Trunk context problems

General Help
1

Hi guys!

I have installed FreePBX 2.10, asterisk version 1.8.7.

I’ve setup a SIP trunk with an Avaya wich was working fine in a previous installation with these same software settings. I had a problem with this server and had to redo it again. I can’t get the Asterisk to take calls from the Avaya system. I’ve setup the trunk with no user/password (I do not wish to add more complexity to the problem).
Here are my trunk specs:

Trunk Name: Avaya
PEER:
host=x.y.z.w
canreinvite=no
type=peer
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw
qualify=yes
context=from-internal

USER context: avaya-sun (sun is the name of my asterisk box)
host=x.y.z.w
type=user
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw
qualify=yes

If I put the PEER in from-trunk context, the Asterisk will fail saying it is an anonymous call and will play ‘ss-noservice’. If I set it to from-internal it will try to connect but will hangup after one tone. In the CLI it would say extension 8199 not found in from-internal.

I have setup extensions_custom.con, added a [from-internal] and [from-internal-custom] and [from-internal-additiona-custom] the following entry:
exten => 8199,1,Answer()
exten => 8199,n,Playback(weasels-eaten-phonesys)
exten => 8199,n,Wait(2)
exten => 8199,n,SayDigits(2)
exten => 8199,n,Hangup()

Still…, no juice… At this time I have no idea what else to do…, I’ve been working on this one for over three weeks. Never ad I spent so much time going though dialplans and everything! Going a bit crazy now…

Any help would be GREATLY appreciated.

2

Please post Asterisk log with a verbosity above 20 and place a call into the system from the Open Office.

If you can do this when no other activity is on the system.

3

Hi SkykingOH

I set verbose to 21, did a dialplan reload (just to know where the error should be -right after the reload-) and this is what I got:

[2012-04-28 22:41:36] WARNING[10052] pbx.c: Context ‘app-dialvm’ tries to include nonexistent context ‘app-dialvm-custom’
[2012-04-28 22:41:36] WARNING[10052] pbx.c: Context ‘macro-systemrecording’ tries to include nonexistent context ‘macro-systemrecording-custom’
[2012-04-28 22:41:36] WARNING[10052] pbx.c: Context ‘app-recordings’ tries to include nonexistent context ‘app-recordings-custom’
[2012-04-28 22:41:36] WARNING[10052] pbx.c: Context ‘macro-outisbusy’ tries to include nonexistent context ‘macro-outisbusy-custom’
[2012-04-28 22:41:36] WARNING[10052] pbx.c: Context ‘ext-featurecodes’ tries to include nonexistent context ‘ext-featurecodes-custom’
[2012-04-28 22:41:57] VERBOSE[26558] netsock2.c: == Using SIP RTP TOS bits 184 Here is where the call 'comes in’
[2012-04-28 22:41:57] VERBOSE[26558] netsock2.c: == Using SIP RTP CoS mark 5
[2012-04-28 22:41:57] NOTICE[26558] chan_sip.c: Call from ‘Avaya’ (10.100.100.14:5060) to extension ‘8199’ rejected because extension not found in context ‘from-internal’.
[2012-04-28 22:42:01] VERBOSE[10057] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
[2012-04-28 22:42:03] VERBOSE[10057] manager.c: == Manager ‘admin’ logged off from 127.0.0.1

As you can see there is not much info… I also did a Debug:

