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| 00:48.19 | Pegasus_RPG | Hello. I have a strange situation: I need to connect to a 20-year-old phone system remotely to make some programming changes. It only supports a 2400 baud connection, so I tried connecting through an ATA using ulaw, but I can't establish a connection (the call drops as soon as the audible modem negotiation completes.) There's nothing in the verbose * log other than a typical hangup message. |
| 00:48.55 | Pegasus_RPG | Do I need to do something special to allow a modem connection? Or is the nature of VoIP such that it just won't work? |
| 00:53.04 | Pegasus_RPG | reads about V.150.1 |
| 01:45.58 | WIMPy | Yes, you need a lot of luck for that. |
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| 03:19.10 | AsterRack | Hey all, I'm trying to get gTalk up and running on a cloud server. When I make a call i get "'sent to invalid extension but no invalid handler: context,exten,priority=incoming'" any suggestions? |
| 03:24.55 | AsterRack | http://pastebin.com/s8iv0Te2 << for config info |
| 04:17.43 | ChannelZ | well you have no extension email@example.com |
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| 12:54.03 | tompaw | Anyone successfully built 11.7 on osx 10.9? |
| 13:00.24 | dozepih | Is it common among SIP providers to have incoming calls coming from an IP-address which differs from the one you register to? |
| 13:01.52 | dozepih | I solved it (i think) by having different peers in sip.conf, one for each address. Is that the correct way of doing it? |
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| 13:22.53 | carrar | dozepih: yes |
| 13:23.21 | carrar | dozepih: that typically means the server you are registering to is a SIP Proxy |
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| 14:03.26 | LemensTS | http://pastebin.com/vjMb4hn8 Anyone know why it is giving me this error? |
| 14:06.06 | carrar | Looks like you aready have defined context 'dentist-in' someplace else |
| 14:07.10 | carrar | maybe |
| 14:07.22 | LemensTS | That would make sense, but just searched extensions.conf for dentist-in and only the one exists. |
| 14:07.34 | carrar | not including anythign? |
| 14:07.36 | LemensTS | I don't even know what -1 is |
| 14:11.13 | carrar | What version you running? |
| 14:11.17 | carrar | it's just a warning also |
| 14:14.00 | carrar | nothing else in that context? |
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| 14:21.03 | LemensTS | Asterisk 1.8.6.0 |
| 14:21.16 | LemensTS | Thats all that is in the context |
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| 15:47.11 | starbuck | hello, i got a problem with a cisco ip phone (7941) - status messge is 'No Trust List Installed' |
| 15:47.21 | starbuck | what could i do about it? |
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| 19:32.24 | vandyk | I've been trying to setup a SPA400 as SIP trunk on Asterisk. I've been searching for week a solution without success. All samples and references I found, is similar to this: http://www.manualslib.com/manual/95884/Linksys-Tribox-Spa400.html?page=6#manual |
| 19:34.57 | kippi | hey |
| 19:35.24 | kippi | has anyone managed to output a audio stream to jack? |
| 19:41.08 | vandyk | the problem is that on Asterisk it always keep telling me that SPA400 does not implement REGISTER |
| 19:41.30 | vandyk | the only difference of my scenario is that SPA400 and Asterisk are on different networks |
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| 19:56.58 | ChannelZ | vandyk: what do you mean Asterisk tells you the SPA400 doesn't implement REGISTER? The SPA should be registering TO Asterisk, not the other way around. |
| 19:57.48 | vandyk | yeah I know, but the way that SPA400 works it says that Asterisk should register on SPA400 |
| 19:58.03 | vandyk | could you please take a look on PDF link that I sent? |
| 19:59.27 | blitzrage | vandyk: either the device is trying to register and you don't have host=dynamic (in which case asterisk will complain if the device attempts to register), or Asterisk needs to register, and you need to configure asterisk with a 'register' line in sip.conf |
| 20:01.33 | vandyk | I'll try add the register on sip.conf |
| 20:01.49 | kippi | on asterisk, if I have a call and I want to take the audio what is the best way to get it out off asterisk? |
| 20:02.50 | blitzrage | kippi: canreinvite=yes or directmedia=yes |
| 20:03.22 | kippi | i need to get the stream to FFMPEG |
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| 20:10.06 | vandyk | @blitzrage: no way. Even putting host=dynamic on trunk peer details and register on sip.conf |
| 20:10.29 | vandyk | Asterisk always tells me that SPA400 does not implement REGISTER |
| 20:11.10 | blitzrage | vandyk: maybe it doesn't then |
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| 20:11.28 | blitzrage | I'm not sure if you're trying to get the spa to register to asterisk, or if you're trying to get asterisk to register to the spa |
| 20:11.48 | ChannelZ | The SPA should register to Asterisk. |
| 20:12.08 | vandyk | I found at least 4 or 5 posts on internet from people that uses SPA400 as trunk. There is something that I'm missing probably. |
| 20:12.08 | blitzrage | either way, those are the methods to allow / enable the registrations. Beyond that, it is either the lack of support on the device, or authentication issues |
| 20:12.17 | blitzrage | runs away |
| 20:12.31 | vandyk | but on SPA there is no option to put user and password |
| 20:12.40 | vandyk | only the host |
| 20:12.51 | vandyk | I'll try remove the register from sip.conf |
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| 20:13.57 | ChannelZ | If you bother to even do it at all. If you're using static IPs and configuring them on both ends, there's no reason for anyone to register. |
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| 20:42.04 | vinise | hi all I've really big trouble configuring asterisk/freepbx with sonatel call. outgoing call work correctly but impossible to get incoming call : "","+3XXXXXXXXXX","s","from-sip-external","""+3XXXXXXXXXX"" <+3XXXXXXXXXX>","SIP/sonetel.net-0000009a","","Playback","ss-noservice","2014-01-18 20:23:07","2014-01-18 20:23:07","2014-01-18 20:23:12",5,5,"ANSWERED","DOCUMENTATION","1390076587.154","" |
| 20:42.50 | [TK]D-Fender | vinise: Your call is not matching your trunk, or your trunk is configured wrong |
| 20:43.00 | WIMPy | You probably haven't configured your number correctly, but |
| 20:43.27 | WIMPy | ~freepbx |
| 20:43.27 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:43.28 | [TK]D-Fender | vinePlease join #freepbx for continued support for this. |
| 20:43.29 | [TK]D-Fender | vinise: Please join #freepbx for continued support for this. |
| 20:43.52 | vinise | ok thanks ;) going to ask into freepbx |
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