benjk_ | a discrete ringtone |
| 20:05.14 | benjk_ | in the sense of discrete hardware |
| 20:05.53 | Strom_C | "tone", at least in the context of telephony, is more suggestive of something generated by an electronic device than a noise generated by striking a bell |
| 20:05.55 | trelane | right |
| 20:06.03 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:06.13 | benjk_ | that depends on your age |
| 20:06.20 | trelane | tone equates to perceived frequency |
| 20:06.22 | trelane | ie how it sounds |
| 20:06.38 | trelane | so the bell on the old phones did in fact have a perceived ringing frequency, or a ring tone |
| 20:06.51 | trelane | in fact some of them rang out of tune and had to be acoustically tuned before shipping |
| 20:07.27 | Strom_C | and the sound of a mecanical bell is more than just tones...there's the noise of the hammer actually striking the bell and the noise of the head of the hammer rattling against the stiff wire it's mounted on |
| 20:07.46 | benjk_ | its not |
| 20:08.10 | Strom_C | s/mechanical bell/mechanical bell ringer/ |
| 20:08.13 | benjk_ | its the entire perception of whatever noise comes out of the phone when it rings |
| 20:08.27 | benjk_ | thats the ringing tone |
| 20:09.13 | benjk_ | you can have the very same ringing tone including the hammer artifacts as a digitally sampled ringing tone on a mobile phone |
| 20:09.37 | Strom_C | well, sure, but the tiny speaker on the mobile phone can't really do the original sound justice |
| 20:09.38 | trelane | benjk_, I have such a ring tone on my cell phone |
| 20:09.43 | trelane | Strom_C, concur |
| 20:09.53 | benjk_ | just because one is created mechanically and the other not, doesn't make them a tone in one case and not a tone in the other |
| 20:11.12 | Strom_C | by your definition, you could also call the sound of a Boeing 747 a tone |
| 20:11.34 | h3x | Strom_C: im sure the wireless companies would sell it for $0.49 if people would buy it |
| 20:11.53 | Qwell | h3x: $0.49? You must not live in the US |
| 20:11.58 | h3x | ok well |
| 20:12.01 | h3x | $0.49 per week |
| 20:12.03 | Strom_C | try $3? |
| 20:12.04 | h3x | until you send CANCEL to 51398753195731985713 |
| 20:12.06 | benjk_ | if you use the sampled noise of a 747 on your phone as a ringing tone, then yes of course |
| 20:12.43 | trelane | Strom_C, the sound of a 747's engine is a tone, and if it's sympathetic to any other resonant frequency on the aircraft it will shake itself apart |
| 20:12.46 | benjk_ | some guy I know uses a flushing toilet sound as a ringing tone for disctinctive ring |
| 20:12.52 | h3x | haha |
| 20:12.53 | Strom_C | benjk_, hahahaha |
| 20:12.56 | Strom_C | that's awesome |
| 20:13.07 | h3x | i bet he dosent answer the phone when in a public bathroom |
| 20:13.20 | h3x | or maybe i should say |
| 20:13.28 | Qwell | tone: a sound of definite pitch; a note. |
| 20:13.29 | h3x | tries to answer the phone when he leans forward to find out there was no call |
| 20:13.33 | Qwell | 747 != note |
| 20:13.40 | h3x | 747 == noise |
| 20:13.44 | Strom_C | mechanical noise != note |
| 20:14.08 | Qwell | 440/480 == tone |
| 20:14.10 | benjk_ | a ringing tone is whatever sound/noise you use to let your phone ring |
| 20:14.29 | benjk_ | you are confusing indication tones with ringing tones |
| 20:14.37 | Qwell | No, the word "tone" |
| 20:14.43 | benjk_ | nobody says ringing noise |
| 20:14.50 | Qwell | well they should ;) |
| 20:15.33 | benjk_ | you can just as well call an indication tone an indication noise |
| 20:15.47 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:15.48 | Qwell | yes, and both would be correct |
| 20:16.09 | Qwell | but calling some guy screaming "answer your phone!" a "tone" is so horribly wrong |
| 20:16.22 | Strom_C | but, just as all squares are rectangles while all rectangles are not squares... |
| 20:16.36 | Qwell | indeed |
| 20:17.10 | benjk_ | ringing tone has become a distinct kind of its own |
| 20:17.49 | benjk_ | independent of the uses otherwise usual for the word tone on its own |
| 20:18.56 | *** join/#asterisk vt (n=vt@MTL-ppp-149614.qc.sympatico.ca) |
| 20:19.35 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 20:22.20 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 20:22.25 | benjk_ | in Japan, ringing tones are called chaku melo |
| 20:22.35 | hmmhesays | heh |
| 20:22.37 | benjk_ | which means arrival melody |
| 20:22.44 | Qwell | yes, and melody makes more sense |
| 20:22.52 | benjk_ | now, you could argue that not all are melodies |
| 20:23.05 | Qwell | a melody is a composition of tones. "ringtones" used to be polyphonic |
| 20:23.05 | benjk_ | but natural language doesn't work this way |
| 20:23.24 | Qwell | So, the term needs to change ;) |
| 20:23.28 | hmmhesays | well the mp3's that get played these days could be considered melody |
| 20:23.44 | Qwell | midis are melodies |
| 20:23.52 | benjk_ | a certain term becomes popular and eventually it gets a meaning on its own, independent of its components |
| 20:23.54 | Strom_C | "arrival melody" somehow makes me think of a Sirius Cybernetics-esque musical jingle played when an automatic taxicab reaches its destination |
| 20:24.12 | hmmhesays | from robocop? |
| 20:24.12 | Qwell | Strom_C: what movie was that? |
| 20:24.16 | hmmhesays | no |
| 20:24.17 | hmmhesays | 6th day |
| 20:24.19 | Strom_C | it wasnt a movie |
| 20:24.33 | Strom_C | it was just something that my mind conjured up |
| 20:24.39 | Strom_C | i do have an independent imagination, you know |
| 20:24.48 | hmmhesays | sure |
| 20:24.55 | hmmhesays | all your imagination are belong to us |
| 20:24.55 | Qwell | You lose |
| 20:24.56 | Qwell | The Sirius Cybernetics Corporation is a fictional company from Douglas Adams' The Hitchhiker's Guide to the Galaxy. |
| 20:25.11 | Strom_C | I know that |
| 20:25.20 | Strom_C | it's known for producing horrid products |
| 20:25.22 | Qwell | :p |
| 20:25.26 | benjk_ | it would be nice if the asterisk code base would be equally scrutinised for misnomers and ambiguous or plain wrong uses of language |
| 20:26.55 | hmmhesays | what do you call those christmas trees you use in car doors |
