[asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M -followup
with log
Jerry Geis
geisj at pagestation.com
Thu Apr 19 12:36:08 MST 2007
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
--
Jerry Geis
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677
(317)663-0808 Fax
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