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| 10:08.41 | Zizo | Hi all, in my sip.conf file there are two "register" lines which match two incoming numbers from the same provider. I need to differentiate their incoming context, context_a for the first incoming number and context_b for the second one, but i'm unable to specify them. Any hint about this? Thanks |
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| 10:42.19 | kafal | hi, I am upgrading from asterisk 12.3 to 12.4. I use static realtime to load agents.conf data. I am getting the following error in the log: [FreeTDS][SQL Server]'LENGTH' is not a recognized built-in function name. |
| 10:42.27 | kafal | any ideas how to fix it? |
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| 16:09.21 | volga629 | Hello Everyone, asterisk tls is should accept sha1 or cert digest need to be md5 ? |
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| 16:32.18 | vaewyn | Hey all ... playing with 12.3.2 with some polycom phones using pjsip... all local network (no nat/firewall between phones and server) ... calls work but I don't receive audio for the first couple seconds until I see on the console "Probation passed -- setting RTP source address to ...." doing a rtp set debug on I see lots of rtp to the phone... but not from the phone until that message is shown. Any ideas how I can debug this? |
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| 17:42.04 | gravspeed | good morning |
| 17:43.23 | gravspeed | i want to make a little vb app to display information about our asterisk pbx and link phone call info to our ticketing system, what is the best method to get the call information out of an asterisk system? |
| 17:45.19 | [TK]D-Fender | "the call information" <-- too vague |
| 17:45.53 | [TK]D-Fender | What information? When? |
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| 17:47.53 | [TK]D-Fender | [13:45][TK]D-Fender"the call information" <-- too vague |
| 17:47.55 | [TK]D-Fender | [13:45][TK]D-FenderWhat information? When? |
| 17:48.27 | gravspeed2 | like when a call comes in, get the callerid |
| 17:49.44 | [TK]D-Fender | And how do you get information into your ticketing system? |
| 17:49.51 | gravspeed2 | i could parse out the cli, which i was playing with using node.js |
| 17:50.06 | gravspeed2 | ticketing system has a very robust api |
| 17:50.25 | gravspeed2 | i will make calls to it using information gathered from the asterisk system |
| 17:50.35 | [TK]D-Fender | Then make a script to push it there and call it when your call comes in. |
| 17:54.23 | gravspeed2 | right, but i'm wondering ioof there is a better way to know when the call comes in then parsing the cli? |
| 17:55.32 | [TK]D-Fender | [13:50][TK]D-FenderThen make a script to push it there and call it when your call comes in. |
| 17:57.34 | gravspeed2 | so... parsing the cli? or is there another interface i should be using? |
| 17:59.54 | [TK]D-Fender | In the DIALPLANNo, parsing the CLI crazy |
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| 18:04.56 | gravspeed2 | so, then what should i be using? |
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| 18:05.50 | [TK]D-Fender | DIALPLAN |
| 18:05.53 | [TK]D-Fender | call your script |
| 18:06.32 | gravspeed2 | i never even thought of that, i can make an external script part of the dialplan? |
| 18:06.54 | [TK]D-Fender | you CALL it from there |
| 18:06.57 | [TK]D-Fender | there are apps for that. |
| 18:07.00 | [TK]D-Fender | and functions. |
| 18:07.10 | [TK]D-Fender | and AGI which you you should already be well aware of |
| 18:08.59 | gravspeed2 | fastagi looks promising... http://www.voip-info.org/wiki/view/Asterisk+FastAGI |
| 18:09.42 | [TK]D-Fender | not needed for basic logging. |
| 18:09.47 | [TK]D-Fender | AGI itself isn't |
| 18:09.58 | [TK]D-Fender | do a shell call and pass it as a parameter |
| 18:12.41 | gravspeed2 | i would like to do something like a long wait http request so as to keep network chatter down and response times low |
| 18:13.20 | gravspeed2 | agi won't do that on it's own will it? |
| 18:13.30 | [TK]D-Fender | AGI is not a magic specific thing |
| 18:14.06 | [TK]D-Fender | YOU write your AGI in whatever language you want and it's YOUR code that gets executed |
| 18:14.19 | [TK]D-Fender | If it calls a process that takes a while then you are blocking the call during that time |
| 18:14.41 | gravspeed2 | hmm... |
| 18:15.47 | [TK]D-Fender | You should considering how you actually are able to do what you need to do. |