| 20:26.57 | benjk_ | coders with lazy fingers find it justified to use something thats cryptic, non-obvious, ambiguous or otherwise confusing because they say it is easier to type |
| 20:27.03 | Qwell | hmmhesays: wreaths? |
| 20:27.17 | hmmhesays | no the plastic plugs used to put door panels on |
| 20:27.35 | benjk_ | but at the same time, ordinary folks shouldn't have the same right to convenience taking short cuts in every day language? |
| 20:27.47 | Qwell | benjk_: give me an example |
| 20:28.24 | benjk_ | for example the variable transfer |
| 20:28.42 | benjk_ | I think its in channel.c |
| 20:28.55 | *** join/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 20:29.07 | benjk_ | it used to be once only for the purpose of transfer flags |
| 20:29.25 | benjk_ | but it has accumulated so many more flags that its become a totally confusing misnomer |
| 20:29.36 | Qwell | I'm not seeing any such thing in channel.c |
| 20:30.21 | benjk_ | maybe its in pbx.c or dial, can't remember now where it was, but if you read any code, there is plenty of this stuff |
| 20:30.21 | hmmhesays | what a sad state of affairs cable tv is in |
| 20:30.39 | mog | and they do if you read the code |
| 20:31.10 | mog | i mean asterisk is far more readable than any other similliarly sized project i have had the pleasure of reading |
| 20:31.11 | mountainm2k | Little help building zaptel on 2.6 kernel... |
| 20:31.31 | hmmhesays | i need a decent dhcp server |
| 20:31.34 | hmmhesays | and a good guide |
| 20:31.42 | mountainm2k | hmmhesays: dhcpd |
| 20:31.50 | hmmhesays | oh yeah? |
| 20:31.51 | benjk_ | non-obvious and ambiguous or misappropriated terms in daily life language also makes sense if you know what it means or if you listen in long enough to learn from context |
| 20:32.27 | mountainm2k | hmmhesays: I actually use Windows 2003 DHCP server, but not because I want to...... |
| 20:32.35 | benjk_ | the point is that readability and comprehensibility suffers if such terms are used |
| 20:32.57 | mountainm2k | All: trying to diag problem building zaptel on 2.6 kernel... It won't build, errors out with a bunch of crap... |
| 20:33.01 | mog | like i said i find asterisk to be quite readable |
| 20:33.19 | mog | as do the other core developers |
| 20:33.26 | Strom_C | mountainm2k, pastebin the error |
| 20:33.28 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 20:33.30 | benjk_ | that's why you wouldn't use ambiguous terms in natural language if the audience is wide enough, for example a report, a newspaper article, an open letter, a news item |
| 20:33.36 | mountainm2k | http://pastebin.ca/151212 |
| 20:34.08 | Strom_C | mountainm2k, you're on centos / RHEL, arent you |
| 20:34.09 | EmleyMoor | Is there an example of how to connect to fwd using SIP? I'm getting unexplained busies using iax, but not to all numbers |
| 20:34.09 | mountainm2k | Strom_C: To make matters worse, it's ABE-B -- they told me I needed to have RHEL4 (which is 2.6 kernel)... |
| 20:34.17 | mountainm2k | Strom_C: Yes |
| 20:34.19 | Strom_C | mountainm2k, see the following: |
| 20:34.22 | Strom_C | ~centosbug |
| 20:34.26 | jbot | it has been said that centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 20:35.02 | mountainm2k | sed: can't read /usr/src/kernels/2.6.9-34.0.2.ELsmp-i686/include/linux/spinlock.h: No such file or directory |
| 20:35.11 | hmmhesays | that's the the problem with rhel4 |
| 20:37.00 | benjk_ | it has absolutely no relevance if you know what it means or anyone else who reads it on a daily basis, readability and comprehensibility is defined by whether or not something is readable/comprehensible without any prior knowledge of the "text" |
| 20:37.00 | hmmhesays | nm |
| 20:37.00 | mountainm2k | (latest-and-greatest kernel from yum) |
| 20:37.51 | mountainm2k | ahha... jbot has a bad bit in the macro there... |
| 20:37.57 | mountainm2k | I see how to fix |
| 20:38.46 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-59.dslextreme.com) |
| 20:39.02 | benjk_ | in any event, the point is that if you take the liberty for yourself to use shorthand in your own domain, then you have to extend the right to others to do the same in their domain |
| 20:39.14 | mountainm2k | It looks like that fixed it, thank you... |
| 20:39.45 | mountainm2k | FYI the issue w/ jbot's command is that the path including the unames doesn't work correctly for SMP kernels... |
| 20:39.54 | mountainm2k | but I got it figured, and it built now, TY... |
| 20:39.57 | Strom_C | mountainm2k: so fix it |
| 20:40.04 | Strom_C | :) |
| 20:40.07 | mountainm2k | hahah |
| 20:40.12 | mountainm2k | Hmmm... |
| 20:40.15 | mountainm2k | ~help |
| 20:40.22 | mountainm2k | nice... :-) |
| 20:40.44 | Strom_C | you can tell it things like so: |
| 20:40.54 | Strom_C | jbot, giggityflorp is total nonsense |
| 20:40.56 | jbot | Strom_C: okay |
| 20:41.02 | Strom_C | ~giggityflorp |
| 20:41.03 | jbot | somebody said giggityflorp was total nonsense |
| 20:41.13 | Qwell | jbot: no, giggityflorp is something else |
| 20:41.15 | jbot | okay, Qwell |
| 20:41.15 | Strom_C | jbot, forget giggityflorp |
| 20:41.41 | benjk_ | so if the general public likes to "misappropriate" the term tone for something you would rather call melody or sound sample because they feel tone is shorter and more convenient, then you should tolerate it as much as you expect others to tolerate your own shortcuts in the asterisk code base |
| 20:42.09 | Strom_C | benjk_: I'm not a coder, so your demand has no relevance to me :) |
| 20:42.33 | mountainm2k | Hmmm, this is probably better: |
| 20:42.34 | mountainm2k | jbot, centosbug is centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 20:42.36 | jbot | I think you lost me on that one, mountainm2k |
| 20:42.37 | benjk_ | I am sure that you are using similar shortcuts in whatever your own domain of expertise is |
| 20:42.45 | mountainm2k | d'oh |
| 20:42.54 | mountainm2k | jbot, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 20:42.56 | jbot | ...but centosbug is already something else... |
| 20:42.57 | benjk_ | at least I would be surprised if you didn't |
| 20:43.22 | Strom_C | jbot, no, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 20:43.24 | jbot | Strom_C: okay |