[2012-04-28 22:42:37] DEBUG[26561] chan_iax2.c: ip callno count decremented to 3 for 67.107.209.xxx
[2012-04-28 22:42:37] DEBUG[26568] chan_iax2.c: schedule decrement of callno used for 67.107.209.xxx in 60 seconds
[2012-04-28 22:42:37] DEBUG[26558] chan_sip.c: Auto destroying SIP dialog ‘[email protected]’
[2012-04-28 22:42:37] DEBUG[26558] chan_sip.c: Destroying SIP dialog [email protected]
[2012-04-28 22:42:38] DEBUG[26558] chan_sip.c: = Looking for Call ID: [email protected] (Checking From) --From tag as469245ae --To-tag
[2012-04-28 22:42:38] DEBUG[26558] acl.c: For destination ‘63.111.11.xxx’, our source address is ‘10.100.100.8’.
[2012-04-28 22:42:38] DEBUG[26558] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.100.100.8:5060
[2012-04-28 22:42:38] DEBUG[26558] chan_sip.c: Allocating new SIP dialog for [email protected] - OPTIONS (No RTP)
[2012-04-28 22:42:38] DEBUG[26558] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[2012-04-28 22:42:38] DEBUG[26558] chan_sip.c: Trying to put ‘SIP/2.0 200’ onto UDP socket destined for 63.111.11.xxx:5060
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: = Looking for Call ID: [email protected] (Checking From) --From tag as469245ae --To-tag
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: Ignoring SIP message because of retransmit (OPTIONS Seqno 102, ours 102)
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: Trying to put ‘SIP/2.0 200’ onto UDP socket destined for 63.111.11.xxx:5060
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: Allocating new SIP dialog for [email protected]:5060 - OPTIONS (No RTP)
[2012-04-28 22:42:39] DEBUG[26558] acl.c: For destination ‘10.100.100.14’, our source address is ‘10.100.100.8’.
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.100.100.8:5060
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: Initializing initreq for method OPTIONS - callid [email protected]:5060
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: Trying to put ‘OPTIONS sip’ onto UDP socket destined for 10.100.100.14:5060
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: = Looking for Call ID: [email protected]:5060 (Checking To) --From tag as6a2a18f5 --To-tag 34df8ce8-a64640e-13c4-55013-71eeb1-36db96be-71eeb1
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: Stopping retransmission on ‘[email protected]:5060’ of Request 102: Match Found
[2012-04-28 22:42:39] DEBUG[26558] chan_sip.c: Destroying SIP dialog [email protected]:5060
[2012-04-28 22:42:40] DEBUG[26558] chan_sip.c: = Looking for Call ID: [email protected] (Checking From) --From tag as469245ae --To-tag
[2012-04-28 22:42:40] DEBUG[26558] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[2012-04-28 22:42:40] DEBUG[26558] chan_sip.c: Ignoring SIP message because of retransmit (OPTIONS Seqno 102, ours 102)
[2012-04-28 22:42:40] DEBUG[26558] chan_sip.c: Trying to put ‘SIP/2.0 200’ onto UDP socket destined for 63.111.11.xxx:5060
[2012-04-28 22:42:41] DEBUG[26558] chan_sip.c: = Looking for Call ID: [email protected] (Checking From) --From tag as469245ae --To-tag
[2012-04-28 22:42:41] DEBUG[26558] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[2012-04-28 22:42:41] DEBUG[26558] chan_sip.c: Ignoring SIP message because of retransmit (OPTIONS Seqno 102, ours 102)
[2012-04-28 22:42:41] DEBUG[26558] chan_sip.c: Trying to put ‘SIP/2.0 200’ onto UDP socket destined for 63.111.11.xxx:5060
[2012-04-28 22:42:41] DEBUG[26558] chan_sip.c: Auto destroying SIP dialog ‘[email protected]’
[2012-04-28 22:42:41] DEBUG[26558] chan_sip.c: Destroying SIP dialog [email protected][2012-04-28 22:42:42] DEBUG[26558] chan_sip.c: = Looking for Call ID: [email protected] (Checking From) --From tag as469245ae --To-tag
[2012-04-28 22:42:42] DEBUG[26558] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[2012-04-28 22:42:42] DEBUG[26558] chan_sip.c: Ignoring SIP message because of retransmit (OPTIONS Seqno 102, ours 102)
[2012-04-28 22:42:42] DEBUG[26558] chan_sip.c: Trying to put ‘SIP/2.0 200’ onto UDP socket destined for 63.111.11.xxx:5060
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: = Looking for Call ID: 34f0a800-a64640e-13c4-55013-71eeb5-443a7861-71eeb5 (Checking From) --From tag 34df9708-a64640e-13c4-55013-71eeb5-6c15eb0f-71eeb5 --To-tag
[2012-04-28 22:42:43] DEBUG[26558] acl.c: For destination ‘10.100.100.14’, our source address is ‘10.100.100.8’.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.100.100.8:5060
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Allocating new SIP dialog for 34f0a800-a64640e-13c4-55013-71eeb5-443a7861-71eeb5 - INVITE (No RTP)
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Begin: parsing SIP “Supported: sipvc,x-nortel-sipvc,100rel,replaces”
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Found SIP option: -sipvc-
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Found no match for SIP option: sipvc (Please file bug report!)
