Hello I’m wondering if someone can help.
I use a BCM50, and I’m looking to create a SIP trunk between this an a SIP PBX. ( I was looking and using trixboxce to start with, but found out no more development on that system)
So I started to look at freePBX. (I’m new to the world of SIP, coming from a Nortel background)
I can get a SIP tunnel up fine and from a handset on the freePBX I can call a handset on the BCM50.
The other way round doesn’t work, I get ‘your call cannot be completed as dialled, please check the number and dial again’
Using the same settings Trixboxce to the BCM50 calls work fine both ways.
I have also tested Trixboxce to freePBX again calls route fine.
It just seems the freePBX is not routing the call to the required handset.
I’ve attached the log below:
[2012-07-02 10:27:51] VERBOSE[2398] netsock2.c: == Using SIP RTP TOS bits 184
[2012-07-02 10:27:51] VERBOSE[2398] netsock2.c: == Using SIP RTP CoS mark 5
[2012-07-02 10:27:51] VERBOSE[3994] pbx.c: – Executing [1001;phone-context=subscriber.private@from-internal:1] ResetCDR(“SIP/BCM50_Out-0000001c”, “”) in new stack
[2012-07-02 10:27:51] VERBOSE[3994] pbx.c: – Executing [1001;phone-context=subscriber.private@from-internal:2] NoCDR(“SIP/BCM50_Out-0000001c”, “”) in new stack
[2012-07-02 10:27:51] VERBOSE[3994] pbx.c: – Executing [1001;phone-context=subscriber.private@from-internal:3] Progress(“SIP/BCM50_Out-0000001c”, “”) in new stack
[2012-07-02 10:27:51] VERBOSE[3994] pbx.c: – Executing [1001;phone-context=subscriber.private@from-internal:4] Wait(“SIP/BCM50_Out-0000001c”, “1”) in new stack
[2012-07-02 10:27:52] VERBOSE[3994] pbx.c: – Executing [1001;phone-context=subscriber.private@from-internal:5] Progress(“SIP/BCM50_Out-0000001c”, “”) in new stack
[2012-07-02 10:27:52] VERBOSE[3994] pbx.c: – Executing [1001;phone-context=subscriber.private@from-internal:6] Playback(“SIP/BCM50_Out-0000001c”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2012-07-02 10:27:52] VERBOSE[3994] file.c: – <SIP/BCM50_Out-0000001c> Playing ‘silence/1.ulaw’ (language ‘en’)
[2012-07-02 10:27:53] VERBOSE[3994] file.c: – <SIP/BCM50_Out-0000001c> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2012-07-02 10:27:56] VERBOSE[3994] file.c: – <SIP/BCM50_Out-0000001c> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[2012-07-02 10:27:58] VERBOSE[3994] pbx.c: – Executing [1001;phone-context=subscriber.private@from-internal:7] Wait(“SIP/BCM50_Out-0000001c”, “1”) in new stack
[2012-07-02 10:27:59] VERBOSE[3994] pbx.c: – Executing [1001;phone-context=subscriber.private@from-internal:8] Congestion(“SIP/BCM50_Out-0000001c”, “20”) in new stack
[2012-07-02 10:27:59] WARNING[3994] channel.c: Prodding channel ‘SIP/BCM50_Out-0000001c’ failed
[2012-07-02 10:27:59] VERBOSE[3994] pbx.c: == Spawn extension (from-internal, 1001;phone-context=subscriber.private, 8) exited non-zero on ‘SIP/BCM50_Out-0000001c’
[2012-07-02 10:27:59] VERBOSE[3994] pbx.c: – Executing [h@from-internal:1] Hangup(“SIP/BCM50_Out-0000001c”, “”) in new stack
[2012-07-02 10:27:59] VERBOSE[3994] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/BCM50_Out-0000001c’
[2012-07-02 10:28:05] WARNING[2398] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6367ms with no response
Thanks
Pat