| 20:43.40 | doolph | how can I dial an extension with agi |
| 20:43.41 | benjk_ | in general I agree with you that proper use of language is important |
| 20:43.46 | mountainm2k | nice |
| 20:43.55 | mountainm2k | jbot, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 20:43.57 | jbot | i already had it that way, mountainm2k |
| 20:44.06 | Strom_C | mountainm2k: I already set it |
| 20:44.08 | mountainm2k | ~centosbug |
| 20:44.09 | jbot | it has been said that centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 20:44.11 | mountainm2k | o i c :-) |
| 20:44.14 | benjk_ | yet, natural language is defined by its use, and use can change over time |
| 20:45.30 | benjk_ | English is the outcome of a Saxon population trying to learn French and they didn't quite get it right |
| 20:45.48 | Strom_C | English is more than that |
| 20:45.53 | mountainm2k | Thanks to hmmhesays and Strom_C for that |
| 20:45.55 | mountainm2k | :-P |
| 20:45.55 | Strom_C | English is the Frankenstein's Monster of language |
| 20:46.00 | benjk_ | French itself is the outcome of a Frankish population to speak Latin and they didn't quite get it right |
| 20:46.46 | hmmhesays | ahaha |
| 20:46.50 | Strom_C | benjk_ is the outcome of someone reacting overly pedantically to what was essentially a JOKE |
| 20:47.05 | hmmhesays | is there any programming languages that aren't english based? |
| 20:47.07 | E-bola | English is easy |
| 20:47.13 | hmmhesays | besides something like assembly |
| 20:47.14 | E-bola | 10x easier to learn than french |
| 20:47.17 | EmleyMoor | Can call my own fwd number over iax and it rings - can't phone a friend |
| 20:47.37 | E-bola | hmmhesays: pascal maybe? |
| 20:47.40 | benjk_ | there have been several localised versions of Pascal for teaching purposes |
| 20:47.51 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 20:47.57 | EmleyMoor | hmmhesays: Like BASIC but in French perhaps? |
| 20:48.00 | *** join/#asterisk dpryo (n=hn@raphael.ondskap.net) |
| 20:48.22 | benjk_ | Pascal itself is all English, but some compilers where created with translations of the key words into local languages for teaching |
| 20:48.45 | h3x | quebec spent a ton of money making a C language in french |
| 20:49.09 | h3x | which is dumb because it could be done with a bunch of #define's |
| 20:49.27 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 20:49.28 | Strom_C | well, that's quebec for you |
| 20:49.33 | benjk_ | its very easy to do if you have the source code for gcc |
| 20:49.42 | benjk_ | just recalculate the hashes |
| 20:49.50 | benjk_ | change 'em and recompile |
| 20:50.04 | h3x | i suppose the error messages and everything were translated |
| 20:50.35 | benjk_ | oh well, that's a few hundred or a few thousand dollars worth of farming out to a translation agency |
| 20:55.01 | benjk_ | also there is APL, which is not English but pure mathematical notation |
| 20:56.46 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:57.07 | mog | benjk_, you realize this is a total fruitless argument rihgt? |
| 20:57.14 | benjk_ | argument? |
| 20:57.17 | h3x | my personal favorite language is Brainfuck |
| 20:57.22 | dhanes | i was learning |
| 20:57.28 | benjk_ | somebody asked if there was a language not based on English |
| 20:57.31 | mog | that asterisk and other code should just be in english |
| 20:57.40 | dhanes | had never heard of APL |
| 20:57.56 | benjk_ | mog you should read more carefully |
| 20:58.05 | mog | sorry i jumped back in |
| 20:58.07 | benjk_ | I responded to somebody asking a questio |
| 20:58.14 | mog | but usually you are running on one track |
| 20:58.15 | mog | my bad |
| 20:58.33 | h3x | i think we should respond to mog in french from now on |
| 20:58.39 | h3x | haha |
| 20:58.43 | mog | lol |
| 20:58.44 | mog | okies |
| 20:58.57 | mog | im always up for learning something new |
| 20:59.06 | dhanes | mog tres penible |
| 20:59.14 | dhanes | mog est tres penible :) |
| 20:59.29 | dhanes | j/k |
| 20:59.37 | mog | lol |
| 20:59.45 | mog | viva la babelfish |
| 21:00.03 | dhanes | i've pissed off in-laws using that damn thing |
| 21:00.13 | dhanes | type something in, have them look at it |
| 21:00.25 | dhanes | and get a strange look from them |
| 21:00.28 | h3x | it says "go screw yourself with a fork" ? |
| 21:00.43 | dhanes | from 'I think your dinner was wonderful!' |
| 21:00.51 | h3x | yep |
| 21:01.14 | hmmhesays | 'your daughters vagina tastes like rotten eggs' |
| 21:01.17 | dhanes | ROFL |
| 21:01.19 | h3x | hahahah |
| 21:01.21 | mog | babel fish only works if you know how to use it |
| 21:01.35 | mog | have to speak in broken english or whatever language |
| 21:01.40 | mog | so it works more like a dictionary |
| 21:01.43 | dhanes | i just rely on the wife, speaks cantonese, castillian spanish, portuguese |
| 21:01.50 | mog | than a real language translator |
| 21:02.07 | dhanes | mon chochon est grand! |
| 21:03.28 | JunK-Y | dhanes: le mien sent la crevette! |
| 21:03.58 | Qwell | oh no, it's a real Quebecian :p |
| 21:04.31 | rg1_ | anyone here use AGI calls to PHP scripts? |
| 21:05.07 | hmmhesays | every once in a great while |
| 21:05.11 | dhanes | sorry about that Junk...i wouldn't be telling ppl that :) |
| 21:05.19 | rg1_ | hmmmehsays - that for me? |
| 21:05.28 | hmmhesays | rg1_: yeah |
| 21:05.28 | blitzrage | Quebecois! |
| 21:06.08 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 21:06.39 | rg1_ | in dialplan - exten => s,n,AGI(getsstatus,"routine1","prompt1") |
| 21:07.17 | rg1_ | in the php script "getsstatus", how would I reference the two arguments? |
| 21:07.51 | hmmhesays | set them before you call the script? |
| 21:07.59 | JunK-Y | argv |
| 21:08.10 | JunK-Y | like any programming language |
| 21:08.23 | hmmhesays | JunK-Y: i didn't know that, but cool |
| 21:08.27 | rg1_ | junky - $argv[0] ? |
| 21:08.37 | rg1_ | 1, and 2? |
| 21:08.49 | h3x | perl starts with 0 |
| 21:08.50 | JunK-Y | rg1_: depends on ur programming language (agi script) |
| 21:08.53 | h3x | everything else starts with 1 |
| 21:08.58 | h3x | i think |
| 21:08.59 | rg1_ | php junky |
| 21:09.31 | h3x | er wait i think its the other way around |
