Hi all,
I just installed the FreePBX distro to replace a Trixbox CE system since this has a newer version of Asterisk
I’m trying to get calls from my Cisco Call Manager SIP trunk to go out to the VOIP.ms trunk. The VOIP.MS trunk works as I have an ATA registered to FreePBX and I can call out no problem. Also incoming calls from VOIP.MS go to the Cisco trunk and they ring my Cisco phones no problem.
The problem is dialing outbound from Cisco phones to UCM which then goes to FreePBX and then out the Voip.MS trunk.
Here’s my trunk details for the Cisco trunk:
Peer Details:
type=peer
host=192.168.10.200
insecure=very
qualify=yes
User context: from-internal
User details:
type=friend
qualify=yes
nat=no
insecure=very
host=192.168.10.200
fromdomain=192.168.10.200
dtmf=rfc2833
canreinvite=yesHere’s the log files from what happens when I try to dial *97 from CUCM
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Executing [*97@from-trunk-sip-CUCM8:1] Set("SIP/CUCM8-0000001b", "GROUP()=OUT_3") in new stack
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Executing [*97@from-trunk-sip-CUCM8:2] Goto("SIP/CUCM8-0000001b", "from-trunk,*97,1") in new stack
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Goto (from-trunk,*97,1)
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Executing [*97@from-trunk:1] Set("SIP/CUCM8-0000001b", "__FROM_DID=*97") in new stack
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Executing [*97@from-trunk:2] NoOp("SIP/CUCM8-0000001b", "Received an unknown call with DID set to *97") in new stack
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Executing [*97@from-trunk:3] Goto("SIP/CUCM8-0000001b", "s,a2") in new stack
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Goto (from-trunk,s,2)
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Executing [s@from-trunk:2] Answer("SIP/CUCM8-0000001b", "") in new stack
[2012-10-23 18:11:19] WARNING[3417] chan_sip.c: Ignoring video media offer because port number is zero
[2012-10-23 18:11:19] VERBOSE[7324] pbx.c: -- Executing [s@from-trunk:3] Wait("SIP/CUCM8-0000001b", "2") in new stack
[2012-10-23 18:11:19] NOTICE[7324] channel.c: Dropping incompatible voice frame on SIP/CUCM8-0000001b of format ilbc since our native format has changed to 0x1000 (g722)
[2012-10-23 18:11:21] VERBOSE[7324] pbx.c: -- Executing [s@from-trunk:4] Playback("SIP/CUCM8-0000001b", "ss-noservice") in new stack
[2012-10-23 18:11:21] VERBOSE[7324] file.c: -- <SIP/CUCM8-0000001b> Playing 'ss-noservice.ulaw' (language 'en')
[2012-10-23 18:11:22] VERBOSE[7324] pbx.c: == Spawn extension (from-trunk, s, 4) exited non-zero on 'SIP/CUCM8-0000001b'
[2012-10-23 18:11:22] VERBOSE[7324] pbx.c: -- Executing [h@from-trunk:1] Macro("SIP/CUCM8-0000001b", "hangupcall,") in new stack
[2012-10-23 18:11:22] VERBOSE[7324] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/CUCM8-0000001b", "1?theend") in new stack
[2012-10-23 18:11:22] VERBOSE[7324] pbx.c: -- Goto (macro-hangupcall,s,3)
[2012-10-23 18:11:22] VERBOSE[7324] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf("SIP/CUCM8-0000001b", "0?Set(CDR(recordingfile)=)") in new stack
[2012-10-23 18:11:22] VERBOSE[7324] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("SIP/CUCM8-0000001b", "") in new stack
[2012-10-23 18:11:22] VERBOSE[7324] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/CUCM8-0000001b' in macro 'hangupcall'
[2012-10-23 18:11:22] VERBOSE[7324] pbx.c: == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/CUCM8-0000001b'
I do have *97 as Dial Pattern in my Voip.MS Outbound route.
I did some testing and made a Inbound Route with DID *97 with destination Voip.MS and that did work but I’ve never had to do that for when I had Trixbox.
Any ideas on how to get any call from CUCM to go out the VOIP.MS trunk without the Inbound Route setting?
Any help would be apprecticated.
Thanks,
Andrew