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Found SIP option: -x-nortel-sipvc-
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Found private SIP option, not supported: x-nortel-sipvc
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Found SIP option: -100rel-
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Matched SIP option: 100rel
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Found SIP option: -replaces-
[2012-04-28 22:42:43] DEBUG[26558] sip/reqresp_parser.c: Matched SIP option: replaces
[2012-04-28 22:42:43] DEBUG[26558] netsock2.c: Splitting ‘10.100.100.14:5060’ into…
[2012-04-28 22:42:43] DEBUG[26558] netsock2.c: …host ‘10.100.100.14’ and port ‘5060’.
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0xb56142b8’
[2012-04-28 22:42:43] DEBUG[26558] res_rtp_asterisk.c: Allocated port 12990 for RTP instance ‘0xb56142b8’
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: RTP instance ‘0xb56142b8’ is setup and ready to go
[2012-04-28 22:42:43] DEBUG[26558] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0xb56142b8’
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Setting NAT on RTP to Off
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing session-level SDP o=- 1335674758 1335674758 IN IP4 10.100.100.14… UNSUPPORTED.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing session-level SDP s=-… UNSUPPORTED.
[2012-04-28 22:42:43] DEBUG[26558] netsock2.c: Splitting ‘10.100.100.14’ into…
[2012-04-28 22:42:43] DEBUG[26558] netsock2.c: …host ‘10.100.100.14’ and port ‘’.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing session-level SDP c=IN IP4 10.100.100.14… OK.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing session-level SDP t=0 0… UNSUPPORTED.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing session-level SDP a=sqn:0… UNSUPPORTED.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing session-level SDP a=cdsc:1 image udptl t38… UNSUPPORTED.
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Setting payload 0 based on m type on 0x72ce370
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Setting payload 111 based on m type on 0x72ce370
[2012-04-28 22:42:43] DEBUG[26558] netsock2.c: Splitting ‘10.100.100.14’ into…
[2012-04-28 22:42:43] DEBUG[26558] netsock2.c: …host ‘10.100.100.14’ and port ‘’.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 10.100.100.14… OK.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:120 telephone-event/8000… OK.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing media-level (audio) SDP a=fmtp:120 0-15… UNSUPPORTED.
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Unsetting payload 111 on 0x72ce370
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 X-nt-inforeq/8000… UNSUPPORTED.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Processing media-level (audio) SDP a=sendrecv… OK.
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Incorporating payload 0 on 0x72ce370
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Incorporating payload 120 on 0x72ce370
[2012-04-28 22:42:43] DEBUG[26558] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xb56142b8’
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Copying payload 0 from 0x72ce370 to 0xb5614464
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Copying payload 120 from 0x72ce370 to 0xb5614464
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: We’re settling with these formats: 0x4 (ulaw)
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Checking SIP call limits for device
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Updating call counter for incoming call
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Trying to put ‘SIP/2.0 404’ onto UDP socket destined for 10.100.100.14:5060
[2012-04-28 22:42:43] NOTICE[26558] chan_sip.c: Call from ‘Avaya’ (10.100.100.14:5060) to extension ‘8199’ rejected because extension not found in context ‘from-internal’.
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Updating call counter for incoming call
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: = Looking for Call ID: 34f0a800-a64640e-13c4-55013-71eeb5-443a7861-71eeb5 (Checking From) --From tag 34df9708-a64640e-13c4-55013-71eeb5-6c15eb0f-71eeb5 --To-tag as42d45c83
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Stopping retransmission on ‘34f0a800-a64640e-13c4-55013-71eeb5-443a7861-71eeb5’ of Response 1: Match Found
[2012-04-28 22:42:43] DEBUG[26558] chan_sip.c: Destroying SIP dialog 34f0a800-a64640e-13c4-55013-71eeb5-443a7861-71eeb5
[2012-04-28 22:42:43] DEBUG[26558] rtp_engine.c: Destroyed RTP instance ‘0xb56142b8’
[2012-04-28 22:42:46] DEBUG[26558] chan_sip.c: Auto destroying SIP dialog ‘[email protected]’
[2012-04-28 22:42:46] DEBUG[26558] chan_sip.c: Destroying SIP dialog [email protected]
[2012-04-28 22:42:47] DEBUG[26565] chan_iax2.c: ip callno count incremented to 4 for 67.107.209.xxx