| 21:11.28 | hmmhesays | it is unfortunate you have to call an interpreter for every single call |
| 21:12.20 | JunK-Y | until u get res_php |
| 21:13.12 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 21:13.15 | benjk_ | hmmhesays, don't worry about it, the dialplan evaluation is so wasteful on cpu cycles, you are acutally getting much better performance once you called an interpreter of another language environment |
| 21:15.49 | hmmhesays | honestly I really don't care, because hardware is cheap |
| 21:17.04 | bugz | hmmhesays: thats ok until you build a box for a call center |
| 21:17.11 | benjk_ | I was only going by your statement |
| 21:17.50 | mog | your gonna say cpu evalutaion to evaluate dial plan is faster than executing a php script? |
| 21:18.14 | mog | maybe you could get close with C agi |
| 21:18.15 | Qwell | mog: Do you have that script of murfs? |
| 21:18.15 | benjk_ | php uses tokenizing and hash table lookups |
| 21:18.19 | mog | but dial plan will still be faster |
| 21:18.23 | mog | ? |
| 21:18.27 | Qwell | the cycles thing |
| 21:18.49 | benjk_ | that's one to two orders of magnitude faster than dialplan evalutaion and lookups |
| 21:19.41 | benjk_ | of course you get most benefit if you do a little bit of work in your AGI script, not just a one liner |
| 21:20.51 | benjk_ | and just in case you wondered, astdb lookups are faster than variable lookups |
| 21:21.34 | benjk_ | too bad you can only access the db from the dialplan by stuffing the intermediate values into variables |
| 21:22.05 | JunK-Y | benjk_: do u have any papers on that: astdb is faster then var? |
| 21:22.19 | benjk_ | astdb uses hashtables |
| 21:22.43 | benjk_ | variable lookup uses linked lists and strcasecmp() many times over |
| 21:24.25 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:24.40 | *** join/#asterisk Crashsys (n=kumba@office.crashsys.com) |
| 21:24.42 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 21:25.12 | JunK-Y | benjk_: ya, i know, the same stuff over and over, like the pbx_findapp |
| 21:25.17 | Crashsys | I know this is an asterisk channel, but is anyone familiar with grandstream phones? (My BT-200 isn't giving a dial tone) |
| 21:25.19 | benjk_ | yep |
| 21:26.01 | JunK-Y | benjk_: that could be changed one day or another. |
| 21:26.29 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 21:26.35 | benjk_ | Junk-Y, http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx/branches/benjk/opbx_dictionary.h |
| 21:27.11 | Qwell | I'm pretty sure variable lookups don't use strcasecmp... |
| 21:27.56 | benjk_ | they do |
| 21:28.01 | Qwell | show me |
| 21:28.15 | JunK-Y | benjk_: ya, i saw ur hash branch yesterday |
| 21:28.20 | benjk_ | its all in pbx.c |
| 21:28.24 | Qwell | where? |
| 21:29.16 | benjk_ | it walks the variable linked list and strcasecmp the name found in the dialplan with the name in the variable struct |
| 21:30.16 | benjk_ | apps and vars are case insensitive, they use strcasecmp |
| 21:30.31 | Qwell | vars are not case insensitive... |
| 21:30.32 | benjk_ | other objects are case sensitive, those use strcmp |
| 21:30.56 | benjk_ | maybe its only the builtin vars then |
| 21:31.14 | benjk_ | still, strcmp isn't much faster |
| 21:32.25 | nDuff | Crashsys: I have some familiarity with the GXP-2000s. Given what junk their "enterprise phones" are, I don't have much faith in any of Grandstream's still-cheaper models. |
| 21:33.09 | benjk_ | <PROTECTED> |
| 21:33.32 | Qwell | benjk_: That is not asterisk |
| 21:33.40 | benjk_ | if current asterisk uses strcmp for this (in pbx.c) then it has been changed in the last 10 months |
| 21:33.42 | mountainm2k | OK, same CentoOS/RHEL-4 server: iaxmodem / libaix2 claims my C++ preprocessor fails sanity check... |
| 21:33.47 | mountainm2k | Is this another similarly easy thing to fix? |
| 21:34.14 | mountainm2k | (or I should say ./configure claims my C++ preprocessor fails sanity check) |
| 21:34.33 | nDuff | Crashsys: That said... is it registered with your server presently? |
| 21:35.07 | *** join/#asterisk ViTa (n=vita@host.200.47.122.173.static.itcsa.net) |
| 21:35.22 | ViTa | hello everyone |
| 21:35.25 | bugz | what would cause mpg123 to eat up the cpu? |
| 21:35.41 | bugz | several running processes of it but only 2 are really using the cpu |
| 21:35.43 | benjk_ | and if it has been changed, it most likely has been changed to strcmp, which isn't really much different |
| 21:36.42 | benjk_ | interpreters have been using tokenizing and hash compares for at least 50 years |
| 21:36.44 | bugz | there are no calls on this box either.. |
| 21:37.56 | JunK-Y | benjk_: feel free to provide a patch to speed up * :) |
| 21:38.16 | mountainm2k | Hahah, nice -- my problem was gcc-c++ wasn't installed... |
| 21:38.20 | mountainm2k | duuuuhhhh. |
| 21:38.52 | benjk_ | JunK-Y, my code is MIT licensed |
| 21:39.19 | JunK-Y | ok |
| 21:39.22 | JunK-Y | lunch time |
| 21:42.12 | Strom_C | I just made my first call to someone who has music instead of audible ringing for the progress audio on their mobile phone, and holy shit is that an irritating feature |
| 21:42.37 | mountainm2k | <PROTECTED> |
| 21:42.55 | Strom_C | pass the m option to Dial() |
| 21:43.24 | mountainm2k | for inbound calls? Interesting, I thought it would actually be difficult... |
| 21:43.29 | *** join/#asterisk Sarum4n (n=some@saruman.demon.nl) |
| 21:43.36 | Strom_C | difficult? Asterisk? |
| 21:43.46 | mountainm2k | lol |
| 21:43.55 | mountainm2k | of course, what the hell was I thinking? |
| 21:45.21 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 21:45.25 | *** part/#asterisk bethaud (n=bethaud@host-84-9-82-198.bulldogdsl.com) |
| 21:45.57 | Strom_C | hmm, latest svn release branch download of zaptel isn't compiling right |
| 21:46.23 | Strom_C | well, actually, it compiles |
| 21:46.27 | Strom_C | but it doesn't install |
| 21:46.58 | Strom_C | "install: cannot stat `*.ima': No such file or directory" - I think it's failing to build the octasic firmware because I don't have that in my system, yet the install script wants to install it... |
