[asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M -followup with log

Fri Apr 20 16:17:38 MST 2007

Just curios, does the CS1000 now support RFC2833? Previously, I know the 
NRS can only support SIP-INFO.

Leo

Jerry Geis wrote:
> Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming 
> calls just fine. However, using outgoing call files the CS1000 is 
> hanging up after I answer the call.
>
> I dont know why?
>
> Thanks, for any assistance.
>
> Jerry
>
> my sip.conf entry is:
>        [Nortel]
>        type=friend
>        dtmfmode=rfc2833
>        username=XXXXXXXXX
>        disallow=all
>        allow=ulaw
>        allow=alaw
>        context=nortel
>        host=XXXXXXXXXXX
>        canreinvite=yes
>        qualify=yes
>     usereqphone=yes
>
>
> ---------------------------------
>
> Use 'exit' when done
>
> Asterisk 1.4.2, Copyright (C) 1999 - 2006 Digium, Inc. and others.
> Created by Mark Spencer <markster at digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' 
> for details.
> This is free software, with components licensed under the GNU General 
> Public
> License version 2 and other licenses; you are welcome to redistribute 
> it under
> certain conditions. Type 'core show license' for details.
> =========================================================================
>  == Parsing '/etc/asterisk/asterisk.conf': Found
> [0;37;40m[1;30;40m  == [0;37;40mParsing 
> '/etc/asterisk/extconfig.conf': Found
> [0mConnected to Asterisk 1.4.2 currently running on hfemsrv (pid = 
> 18420)
> hfemsrv*CLI> Verbosity is at least 5
>
> [Khfemsrv*CLI> sip debug
> hfemsrv*CLI> SIP Debugging enabled
> The 'sip debug' command is deprecated and will be removed in a future 
> release. Please use 'sip set debug' instead.
>
> [Khfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060:
> OPTIONS sip:192.168.45.129 SIP/2.0
> Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK2508d83c;rport
> From: "asterisk" <sip:asterisk at 161.49.142.250>;tag=as2cc96e52
> To: <sip:192.168.45.129>
> Contact: <sip:asterisk at 161.49.142.250>
> Call-ID: 3ee92dbe77f51a1748f736be4593719d at 161.49.142.250
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Thu, 19 Apr 2007 19:25:53 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> ?
> [Khfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
> SIP/2.0 200 OK
> From: "asterisk"<sip:asterisk at 161.49.142.250>;tag=as2cc96e52
> To: <sip:192.168.45.129>;tag=812da8c0-13c4-46277c06-279cd106-42ff
> Call-ID: 3ee92dbe77f51a1748f736be4593719d at 161.49.142.250
> CSeq: 102 OPTIONS
> Allow: 
> INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE 
>
> Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK2508d83c
> Supported: 100rel,sipvc,replaces
> User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
> Content-Length: 0
>
>
> <------------->
> ?--- (10 headers 0 lines) ---
> ?
> [Khfemsrv*CLI> Really destroying SIP dialog 
> '3ee92dbe77f51a1748f736be4593719d at 161.49.142.250' Method: OPTIONS
> ?
> [Khfemsrv*CLI>    -- Attempting call on SIP/QuadNortel/7113 for 
> smvoice_callprogress at smvoice-dialout:1 (Retry 1)
> ?
> [Khfemsrv*CLI> Audio is at 161.49.142.250 port 10000
> ?
> [Khfemsrv*CLI> Adding codec 0x4 (ulaw) to SDP
> ?Adding codec 0x8 (alaw) to SDP
> ?
> [Khfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060:
> INVITE sip:7113 at 192.168.45.129 SIP/2.0
> Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK11268a7d;rport
> From: "Admin System 34" <sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>
> Contact: <sip:0 at 161.49.142.250>
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Thu, 19 Apr 2007 19:25:58 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 212
>
> v=0
> o=root 18420 18420 IN IP4 161.49.142.250
> s=session
> c=IN IP4 161.49.142.250
> t=0 0
> m=audio 10000 RTP/AVP 0 8
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> ?
> [Khfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
> SIP/2.0 100 Trying
> From: "Admin System 34"<sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 102 INVITE
> Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
> Supported: 100rel,sipvc,replaces
> User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
> Contact: <sip:7113 at 192.168.45.129>
> Allow: 
> INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE 
>
> Content-Length: 0
>
>
> <------------->
> ?--- (11 headers 0 lines) ---
> ?
> [Khfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
> SIP/2.0 180 Ringing
> From: "Admin System 34"<sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 102 INVITE
> Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
> Supported: 100rel,sipvc,replaces
> User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
> Contact: 
> <sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> 
>
> Allow: 
> INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE 
>
> Content-Length: 0
>
>
> <------------->
> ?--- (11 headers 0 lines) ---
> ?
> [Khfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
> SIP/2.0 200 OK
> From: "Admin System 34"<sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 102 INVITE
> Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d
> Supported: 100rel,sipvc,replaces
> User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
> P-Asserted-Identity: <sip:7113;phone-context=cdp.udp at qg.com;user=phone>
> Privacy: none
> Contact: 
> <sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> 
>
> Allow: 
> INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE 
>
> Content-Type: application/SDP
> Content-Length: 137
