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| 00:25.27 | last1 | I know it's not an asterisk question...but can anyone help with a polycom phone question ? |
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| 00:38.01 | Samot | kfife: There really isn't a command for that. |
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| 01:58.27 | drmessano | Any sane provisioning system would create configs and set the credentials in Asterisl |
| 01:58.31 | drmessano | asterisk* |
| 01:58.44 | drmessano | not try to scrape them from Asterisk to shove in a config |
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| 05:54.18 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.35.0 (2020/07/16) 16.12.0 (2020/07/16) Standard: 17.6.0 (2020/07/16); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 11:11.51 | lokensap | Hello All |
| 11:12.02 | lokensap | I need help in configuring freepbx |
| 11:12.15 | lokensap | I am near to giving up on it |
| 11:12.38 | lokensap | I am registered to SIP provider but the incoming and outgoing calls fails |
| 11:13.00 | lokensap | The provider says that they have received traffic related to registration only |
| 11:13.21 | lokensap | no any outgoing call packets have reached to them! |
| 11:14.57 | lokensap | I can provide configuration details too! can anyone help here! |
| 11:15.23 | lokensap | I am near to giving up on this! and I am new to asterisk too! |
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| 11:16.23 | lokensap | hello |
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| 11:35.02 | lokensap | ?ask https://pastebin.com/rK3p3S91 |
| 11:35.26 | lokensap | !help |
| 11:36.05 | lokensap | Freepbx is registered but incoming and outgoing calls are not happening |
| 11:40.53 | lokensap | This is the response I am getting: Received response: "Forbidden" from '<sip:9003@ims.ntc.net.np>;tag=as3d2c45a1'Any suggestions? |
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| 13:37.42 | cal-tec | Does anyone know why an isdn2e (bri) connected to a B410P card would not be displaying the full caller id? It is missing a digit or more for international calls. eg. 07887xxxxxx becomes 7887xxxxxx for inbound calls. outbound also needs to be in the format 7887xxxxxx. It is a UK isdn2e line with BT as the provider. Im not sure if this is as the number is being sent by BT or if it it a config error. Imusing freepbx as the front |
| 13:37.42 | cal-tec | end and have tried all combinations of settings I can find but im guessing it is something simple |
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| 14:06.37 | *** mode/#asterisk [+o kharwell] by ChanServ |
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| 15:56.01 | lokensap | HelloI have configured and registered the Freepbx to the provider! I am not able to do any inbound and outbound calls! can anyone assist me!Everytime I am running into the error: app_dial.c:2591 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)I have dial plans configured too! |
| 15:57.22 | igcewieling | stop using chan_sip. |
| 15:57.49 | lokensap | I am unable to register using the pjsip |
| 15:58.24 | lokensap | I have tried but the registration failed |
| 15:59.32 | lokensap | I will try it again! |
| 15:59.39 | lokensap | Thank You |
| 16:03.38 | post-factum | lokensap: use pjsip and try to follow some good manual like https://zadarma.com/en/support/instructions/asteriskpjsip/ |
| 16:03.50 | post-factum | i'm not sure using freepbx is a good idea |
| 16:03.54 | post-factum | but whatever |
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| 16:09.43 | Samot | Why isn't it a good idea? |
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| 16:32.56 | post-factum | for simple configuration it is just an extra unneeded abstraction. for complex configurations there are other ways |
| 16:33.07 | post-factum | but who knows what's the use case, hence "not sure" |
| 16:33.41 | Samot | There are other ways to provide a GUI interface like that? |
| 16:33.51 | Samot | I mean outside of making your own. |
| 16:33.57 | igcewieling | It isn't a good idea because this is #asterisk, not #FreePBX. |
| 16:40.28 | post-factum | ⺠|
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| 19:44.08 | lokensap | Anyone knows about the error: chan_sip.c:11118 process_sdp_a_audio: Got Opus sprop-maxcapturerate=1600023345 handle_response_invite: Received response: "Forbidden" from '<sip:9003@ims.ntc.net.np>;tag=as28578d2a' |
| 19:44.41 | lokensap | Sorry this is the error: |
| 19:45.02 | lokensap | chan_sip.c:11118 process_sdp_a_audio: Got Opus sprop-maxcapturerate=16000 |
| 19:45.09 | lokensap | handle_response_invite: Received response: "Forbidden" from '<sip:9003@ims.ntc.net.np>;tag=as28578d2a' |
| 19:45.18 | lokensap | ?ask |
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| 20:06.49 | lokensap | Anyone knows about the error: chan_sip.c:11118 process_sdp_a_audio: Got Opus sprop-maxcapturerate=16000 |
| 20:07.01 | lokensap | 23345 handle_response_invite: Received response: "Forbidden" from '<sip:9003@ims.ntc.net.np>;tag=as28578d2a' |
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| 21:28.46 | post-factum | incorrect user/password? |
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| 21:32.49 | post-factum | maybe, you are just not registered |
| 21:32.58 | post-factum | or maybe your codecs do not match |
| 21:33.19 | post-factum | or maybe there's/isn't srtp enforcement |
| 21:33.20 | post-factum | who knows |
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