| 21:47.40 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 21:48.48 | sx-wks | newbie question: on an E1 span where multiple numbers terminate, how can I get the requested number ? |
| 21:48.53 | Strom_C | DNIS |
| 21:49.12 | Strom_C | or whatever the equivalent name for it in Europe is |
| 21:50.05 | benjk_ | doesn't matter if it is called differently, the identifier in asterisk is still the same |
| 21:50.32 | Strom_C | yes, but it does matter if he's trying to order the service from the telco |
| 21:51.10 | benjk_ | looked more like a how-do-I-access-this-in-my-dialplan question to me |
| 21:51.25 | Strom_C | it's ambiguous |
| 21:52.21 | Strom_C | but it makes sense to confirm that the number is being delivered in the first place before banging on asterisk :) |
| 21:55.25 | sx-wks | well. the thing is that I get surcharged and regular numbers ending on the same span, and want to differentiate users on what number they dialled (that is, on the surcharged number, saturation does not apply) |
| 21:56.01 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.142.5) |
| 21:56.02 | Strom_C | sx-wks: is the telco delivering the number to your equipment? |
| 21:56.14 | Strom_C | sx-wks: and is this a PRI or is it straight channelized E1? |
| 21:56.40 | sx-wks | Strom_C: yes. this is euro-isdn blah in .fr |
| 21:57.21 | sx-wks | (we are working on replacing Rekoll/dialogic with asterisk, and rekoll can see the number, and act on it) |
| 21:57.58 | Strom_C | sx-wks: ok...assuming the telco hasn't munged things up, and assuming what you have doesn't operate too differently from DNIS here in the U.S., the dialed number should be available within your dialplan as the extension the zaptel channel attempts to reach upon call setup |
| 21:58.40 | *** join/#asterisk redondos (n=redondos@190.48.11.72) |
| 21:59.09 | Strom_C | so, assuming you have "context=inbound_pri" or somesuch in zapata.conf, having a single extension in that context called _. will let you see what the telco is passing as DNIS information |
| 21:59.18 | sx-wks | hah, so I should have "exten => _XXXXXXXXXX,1,blah" as the first instruction ? |
| 21:59.28 | redondos | Hello. I'm looking for a way of limiting the number simultaneous connections using a VoIP account (a sip peer). I've tried setting type=peer and call-limit=1 in sip.conf with no success: asterisk lets me use that SIP account any number of times, simultaneously. |
| 21:59.33 | Strom_C | exten => _.,1,NoOP(${EXTEN}) |
| 21:59.40 | sx-wks | ok... |
| 21:59.45 | Strom_C | then base your pattern match strings off of what the telco is sending you |
| 22:00.05 | Strom_C | redondos: loo into using the GROUP() function |
| 22:00.07 | Strom_C | s/loo/look/ |
| 22:00.45 | redondos | Oh, ok, thanks. No way to do it like it's explained with call-limit? |
| 22:00.55 | Strom_C | I believe call-limit is deprecated |
| 22:00.55 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:01.02 | sx-wks | Strom_C: so, I start the thing with one extension per number, doing the right thing for each... |
| 22:01.08 | sx-wks | ok. Pretty simple :D |
| 22:01.32 | Strom_C | yep |
| 22:01.56 | redondos | I see. Thanks man. The wiki is very-very outdated. |
| 22:02.03 | Strom_C | yes it is |
| 22:02.27 | sx-wks | Strom_C: btw, I have a patch for app_meetme . how do I submit it ? |
| 22:02.40 | Strom_C | via the bug tracker |
| 22:02.43 | sx-wks | ok |
| 22:04.58 | *** part/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 22:07.41 | *** join/#asterisk daysmen3 (n=primus@host86-138-208-251.range86-138.btcentralplus.com) |
| 22:09.09 | redondos | Strom_C: Will I have to SET(GROUP) for *every* extension that wants to use the VoIP channel? |
| 22:09.27 | Strom_C | no no no, you set it in the extension that dials the channel |
| 22:09.35 | Strom_C | you put it before the dial() statement |
| 22:09.40 | redondos | Oh, right on. |
| 22:09.59 | redondos | Well, but there are many extensions that dial it. |
| 22:10.27 | Strom_C | look |
| 22:10.33 | Strom_C | when someone dials 13115552368 |
| 22:10.48 | Strom_C | there's a pattern-match in your dialplan for _1NXXNXXXXXX, right? |
| 22:10.56 | redondos | Yeah. |
| 22:11.05 | Strom_C | you do the group checking in THAT extension |
| 22:11.11 | Strom_C | don't confuse extension with station |
| 22:11.14 | redondos | That was what I was asking. |
| 22:11.22 | Strom_C | extension in asterisk == series of commands |
| 22:11.25 | Strom_C | station == telephone set |
| 22:11.29 | redondos | If I had to do the group checking in all of the extensions that will be using it. |
| 22:11.40 | redondos | I think we misunderstood each other, that's all. |
| 22:11.43 | redondos | Let me give you an example. |
| 22:11.49 | redondos | exten=_0030.,1,dial(SIP/011${EXTEN:2}@broadvoice,30) |
| 22:12.16 | redondos | If I want to make the checks there, would I have to set the group first? |
| 22:12.23 | Strom_C | yes |
| 22:12.26 | redondos | And then for _0031 set the group once again? |
| 22:12.30 | Strom_C | yes |
| 22:12.37 | redondos | Ok, that was my question :) |
| 22:12.58 | Strom_C | but if you were smart, you'd set the pattern match to minimize the number of extensions in your dialplan that do exactly the same thing :) |
| 22:12.59 | redondos | I guess I'll have to use a different context to summarize. |
| 22:13.23 | Strom_C | right...you could have them all funnel through a catch-all extension and then do a goto() |
| 22:13.28 | redondos | Well, yeah. But that's one hard regex right there :] |
| 22:16.18 | mountainm2k | Hey, why does * restart the B-channels every so often? Or, a better question is, can, or should, I make it stop doing that? |
| 22:17.11 | Strom_C | mountainm2k: it restarts them once per hour |
| 22:17.26 | Strom_C | mountainm2k: it is completely harmless and actually a Good Thing(tm) |
| 22:17.39 | mountainm2k | but it _doesn't_ restart those that are active, right? |
| 22:17.43 | Strom_C | exactly |
| 22:18.05 | mountainm2k | So, even if it doesn't do anything useful, it can't hurt, so why monkey with it? |
| 22:18.07 | *** join/#asterisk delmar (n=delmar@ip-58-28-149-135.ubs-dsl.xnet.co.nz) |
| 22:18.19 | Strom_C | ? |
| 22:18.48 | mountainm2k | In other words, it doesn't do any harm, and it can be a good thing, so let it be, rather than trying to disable it... |