My FreePBX’s IP is 10.100.100.8, Avaya is 10.100.100.14; 67.x.x.x is a service provider and so is 200.x.x.x

Thanks for your help!! :slight_smile:

4

Your answer is right here:

Do you have en extension 8199 on the FreePBX box?

If you do a dialplan show 8199@from-internal that’s a fail safe check.

5

I have added the ‘declaration’ of extension 8199 in extensions_custom.conf:

[from-internal-custom]
exten => 8199,1,Answer()
exten => 8199,n,Playback(weasels-eaten-phonesys)
exten => 8199,n,Wait(2)
exten => 8199,n,SayDigits(2)
exten => 8199,n,Hangup()

When I do a ‘dialplan show 8199@from-internal’ the results are:
Connected to Asterisk 1.8.7.0 currently running on Sun (pid = 3753)
Sun*CLI> dialplan show 8199@from-internal
[ Included context ‘from-internal-additional-custom’ created by ‘pbx_config’ ]
‘8199’ => 1. Answer() [pbx_config]
2. Playback(weasels-eaten-phonesys) [pbx_config]
3. Wait(2) [pbx_config]
4. SayDigits(2) [pbx_config]
5. Hangup() [pbx_config]

[ Included context ‘outrt-7’ created by ‘pbx_config’ ]
’_[589]XXX’ => 1. Macro(user-callerid,LIMIT,) [pbx_config]
2. Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})}) [pbx_config]
3. Set(_NODEST=) [pbx_config]
4. Gosub(sub-record-check,s,1(out,${EXTEN},)) [pbx_config]
5. Macro(dialout-trunk,11,${EXTEN},) [pbx_config]
6. Macro(outisbusy,) [pbx_config]

-= 2 extensions (11 priorities) in 2 contexts. =-

Which pretty much does show the extension there, but it is still not woriking… Would it be that there is a problem with some module or something…, perhpas a faulty installation…?

6

There is no problem with any module, you aren’t even using FreePBX modules.

I would drop back a bit, create an extension in FreePBX and make sure that you can reach it from the trunk.

This is odd, and is one of those problems that will be very difficult to troubleshoot from the forums (the back and forth thing).

If you want to pop an hour of FreePBX support and request me (my real name is Scott) I would be glad to work on it with you. I am actually stuck loading OS’s on computers this afternoon until about 7PM EST if you are interested.

7

Hi SkykingOH.

Sorry for the delay of my answer… I’m actually a consultant for another company and they don’t really like the idea of providing access to a third party to the system.

I re-installed the server with FreePBX Distro. I re-tested again and entered the extension in the from-internal-custom, still no juice. A peer debug shows:

– Executing [8199;phone-context=Unknown@from-internal:1] ResetCDR(“SIP/Avaya-0000011c”, “”) in new stack
– Executing [8199;phone-context=Unknown@from-internal:2] NoCDR(“SIP/Avaya-0000011c”, “”) in new stack
– Executing [8199;phone-context=Unknown@from-internal:3] Progress(“SIP/Avaya-0000011c”, “”) in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 10.100.100.14:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.100.100.14:5060;branch=z9hG4bK-774b1f-d1fd7338-4a35e6b;received=10.100.100.14
From: sip:88402270;[email protected];user=phone;tag=34c77f38-a64640e-13c4-55013-774b1f-770c422a-774b1f
To: sip:8199;phone-context=Unknown@sun;user=phone;tag=as1b8980fa
Call-ID: 34f28e50-a64640e-13c4-55013-774b1f-8d32cc4-774b1f
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:8199;[email protected]:5060
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 1825158057 1825158057 IN IP4 10.100.100.8
s=Asterisk PBX 1.8.7.1
c=IN IP4 10.100.100.8
t=0 0
m=audio 10930 RTP/AVP 0 120
a=rtpmap:0 PCMU/8000
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-16
a=ptime:20
a=sendrecv

