Re: [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote: Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 66) Maybe this: Local channel Description: Local Proxy Channel Driver Syntax: Local/[EMAIL PROTECTED]/n Configuration file: none chan_local is a pseudo-channel. Use of this channel simply loops calls back into the dialplan in a different context. Useful for recursive routing. Notes: Adding /n at the end of the string will make the Local channel not do a native transfer (the n stands for no release) upon the remote end answering the line. This is an esoteric, but important feature if you expect the Local channel to handle calls _exactly_ like a normal channel. If you do not have the no release feature set, then as soon as the destination (inside of the Local channel0 answers the line, the variables and dial plan will revert back to that of the original call, and the Local channel will become a zombie and be removed from the active channels list. This is desirable in some circumstances, but can result in unexpected dialplan behavior if you are doing fancy things with variables in your call handling. Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
4 jan 2008 kl. 11.50 skrev Benchev: On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote: Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED] sip' (cause = 66) Maybe this: Local channel Description: Local Proxy Channel Driver Syntax: Local/[EMAIL PROTECTED]/n Configuration file: none chan_local is a pseudo-channel. Use of this channel simply loops calls back into the dialplan in a different context. Useful for recursive routing. You have to enable chan_local in menuselect (1.4) and make sure it's not disabled in modules.conf. This is not a developer question, so please take this kind of questions to asterisk-users in the future. Thank you! Best regards, /Olle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
4 jan 2008 kl. 13.06 skrev Johansson Olle E: 4 jan 2008 kl. 11.50 skrev Benchev: On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote: Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED] sip' (cause = 66) Maybe this: Local channel Description: Local Proxy Channel Driver Syntax: Local/[EMAIL PROTECTED]/n Configuration file: none chan_local is a pseudo-channel. Use of this channel simply loops calls back into the dialplan in a different context. Useful for recursive routing. You have to enable chan_local in menuselect (1.4) and make sure it's not disabled in modules.conf. This is not a developer question, so please take this kind of questions to asterisk-users in the future. Thank you! My fault, really sorry. You were asking on -users! Really time to get some energy. Lunch! /Olle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
Thanks it helped, I had the noload in modules.conf. But now I have another problem: When 302 response is received by asterisk it falls in to some context. according to rfc 3261 uac which receives 302 should retry the request at the address given by the contact header filed. I am not able to make the same routing decision because conditions are different. What can I do here. I have for instance such problem that my asterisk works as a gateway. When there is an external call this call is forwarded to some internal sip address. After this my sip client responses with 302 which point to his voicemail (sip uri in the contact). What can be done in such situation to make is work?? On Jan 4, 2008 1:06 PM, Johansson Olle E [EMAIL PROTECTED] wrote: 4 jan 2008 kl. 11.50 skrev Benchev: On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski wrote: Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED] sip' (cause = 66) Maybe this: Local channel Description: Local Proxy Channel Driver Syntax: Local/[EMAIL PROTECTED]/n Configuration file: none chan_local is a pseudo-channel. Use of this channel simply loops calls back into the dialplan in a different context. Useful for recursive routing. You have to enable chan_local in menuselect (1.4) and make sure it's not disabled in modules.conf. This is not a developer question, so please take this kind of questions to asterisk-users in the future. Thank you! Best regards, /Olle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