| 22:18.59 | mountainm2k | (the answer is yes, that is correct) |
| 22:19.00 | mountainm2k | heh |
| 22:19.11 | Strom_C | yes |
| 22:19.29 | Strom_C | i thought you were asking me why you should monkey with it after I said you don't need to |
| 22:20.11 | mountainm2k | heh, no, I was basically answering my own question... I just had never heard of a PBX doing that -- but perhapps they all do it and just don't bother telling me about it... :-P |
| 22:20.19 | Strom_C | perhaps |
| 22:20.37 | Strom_C | there is actually a fairly good reason for why it does that, but I don't remember what it is |
| 22:22.04 | delmar | Has anyone come accross an issue where a router (in this case a WAG54GP2 .. the DSL router with 2 FXS ports) ... goes into a state where calls are suddenly "one way audio" and all sorts of problems... resetting the router and things are ok again.... |
| 22:22.29 | Strom_C | delmar: what protocol? |
| 22:22.35 | Strom_C | and what's on the rest of the network |
| 22:22.36 | delmar | SIP |
| 22:23.10 | delmar | Strom_C, the * box is behind the router, talking to a SIP based DID service. |
| 22:23.31 | Strom_C | uh, no. don't do that :) |
| 22:23.39 | delmar | Strom_C, I was even starting to see the SIP registrations failing all over the place |
| 22:23.49 | Strom_C | sip + nat == headache |
| 22:23.52 | delmar | Strom_C, no choice i the matter. it works fine... |
| 22:24.26 | Strom_C | if it worked fine, you wouldn't be asking this question |
| 22:24.34 | delmar | Strom_C, just recently tho.. it all turned to crap... ive just replaced the router .. if this Linksys is going to go nuts like this i might as well test another type |
| 22:25.01 | delmar | Strom_C, SIP and NAT is a pain yes.. but it works... im just havin router issues. |
| 22:25.14 | delmar | Frankly I think these new Linksys WAG's are hopeless |
| 22:25.22 | Strom_C | well, the problem is definitely your router; swap it out or something |
| 22:25.25 | delmar | there isnt even any documentation on the QoS for this router |
| 22:25.49 | redondos | Strom_C: I'm having a hard time setting this up, do you mind giving me a hand please? |
| 22:26.05 | Strom_C | i'll try |
| 22:26.06 | delmar | Strom_C, yep. back to the alcatel for now. it all came back up nice right away. two-way audio.. calls in/out.. fully tested... all going mint. |
| 22:27.02 | delmar | Strom_C, I wonder what the Linksys is doing to need a reset after a while like this and stuff. |
| 22:27.05 | redondos | http://pastebin.ca/151290 |
| 22:27.06 | sx-wks | delmar: you can try 2.6.18 something that has experimental sip support :D |
| 22:27.33 | delmar | sx-wks, how would that help my situation ? :) |
| 22:27.34 | Strom_C | redondos: your gotoif syntax is borked |
| 22:27.46 | delmar | sx-wks, im not using a Linux box as the router in the middle or anything. |
| 22:27.47 | sx-wks | delmar: use this as your router :D |
| 22:27.47 | redondos | Again, copy/pasted from the wiki. |
| 22:27.56 | Strom_C | line should read: |
| 22:28.05 | redondos | s/think/thinks/ |
| 22:28.15 | Strom_C | redondos: by the way, what version of asterisk are you running? |
| 22:28.19 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 22:28.24 | delmar | sx-wks, nah. if I had a Linux box as the router in this situation... things would be real nice.. * would be running on the box with a public IP.. no NAT issues... and .. I would have better QoS. |
| 22:28.31 | redondos | Strom_C: 1.2.10 |
| 22:28.44 | Strom_C | exten => _0.,2,GotoIf($[${GROUP_COUNT()} > 10]?103) |
| 22:28.45 | delmar | sx-wks, so yeah... I would love to do that... might be able to talk them into letting me do that... we shall see. |
| 22:28.57 | Strom_C | redondos: ok...because you're using really outdated extensions.conf syntax |
| 22:29.17 | redondos | Because of the =_ instead of '=> _' ? |
| 22:29.26 | Strom_C | no |
| 22:29.30 | Strom_C | priority numbering |
| 22:29.34 | redondos | Oh. |
| 22:29.55 | redondos | As I said, I pasted those few lines from an article explaining how to use Groups. |
| 22:29.59 | redondos | How should it read otherwise? |
| 22:30.01 | Strom_C | exten => _0.,2,GotoIf($[${GROUP_COUNT()} > 10]?103) |
| 22:30.10 | Strom_C | for starters |
| 22:30.19 | Strom_C | but also, you |
| 22:30.26 | Strom_C | you're passing the wrong number to broadvoice |
| 22:30.32 | Strom_C | it needs to be 011xxxxxx |
| 22:30.33 | Strom_C | etc |
| 22:30.44 | mountainm2k | OK, so if somebody his * from within Voicemail() it goes to extension 'a' in the current context... |
| 22:30.51 | Strom_C | yes |
| 22:30.51 | mountainm2k | How to set that based on who's voicemail it is? |
| 22:31.11 | mountainm2k | In other words, I want *MY* cover to go to my cell... |
| 22:31.15 | mountainm2k | but not everybody else's... |
| 22:31.20 | Strom_C | exten => a,1,VoicemailMain(${CALLERID(num)}) ? |
| 22:31.23 | mountainm2k | Heh, that should be an option to Voicemail() |
| 22:31.34 | *** join/#asterisk jpeeler (n=jpeeler@130-127-132-164.wireless.clemson.edu) |
| 22:31.41 | Strom_C | or you can set a variable before executing Voicemail() |
| 22:31.42 | redondos | Strom_C: Ok, it's good now. Now how can I pass the real extension to the new context? |
| 22:31.50 | Strom_C | what? |
| 22:31.56 | Strom_C | what do you mean |
| 22:32.01 | redondos | Well, the call matches: exten=_00054[1-8].,1,Goto(broadvoice,_0.,1) |
| 22:32.16 | redondos | Then how do I reference ${EXTEN} from within the [broadvoice] context? |
| 22:32.19 | Strom_C | why arent you just having users dial 011 from the outset? |
| 22:32.32 | redondos | It's a requirement of the company. |
| 22:32.35 | *** join/#asterisk adker (n=adker@74-33-205-58.br1.glv.ny.frontiernet.net) |
| 22:32.38 | *** join/#asterisk num000 (n=numer@e177182208.adsl.alicedsl.de) |
| 22:32.55 | Strom_C | redondos: please find a large screwdriver and find whoever made that requirement and stab them in the eyeball |
| 22:33.08 | Strom_C | but I digress |
| 22:33.11 | redondos | haha |
| 22:33.14 | redondos | I agree. |
| 22:33.20 | Strom_C | your broadvoice context already contains a pattern match |