| 18:16.59 | gravspeed2 | i've been tinkering with loing wait stuff in php and node.js, seems like that could work, but now i'm worried about my script hanging and leaveing a phone call in the wind |
| 18:18.19 | [TK]D-Fender | You've mentioned "long wait" without any details about what you're intending to communicate with, where it is, what ways you can push your data to it, how it's stored... or much else. |
| 18:19.08 | [TK]D-Fender | So far it's "push caller-id 'somewhere'".... |
| 18:19.20 | [TK]D-Fender | Not sure why this should be a long process.... |
| 18:20.59 | gravspeed2 | no storage really, just a relay to a windows vb app to pop up a notifier with the caller id and info from our ticketing system, company/contact info, active tickets, etc. |
| 18:21.53 | [TK]D-Fender | Then you should probably be thinking of sending it via AMI instead |
| 18:22.18 | gravspeed2 | so the vb app will get told what extension it should be watching for and then use something like a long wait http request to approximate push (or maybe i'll make the app listen on the network, but it seems better if it connects out) |
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| 19:49.24 | johnnygun1 | hello, I am very new to asterisk but I have a couple of questions and perhaps a very small beginners project that I need assistance with, I am willing to pay either paypal or BTC, However, it will take me some time ot exchange from paypal to BTC. Please private message me if interested |
| 19:51.33 | johnnygun1 | I also can communicate using skype as a medium or TOX |
| 19:55.59 | johnnygun1 | Is this chat even active? |
| 19:56.29 | WIMPy | You're unlikely to get serious offers if you don't provide any information on what you want. |
| 19:58.01 | johnnygun1 | I just rented a asterisk pbx and i need help setting it up, i also have a python script that i found and it supposedly works with asterisk but I am not sure how to implement it, I have been watching various youtube tutorials on how to add a sip trunk( believe thats the term) which I also bought (flowroute) I am wondering if there are better providers other then flowroute and if I could have help |
| 19:58.38 | WIMPy | There are plenty. |
| 19:59.09 | WIMPy | But if you rented an Asterisk, you probably can only use some web interface your provider gave you. |
| 19:59.38 | WIMPy | They are not supported here and may even be completely unknown. |
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| 20:00.05 | johnnygun1 | i can ssh into it |
| 20:00.07 | johnnygun1 | there is no gui |
| 20:00.21 | WIMPy | ok. |
| 20:00.33 | WIMPy | Might be something better then. |
| 20:00.45 | johnnygun1 | <PROTECTED> |
| 20:00.53 | johnnygun1 | do you mind if I private message you |
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| 20:01.49 | WIMPy | You can, but that way you won't have a chance of other people helping you. |
| 20:05.36 | WIMPy | ~itsp |
| 20:05.36 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:05.43 | WIMPy | ~itsplist-us |
| 20:05.44 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
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| 20:47.20 | kafal | hi, I am facing an issue with static realtime in asterisk 12.4. My setup was working fine for 12.3. Now I am upgrading to asterisk 12.4 and static realtime is not working. |
| 20:47.59 | kafal | I am storing my agents.conf in MS SQL Server. I keep getting: res_odbc.c: SQL Execute returned an error -1: 42000: [FreeTDS][SQL Server]'LENGTH' is not a recognized built-in function name. error |
| 20:48.37 | kafal | I debugged and found that it could be due to the chnages done in this ticket: https://issues.asterisk.org/jira/browse/ASTERISK-23582 |
| 20:49.04 | kafal | As there is no LENGTH function in MS Sql Server, therefore, I am getting this error. |
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| 20:49.19 | lorenzo | hi! what could cause no audio output |
| 20:49.23 | lorenzo | both with Festival and eSpeak? |
| 20:49.26 | kafal | Is there any way I can configure ODBC to translate LENGTH to LEN? |
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| 20:54.13 | sekil | you would need to change the stmt |
| 20:55.55 | kafal | But the statement is in res_odbc.c file |
| 20:55.58 | lorenzo | I can't even get gsm sounds to play :/ |
| 20:56.11 | kafal | It was added due to the changes in this ticket https://issues.asterisk.org/jira/browse/ASTERISK-23582 |
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| 21:14.04 | lorenzo | ha nvm I was just missing nat |
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