>
> v=0
> o=- 91 1 IN IP4 192.168.45.129
> s=-
> t=0 0
> m=audio 5260 RTP/AVP 0
> c=IN IP4 192.168.45.199
> a=ptime:20
> a=maxptime:20
> a=sendrecv
>
> <------------->
> ?--- (14 headers 9 lines) ---
> ?
> [Khfemsrv*CLI> Found RTP audio format 0
> ?Peer audio RTP is at port 192.168.45.199:5260
> ?Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
> (nothing), combined - 0x4 (ulaw)
> ?Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 
> (nothing), combined - 0x0 (nothing)
> ?Peer audio RTP is at port 192.168.45.199:5260
> ?
> [Khfemsrv*CLI> [Apr 19 14:26:03] WARNING[18442]: chan_sip.c:7724 
> set_address_from_contact: ?Invalid host name in Contact: (can't 
> resolve in DNS) : '7113'
> ?list_route: hop: 
> <sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> 
>
> ?set_destination: Parsing 
> <sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> 
> for address/port to send to
> ?set_destination: set destination to 192.168.45.129, port 5060
> ?Transmitting (no NAT) to 192.168.45.129:5060:
> ACK 
> sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone 
> SIP/2.0
> Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK4245bbdd;rport
> From: "Admin System 34" <sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
> Contact: <sip:0 at 161.49.142.250>
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
> ?set_destination: Parsing 
> <sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> 
> for address/port to send to
> ?set_destination: set destination to 192.168.45.129, port 5060
> ?
> [Khfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060:
> BYE 
> sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone 
> SIP/2.0
> Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK1be97cfa;rport
> From: "Admin System 34" <sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> X-Asterisk-HangupCause: Normal Clearing
> X-Asterisk-HangupCauseCode: 16
> Content-Length: 0
>
>
> ---
> ?Scheduling destruction of SIP dialog 
> '1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250' in 6400 ms (Method: 
> INVITE)
> ?       > Channel SIP/QuadNortel-09a4c0e0 was answered.
> ?
> [Khfemsrv*CLI>    -- Executing 
> [smvoice_callprogress at smvoice-dialout:1] 
> GotoIf("SIP/QuadNortel-09a4c0e0", 
> "1?smvoice_callprogress|3:smvoice_callprogress|2") in new stack
> ?    -- Goto (smvoice-dialout,smvoice_callprogress,3)
> ?    -- Executing [smvoice_callprogress at smvoice-dialout:3] 
> AGI("SIP/QuadNortel-09a4c0e0", "smvoice|-digium_asterisk") in new stack
> ?    -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice
> ?
> [Khfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 --->
> SIP/2.0 200 OK
> From: "Admin System 34"<sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 103 BYE
> Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK1be97cfa
> Supported: 100rel,sipvc,replaces
> User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55
> Allow: 
> INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE 
>
> Content-Length: 0
>
>
> <------------->
> ?--- (10 headers 0 lines) ---
> ?
> [Khfemsrv*CLI>    -- Playing '/tmp/smvoice.19294_0' 
> (escape_digits=0123456789#) (sample_offset 0)
> ?
> [Khfemsrv*CLI> quit
>    -- Playing '/tmp/smvoice.19294_0' (escape_digits=0123456789#) 
> (sample_offset 0)
> ?
> [Khfemsrv*CLI> quit
> [Apr 19 14:26:09] WARNING[18442]: chan_sip.c:2013 __sip_autodestruct: 
> ?Autodestruct on dialog 
> '1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250' with owner in place 
> (Method: INVITE)
> ?
> [Khfemsrv*CLI> quit
>  == Spawn extension (smvoice-dialout, smvoice_callprogress, 3) exited 
> non-zero on 'SIP/QuadNortel-09a4c0e0'
> ?
> [Khfemsrv*CLI> quit
> Scheduling destruction of SIP dialog 
> '1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250' in 6400 ms (Method: 
> INVITE)
> ?
> [Khfemsrv*CLI> quit
> set_destination: Parsing 
> <sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> 
> for address/port to send to
> ?
> [Khfemsrv*CLI> quit
> set_destination: set destination to 192.168.45.129, port 5060
> ?
> [Khfemsrv*CLI> quit
> Reliably Transmitting (no NAT) to 192.168.45.129:5060:
> BYE 
> sip:7113;phone-context=cdp.udp at qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone 
> SIP/2.0
> Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK4ede3adc;rport
> From: "Admin System 34" <sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 104 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
> ?
> [Khfemsrv*CLI> quit
> [Apr 19 14:26:09] NOTICE[19253]: pbx_spool.c:351 attempt_thread: ?Call 
> completed to SIP/QuadNortel/7113
> ?
> [Khfemsrv*CLI> quit
> <--- SIP read from 192.168.45.129:5060 --->
> SIP/2.0 481 Call Leg/Transaction Does Not Exist
> From: "Admin System 34"<sip:0 at 161.49.142.250>;tag=as4e5a553d
> To: <sip:7113 at 192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
> Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250
> CSeq: 104 BYE
> Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK4ede3adc
> Supported: 100rel,sipvc,replaces
> Content-Length: 0
>
>
> <------------->
> ?
> [Khfemsrv*CLI> quit
> --- (8 headers 0 lines) ---
> ?
> [Khfemsrv*CLI> quit
> [Apr 19 14:26:09] WARNING[18442]: chan_sip.c:12311 handle_response: 
> ?Remote host can't match request BYE to call 
> '1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250'. Giving up.
> ?
> [Khfemsrv*CLI> quit
> Really destroying SIP dialog 
> '1a17a38a73f5418d5b23c2ab2c2623a8 at 161.49.142.250' Method: INVITE
> ?
> [Khfemsrv*CLI> quit
> Executing last minute cleanups
> Asterisk cleanly ending (0).
> [0m
>