| 22:33.32 | num000 | can i tell asterisk in the CLI to load all the modules which are mentioned in modules.conf? |
| 22:33.32 | Strom_C | so why are you putting that pattern match again in your goto statement? |
| 22:34.00 | redondos | This? -> _00054[1-8]. |
| 22:34.06 | Strom_C | no |
| 22:34.07 | redondos | That's the pattern match itself. |
| 22:34.18 | redondos | Oh, _0 ? Because I copy/pasted, my bad again. |
| 22:34.19 | Strom_C | you have Goto(broadvoice,_0.,1) |
| 22:34.31 | Strom_C | you need to think rather than just blindly copying and pasting |
| 22:34.42 | redondos | Should I name it whatever I want? |
| 22:34.45 | Strom_C | no |
| 22:35.08 | Strom_C | you should be doing Goto(broadvoice,${EXTEN},1) |
| 22:35.31 | redondos | But then broadvoice will have to have a set of these for every possible extension? |
| 22:35.37 | Strom_C | NO |
| 22:35.51 | Strom_C | your pattern match in the broadvoice context will catch all of those |
| 22:35.51 | redondos | This got me confused. |
| 22:35.57 | redondos | Ah. |
| 22:36.00 | redondos | Neat. |
| 22:36.05 | Strom_C | yes |
| 22:36.14 | Strom_C | then of course in your broadvoice context, you strip off the company's required 000 nonsense and then prepend 011 before the country code |
| 22:36.27 | num000 | what can be the problem although i say autoload=yes in the modules.conf asterisk does not load the modules |
| 22:36.39 | redondos | Sure. 011${EXTEN:3} |
| 22:36.47 | Strom_C | exactly |
| 22:37.19 | redondos | Nicely done. Thanks a bunch. |
| 22:37.51 | redondos | Yup :) |
| 22:38.04 | Strom_C | you're welcome :) |
| 22:38.48 | redondos | The limit isn't really being applied. |
| 22:39.07 | redondos | (I changed it to >1) I can make any number of simultaneous calls, it seems. |
| 22:39.24 | Strom_C | do a noop() and see what groupcount is returning |
| 22:41.00 | redondos | -- Executing NoOp("SIP/101-b760b298", "#{GROUP_COUNT()}") in new stack |
| 22:41.09 | redondos | Sorry |
| 22:41.13 | redondos | s/#/$ |
| 22:41.57 | Strom_C | well what happens when you don't nub up the syntax? |
| 22:42.09 | X-Rob | heh |
| 22:45.57 | redondos | Ok, what is happening is that the GROUP_COUNT var is local to every SIP user. |
| 22:46.02 | redondos | I want it to be global. |
| 22:46.34 | redondos | Currently, the limit doesn't get applied if it's two different users the ones making the calls. |
| 22:47.41 | Strom_C | are you sure you're using it correctly? |
| 22:48.52 | redondos | Pretty much, http://pastebin.ca/151316 |
| 22:50.18 | redondos | Does it look ok to you/ |
| 22:51.20 | Strom_C | you should be doing GROUP_COUNT(Broadvoice) |
| 22:51.32 | Strom_C | I think |
| 22:52.53 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 22:53.29 | redondos | same prolem: a same user can't make 2 calls |
| 22:53.38 | Strom_C | thats what you want :) |
| 22:53.55 | Strom_C | limit the number of calls, right? |
| 22:53.57 | redondos | No. I want to limit it globally. :) |
| 22:54.09 | redondos | If a user is using the Broadvoice channel, then it's busy for everyone else. |
| 22:56.12 | redondos | Maybe with SerGlobalVar instead of just Set? Hm... |
| 22:56.16 | Strom_C | i'll fiddle with it shortly |
| 22:56.22 | Strom_C | im tackling a zaptel problem |
| 22:56.30 | redondos | Awesome. |
| 22:57.25 | redondos | It works! |
| 22:57.44 | redondos | I'm immensely thankful, Strom_C for your attention. :} |
| 22:57.46 | Strom_C | ah, using group with setglobalvar? |
| 22:59.30 | file | zaptel problem? there is no zaptel problem. |
| 22:59.36 | file | I don't know what you're talking about! |
| 22:59.51 | Strom_C | hehehe |
| 22:59.58 | Strom_C | file is my hero |
| 23:01.02 | file | hero? sorry, we don't serve that |
| 23:01.04 | file | only french fries |
| 23:01.55 | Strom_C | file is my french fries |
| 23:02.06 | file | woah, let's not get too personal |
| 23:02.17 | Strom_C | ok ok |
| 23:02.26 | Strom_C | would you settle for being my ketchup/ |
| 23:02.56 | file | fine |
| 23:04.27 | redondos | Strom_C: No, there was no need. |
| 23:04.34 | redondos | Strom_C: I had a problem in the way I was making the calls. |
| 23:04.39 | Strom_C | oh heh |
| 23:04.42 | Strom_C | what was the problem? |
| 23:05.06 | redondos | That the other person that was helping me test this was using the 'actual' format for making calls and not my 'testing groups' format. |
| 23:05.22 | Strom_C | azh |
| 23:05.23 | Strom_C | er, ah |
| 23:05.26 | redondos | Can't trust anyone, I just didn't want to load another softphone... my bad. |
| 23:05.39 | Strom_C | this is the point at which you get a big Sharipe and write NUB on their forehead |
| 23:05.40 | redondos | Anyway, is it possible to send a console command from the shell? |
| 23:05.43 | Strom_C | yes |
| 23:05.44 | redondos | Without attaching to the console. |
| 23:05.48 | redondos | Hehe :) |
| 23:05.50 | Strom_C | asterisk -rx "command goes here" |
| 23:06.19 | redondos | Nice! |
| 23:06.29 | redondos | I was on the -U.. -v part of the man page. |
| 23:06.33 | redondos | I Was getting there :) |
| 23:06.33 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:06.37 | *** join/#asterisk f1assistance (n=carl@cpe-024-163-085-150.nc.res.rr.com) |
| 23:07.01 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 23:07.20 | mtgh | Hi everyone, does anyone know where I can find some doco on the sip config files for cisco 7960 version 8.5 I have a cco acccount |
| 23:09.15 | mtgh | sorry thats 8.4 |
| 23:11.15 | *** join/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net) |
| 23:14.28 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 23:15.10 | tuxd00d | Anyone want to help be figure out noise on Z |
| 23:15.16 | tuxd00d | ZAP channels |
| 23:15.28 | Strom_C | sure |
| 23:15.38 | Strom_C | what kind of zap channels |
| 23:16.29 | tuxd00d | It is making awful noise on both FXO and fXS |
| 23:16.31 | Strom_C | tuxd00d: you want me to call tha number? |
| 23:16.39 | tuxd00d | sure, you can hear it |
| 23:16.50 | Strom_C | is it an analog line going into the zap channel? |
| 23:17.00 | tuxd00d | when the VM picks up |
| 23:17.08 | tuxd00d | it is an analog |
| 23:17.33 | Strom_C | what kind of card |
| 23:18.05 | Strom_C | ok |