<------------>
– Executing [8199;phone-context=Unknown@from-internal:4] Wait(“SIP/Avaya-0000011c”, “1”) in new stack
– Executing [8199;phone-context=Unknown@from-internal:5] Progress(“SIP/Avaya-0000011c”, “”) in new stack
– Executing [8199;phone-context=Unknown@from-internal:6] Playback(“SIP/Avaya-0000011c”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/Avaya-0000011c> Playing ‘silence/1.ulaw’ (language ‘en’)
– <SIP/Avaya-0000011c> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
– <SIP/Avaya-0000011c> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
– Executing [8199;phone-context=Unknown@from-internal:7] Wait(“SIP/Avaya-0000011c”, “1”) in new stack
– Executing [8199;phone-context=Unknown@from-internal:8] Congestion(“SIP/Avaya-0000011c”, “20”) in new stack

<— Reliably Transmitting (no NAT) to 10.100.100.14:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.100.100.14:5060;branch=z9hG4bK-774b1f-d1fd7338-4a35e6b;received=10.100.100.14
From: sip:88402270;[email protected];user=phone;tag=34c77f38-a64640e-13c4-55013-774b1f-770c422a-774b1f
To: sip:8199;phone-context=Unknown@sun;user=phone;tag=as1b8980fa
Call-ID: 34f28e50-a64640e-13c4-55013-774b1f-8d32cc4-774b1f
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.7.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[2012-05-03 00:21:38] WARNING[10307]: channel.c:4641 ast_prod: Prodding channel ‘SIP/Avaya-0000011c’ failed
== Spawn extension (from-internal, 8199;phone-context=Unknown, 8) exited non-zero on ‘SIP/Avaya-0000011c’
– Executing [h@from-internal:1] Hangup(“SIP/Avaya-0000011c”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/Avaya-0000011c’

<— SIP read from UDP:10.100.100.14:5060 —>
ACK sip:8199;phone-context=Unknown@sun;maddr=10.100.100.8;transport=udp;user=phone SIP/2.0
From: sip:88402270;[email protected];user=phone;tag=34c77f38-a64640e-13c4-55013-774b1f-770c422a-774b1f
To: sip:8199;phone-context=Unknown@sun;user=phone;tag=as1b8980fa
Call-ID: 34f28e50-a64640e-13c4-55013-774b1f-8d32cc4-774b1f
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.100.100.14:5060;branch=z9hG4bK-774b1f-d1fd7338-4a35e6b
Max-Forwards: 70
User-Agent: Nortel Networks BCM VoIP Gateway release_46 version_46.46.0.33
Supported: sipvc,x-nortel-sipvc,100rel,replaces
x-nt-corr-id: 34f28e50-a64640e-13c4-55013-774b1f-8d32cc4-774b1f
Contact: sip:88402270;[email protected]:5060;maddr=10.100.100.14;transport=udp;user=phone
Allow: INVITE,INFO,ACK,OPTIONS,CANCEL,BYE,NOTIFY,PRACK,UPDATE,REFER
Content-Length: 0

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— (13 headers 0 lines) —
Really destroying SIP dialog ‘34f28e50-a64640e-13c4-55013-774b1f-8d32cc4-774b1f’ Method: ACK

Interesting thing here is that I don’t see it actually dialing 8199, it just plays “you call cannot be completed as dialed…”

Any ideas…?

Thank you soooo much!

Cheers!

8

It looks like it is not matching the trunk so it is ignoring the context declarative.

If you are the consultant I assume you are getting paid? You going to split the money with me when I get this working?

9

Thanks for the help SkykingOH. I appreciate it.