Hi all I hope someone can help, new user to freepbx and for that matter Linux. I used the latest stable distro from freepbx to setup a conference server. this part works well. Current setup is asterisk 1.8.12 and freepbx 2.10 connected with 6 sip trunks to 6 sip stations on an Icon ECS version 9.0. inbound calls via did to ecs ring a hunt group with the 6 sip stations and all go to the conference room. inbound all work.
I installed a sip client (3cx) on a pc and it registers fine to the freepbx. I programed one trunk to ring the sip client (500) and this works also. However outbound from the sip client I get all circuits are busy. with wire shark I did a packet capture and asterisk is returning a 401 unauthorized to the sip client on outbound calls. I have no dial manipulation tables on the trunks and have an outbound rule of 21x in the outbound rule table adix.
Is there somewhere I have to enble the extension for outbound calling? any help would be appriciated.
Gene
one more item, wire shark does not show the asterisk/freepbx sending any sip requests to the ECS when I try an place the outbound call. So this makes me think the issue is somewhere in the freepbx/asterisk settings.
[2012-06-12 15:06:17] VERBOSE[3102] netsock2.c: == Using SIP RTP TOS bits 184
[2012-06-12 15:06:17] VERBOSE[3102] netsock2.c: == Using SIP RTP CoS mark 5
[2012-06-12 15:06:17] VERBOSE[5309] pbx.c: -- Executing [7216@from-internal:1] ResetCDR("SIP/500-00000087", "") in new stack
[2012-06-12 15:06:17] VERBOSE[5309] pbx.c: -- Executing [7216@from-internal:2] NoCDR("SIP/500-00000087", "") in new stack
[2012-06-12 15:06:17] VERBOSE[5309] pbx.c: -- Executing [7216@from-internal:3] Progress("SIP/500-00000087", "") in new stack
[2012-06-12 15:06:17] VERBOSE[5309] pbx.c: -- Executing [7216@from-internal:4] Wait("SIP/500-00000087", "1") in new stack
[2012-06-12 15:06:18] VERBOSE[5309] pbx.c: -- Executing [7216@from-internal:5] Progress("SIP/500-00000087", "") in new stack
[2012-06-12 15:06:18] VERBOSE[5309] pbx.c: -- Executing [7216@from-internal:6] Playback("SIP/500-00000087", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2012-06-12 15:06:18] VERBOSE[5309] file.c: -- Playing 'silence/1.ulaw' (language 'en')
[2012-06-12 15:06:19] VERBOSE[5309] file.c: -- Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
[2012-06-12 15:06:22] VERBOSE[5309] file.c: -- Playing 'check-number-dial-again.ulaw' (language 'en')
[2012-06-12 15:06:24] VERBOSE[5309] pbx.c: -- Executing [7216@from-internal:7] Wait("SIP/500-00000087", "1") in new stack
[2012-06-12 15:06:25] VERBOSE[5309] pbx.c: -- Executing [7216@from-internal:8] Congestion("SIP/500-00000087", "20") in new stack
[2012-06-12 15:06:25] WARNING[5309] channel.c: Prodding channel 'SIP/500-00000087' failed
[2012-06-12 15:06:25] VERBOSE[5309] pbx.c: == Spawn extension (from-internal, 7216, 8) exited non-zero on 'SIP/500-00000087'
[2012-06-12 15:06:25] VERBOSE[5309] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/500-00000087", "") in new stack
[2012-06-12 15:06:25] VERBOSE[5309] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/500-00000087'
You dont have a matching outbound route
I have the last sip trunk setup in outbound route named ADIX. for a dial pattern I have [ ]+[7 ]|[21x ]/[ ]. route CID is 352, trunk in route group is 352. Is there something wrong with this dial plan? all I want to beable to do is call ext 216 on the ECS PBX.
I am talking about in FreePBX, the Syntax you gave I have no idea what machine you are about.
In FrePBX 2XX will route all 3 digit extensions starting with 2 down your trunk.
Yes so am I, That is the oubound route dial pattern to match. Is there a document that would give a step by step to setting up outbound calling? I have found a lot on dial plans and dial patterns, but nothing that give a clear explanation of outbound dialing setup from start to finish.
There is not a whole lot to it. Patterns are matched top to bottom and the route has a destination.