| 23:18.12 | Strom_C | what kind of card is it? |
| 23:18.44 | Strom_C | hello? |
| 23:18.54 | Strom_C | this is not a complicated question :) |
| 23:19.13 | file | Strom_C: tuxd00d is a friend, be nice! or I revert the Makefile fixes! |
| 23:19.23 | tuxd00d | TDM400P Rev e/f |
| 23:19.26 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:19.27 | Strom_C | hmmm |
| 23:19.35 | Strom_C | what does zttest return? |
| 23:19.37 | tuxd00d | had to look it up |
| 23:20.25 | tuxd00d | Strom_C: hold please |
| 23:20.32 | tuxd00d | thanks file ;-) |
| 23:20.47 | Strom_C | hehe, i was just getting impatient, not angry :) |
| 23:21.19 | file | do you want to try our new lobster sub? |
| 23:21.22 | file | is it made with real lobster? |
| 23:21.26 | file | we got another one asking if it's real lobster! |
| 23:21.30 | *** part/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 23:21.33 | Qwell | eh? |
| 23:21.42 | file | they say lobster in this commercial in every sentence |
| 23:21.46 | Qwell | oic |
| 23:21.49 | Qwell | Is it real lobster? |
| 23:21.55 | *** part/#asterisk f1assistance (n=carl@cpe-024-163-085-150.nc.res.rr.com) |
| 23:21.56 | file | I honestly don't know |
| 23:22.37 | tuxd00d | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.988091 |
| 23:22.42 | Strom_C | weeeird |
| 23:22.43 | Strom_C | hmmm |
| 23:22.58 | Strom_C | is it sharing an interrupt? |
| 23:23.14 | tuxd00d | It is sharing an IRQ with ACPI |
| 23:23.43 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 23:23.44 | tuxd00d | Must have chaned, it didn't used to |
| 23:24.06 | Strom_C | that might be the problem |
| 23:24.17 | BlepsoaF | Hello all, I just hooked up my nortel system to asterisk ( also new to this ), but I'm having trouble with calls outgoing... can someone take a look at http://pastebin.com/777531 to see what is wrong in my dial plan...Im sure its stupid |
| 23:24.28 | tuxd00d | okay, I'll find a monitor and keyboard and turn it off |
| 23:24.36 | tuxd00d | or move it or something |
| 23:24.41 | Strom_C | BlepsoaF: i'm looking |
| 23:24.52 | tuxd00d | thanks Strom_C |
| 23:25.08 | Strom_C | BlepsoaF: because you dont have a pattern match for seven digit numbers |
| 23:25.08 | BlepsoaF | its not working when I try to dial XXX-XXX-XXXX but works for X-XXX-XXX-XXXX |
| 23:25.53 | BlepsoaF | im not trying to dial 7 digits |
| 23:26.04 | Strom_C | BlepsoaF: in that example, something is dialing 203-9552 |
| 23:26.11 | Strom_C | and nothing but 203-9552 |
| 23:26.18 | BlepsoaF | right thats the issue |
| 23:26.43 | Strom_C | well, either fix the translations in your nortel system or add a seven digit pattern match |
| 23:27.10 | Strom_C | to asterisk |
| 23:27.32 | *** join/#asterisk ANTILOCAS (n=uoiuyiu@200.87.51.226) |
| 23:27.33 | BlepsoaF | its works fine with nortel just plugged into the t1 so I dont think its something with the translation there |
| 23:27.45 | Strom_C | asterisk is between nortel and t1? |
| 23:27.50 | BlepsoaF | yes |
| 23:27.55 | file | I wish I had a Lenny's sub right about now |
| 23:27.59 | BlepsoaF | t1 into asterisk |
| 23:28.00 | Strom_C | well, the T1 accepts seven digits then |
| 23:28.09 | BlepsoaF | then a patch to the nortel from asterisk |
| 23:28.13 | Strom_C | and the nortel is picking up the T1 to asterisk and dialing seven digits into it |
| 23:28.18 | Strom_C | so, like I said |
| 23:28.45 | Strom_C | either modify your nortel translations to send ten digits to asterisk, or modify your asterisk configuration to accept seven digits from the nortel system |
| 23:29.29 | ANTILOCAS | Abybody uses the Grandstream HandyTone HT-496? |
| 23:30.21 | BlepsoaF | Ok thanks |
| 23:30.38 | Strom_C | BlepsoaF: are the T1s PRIs or just straight channelized T1s? |
| 23:31.41 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 23:33.48 | ANTILOCAS | Im trying to install the Grandstream HandyTone HT-496. My network arquitecture is star topology where even the modem-router is conected to my switch. The HandyTone manual says conect |
| 23:33.57 | ANTILOCAS | a) insert the eternet cavble into the WAN port of HandyTone and conect the other end of the eternet cable to an uplink port" |
| 23:33.58 | ANTILOCAS | b) connect a PC to the LAN port of HandyTone |
| 23:34.06 | ANTILOCAS | Knowing my architecture how would this be? |
| 23:34.12 | Strom_C | ignore step B |
| 23:34.42 | Strom_C | or, alternatively, buy the handytone without the built-in router |
| 23:37.46 | ANTILOCAS | u mean this handytone comes with a router? |
| 23:38.24 | Strom_C | hence the WAN and LAN ports |
| 23:38.55 | ANTILOCAS | but if i ignore step B, how would i control the calls? with a software? |
| 23:38.59 | ANTILOCAS | is it independetly? |
| 23:39.41 | *** join/#asterisk f1assistance (n=carl@cpe-024-163-085-150.nc.res.rr.com) |
| 23:39.53 | Strom_C | ANTILOCAS: you plug a telephone into the handytone |
| 23:40.22 | ANTILOCAS | yeah i know but when i sign with voip provider. will they just need the ATA ip? |
| 23:40.47 | Strom_C | no, you need to configure the ATA |
| 23:41.08 | Strom_C | preferably you configure your asterisk box to talk to the provider and configure your ATA to talk to the asterisk box |
| 23:41.19 | ANTILOCAS | i dont use asterisk yet |
| 23:41.31 | Strom_C | now would be a fine time to start :) |
| 23:41.36 | ANTILOCAS | i was thinking to use net2phone |
| 23:41.58 | tuxd00d | Strom_C: It's on it's on IRQ now, but it is still wack |
| 23:42.11 | Strom_C | tuxd00d: weirdful |
| 23:42.23 | Strom_C | that's the only card in your system, right? |
| 23:42.24 | tuxd00d | Yes, very weirdful |
| 23:42.49 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:43.26 | tuxd00d | Strom_C: Yes, online on TDM |
| 23:43.35 | tuxd00d | only one |
| 23:43.37 | Strom_C | heh |
| 23:43.48 | Strom_C | i'm stumped...wait till tomorrow and call digium tech support |
| 23:44.08 | tuxd00d | sounds good, thanks for you help |
| 23:44.37 | Strom_C | sorry i couldnt fix the problem |
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| 23:49.49 | mmurdock | howdy all. |
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| 23:50.27 | Strom_C | howdy |
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