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| 02:50.30 | mnathani | How can I troubleshoot a SIP connection using my IP Phone with the Aterisk server? |
| 02:50.47 | mnathani | The phone does not seem to register |
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| 03:57.44 | lorsungcu | mnathani: im fukcing wasted |
| 03:58.09 | lorsungcu | mnathani: but u thunk almost anyone would askj for some logs |
| 03:58.30 | lorsungcu | mnathani: so fucking pb some goddamn logs |
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| 08:45.06 | kresp0 | Hi all, I'm having no audio issues when calling from a sip phone registered on my * to another * echo test. Network map is like this: |
| 08:45.09 | kresp0 | sip phoneLAN my * pbx |
| 08:45.13 | kresp0 | 192.168.2.4 <---------> 192.168.2.1 VPN friend * pbx |
| 08:45.17 | kresp0 | <PROTECTED> |
| 08:45.20 | kresp0 | upss |
| 08:45.45 | kresp0 | i mean, network is like this: http://pastebin.com/NKHGsHkz |
| 08:46.50 | kresp0 | damit, screwed again, sorry!. http://pastebin.com/NGiyiD3z |
| 08:47.03 | kresp0 | My sip.conf: http://pastebin.com/ZzcWiqhP |
| 08:47.08 | kresp0 | And sip debug on friend pbx: http://pastebin.com/6TpQxcs7 |
| 08:47.29 | kresp0 | Tried "nat=force_rport,comedia" (old sip=yes) on peer definition at sip.conf, but still get "Peer audio RTP is at port 192.168.2.4:41000" on remote PBX. |
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| 09:42.33 | Rhomber | i've found a nice place to get the prompts.. but they only provide in 3 formats.. gsm, alaw and slin |
| 09:43.26 | Rhomber | i'm guessing it's better to convert the slin to ulaw to have it as well rather than rely on on-the-fly re-encoding... what's the best way to do that? and i assume slin is the best file to use for that? being 128Kbps |
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| 10:10.13 | ChannelZ | unless you have thousands of calls or something, letting asterisk transcode is nothing |
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| 10:54.07 | Rhomber | ah cool, fair enough then :) |
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| 12:09.45 | StaRetji | Howdy folks |
| 12:09.59 | kresp0 | hi |
| 12:10.36 | StaRetji | is there a way to set asterisk to allow only 1 consequent calls for any caller id in given time period? |
| 12:11.15 | StaRetji | for example, if 4455 makes a call, then in next 5 minutes can not make another call |
| 12:11.29 | Gugge | StaRetji: yes, using GROUP_COUNT and IFTIME |
| 12:11.45 | Gugge | ohh, then you need some kind of database to save the state in too |
| 12:11.45 | StaRetji | Gugge: thx man |
| 12:12.13 | StaRetji | seems like one hellawa task for me :D |
| 12:12.34 | StaRetji | let me google if there are examples |
| 12:17.11 | StaRetji | hm, google is not my friend this round :/ |
| 12:23.05 | StaRetji | basically, I need to limit cal attempts, 1 clid 1 call attempt in 5 minutes |
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| 12:52.14 | Rhomber | hmm, Voicemail(${mbx},u) ... the 'u' part isn't generating the right sound .. it's using the system sounds and not the language=au ones.. is there some config i might have overlooked, i changed language=au everywhere i could think of |
| 12:58.53 | kresp0 | StaRetji, try using GotoIfTime and astdb. Sorry I dont know how to extract current time using asterisk functions. But maybe you can use a shell comand (date +%s) to get the current unix time |
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| 13:13.30 | gavimobile | hey folks, I need some assistance here. sometimes people are waiting in the queue too long. I want to give them the option to press a number which will forward to a voicemail box |
| 13:14.10 | gavimobile | periodic-announce will periodically play my announcement while waiting in the queue && |
| 13:14.11 | gavimobile | periodic-announce-frequency = 45 ; every 60 seconds |
| 13:14.47 | gavimobile | but how do I give them the option to press a number? via context from queues.conf? |
| 13:20.31 | kresp0 | ; A context may be specified, in which if the user types a SINGLE |
| 13:20.31 | kresp0 | ; digit extension while they are in the queue, they will be taken out |
| 13:20.31 | kresp0 | ; of the queue and sent to that extension in this context. |
| 13:20.31 | kresp0 | ; |
| 13:20.32 | kresp0 | ;context = qoutcon |
| 13:21.18 | kresp0 | try that option gavimobile |
| 13:21.23 | kresp0 | on queues.conf |
| 13:22.31 | gavimobile | kresp0: I got it, now in addition, I would like to save the queue position number and have my pbx call them back when their turn comes |
| 13:22.46 | gavimobile | what's the logic here |
| 13:23.28 | gavimobile | in the dialplan which == queues context , I need a variable which saves the queue position of the caller I assume, but how do I save the position? |
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| 13:35.21 | gavimobile | maybe ill reword my question.. when going to the context specified in queues.conf, does that caller loose his position in the queue? |
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| 13:57.49 | kresp0 | that caller gets kicked out of the queue. Maybe you should send them to another queue then. |
| 13:58.22 | kresp0 | and in that other queue put the Dial to call them back |
| 14:00.48 | kresp0 | oh now I understand what you want to do |
| 14:02.18 | kresp0 | Caller -> queue -> press key to exit -> hung up call -> add call to an outgoing queue |
| 14:02.37 | kresp0 | they will be in order |
| 14:25.26 | gavimobile | kresp0: thanks, what do you mean by outgoing queue? |
| 14:26.15 | gavimobile | like a spool of outgoing calls? |
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| 14:42.40 | kresp0 | a queue where members are outgoing calls |
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| 22:24.42 | kowala | hello. i have a question regarding asterisk configuration and existing modules. is there a resource module available that cloud connect to a rest service (instead of mysql or pgsql) and retrieve sip account data? is this even possible to achieve? |
| 22:25.26 | beefcafe | ~~ |
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| 22:35.03 | [TK]D-Fender | The words are English... but the order and usage is incomprehensible |
| 22:39.39 | edong23 | im thinking that is "could connect" |
| 22:39.49 | edong23 | oh.. maybe not |
| 22:39.52 | edong23 | hm... |
| 22:40.05 | edong23 | kowala: where is the decoder ring? |
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| 22:43.23 | kowala | sorry about my spelling. |
| 22:44.05 | kowala | the issue I have is that databases cant be exposed to asterisk and all data available for integration is available through web services |
| 22:46.01 | kowala | i need to make asterisk to communicate with those webservices for realtime data |
| 22:49.32 | [TK]D-Fender | kowala: Go look at the Database integration section of the book. That is what we have. |
| 22:49.32 | [TK]D-Fender | ~book |
| 22:49.32 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:49.33 | [TK]D-Fender | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:52.36 | kowala | at the moment I am thinking of writing a resource module, similar to the LDAP one |
| 22:52.37 | kowala | if there will be no other way |
| 22:53.33 | kaldemar | kowala: take a look at res_config_curl |
| 22:55.16 | kowala | [TK]D-Fender: thanks for the reply. I have the definitive guide 3rd edition. the database integration is not possible in my deployment scenario. |
| 22:55.21 | kowala | kaldemar: thank you. I will check it out. |
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