Jerry Geis
2007-Apr-19 12:36 UTC
[asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M -followup with log
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@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
Connected 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@161.49.142.250>;tag=as2cc96e52
To: <sip:192.168.45.129>
Contact: <sip:asterisk@161.49.142.250>
Call-ID: 3ee92dbe77f51a1748f736be4593719d@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@161.49.142.250>;tag=as2cc96e52
To: <sip:192.168.45.129>;tag=812da8c0-13c4-46277c06-279cd106-42ff
Call-ID: 3ee92dbe77f51a1748f736be4593719d@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@161.49.142.250' Method: OPTIONS
?
[Khfemsrv*CLI>
-- Attempting call on SIP/QuadNortel/7113 for
smvoice_callprogress@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@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>
Contact: <sip:0@161.49.142.250>
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@qg.com;user=phone>
Privacy: none
Contact:
<sip:7113;phone-context=cdp.udp@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@qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone>
?set_destination: Parsing
<sip:7113;phone-context=cdp.udp@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@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Contact: <sip:0@161.49.142.250>
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@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@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@161.49.142.250' in 6400 ms (Method:
INVITE)
? > Channel SIP/QuadNortel-09a4c0e0 was answered.
?
[Khfemsrv*CLI>
-- Executing [smvoice_callprogress@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@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@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@161.49.142.250' in 6400 ms (Method:
INVITE)
?
[Khfemsrv*CLI> quit
set_destination: Parsing
<sip:7113;phone-context=cdp.udp@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@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@161.49.142.250>;tag=as4e5a553d
To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a
Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@161.49.142.250'. Giving up.
?
[Khfemsrv*CLI> quit
Really destroying SIP dialog
'1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250' Method: INVITE
?
[Khfemsrv*CLI> quit
Executing last minute cleanups
Asterisk cleanly ending (0).
--
Jerry Geis
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677
(317)663-0808 Fax
Leo Ann Boon
2007-Apr-20 16:17 UTC
[asterisk-users] Asterisk 1.4.2 connection to Nortel CS1000M -followup with log
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@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 > Connected 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@161.49.142.250>;tag=as2cc96e52 > To: <sip:192.168.45.129> > Contact: <sip:asterisk@161.49.142.250> > Call-ID: 3ee92dbe77f51a1748f736be4593719d@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@161.49.142.250>;tag=as2cc96e52 > To: <sip:192.168.45.129>;tag=812da8c0-13c4-46277c06-279cd106-42ff > Call-ID: 3ee92dbe77f51a1748f736be4593719d@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@161.49.142.250' Method: OPTIONS > ? > [Khfemsrv*CLI> -- Attempting call on SIP/QuadNortel/7113 for > smvoice_callprogress@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@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129> > Contact: <sip:0@161.49.142.250> > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@qg.com;user=phone> > Privacy: none > Contact: > <sip:7113;phone-context=cdp.udp@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@qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> > > ?set_destination: Parsing > <sip:7113;phone-context=cdp.udp@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@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a > Contact: <sip:0@161.49.142.250> > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@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@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@161.49.142.250' in 6400 ms (Method: > INVITE) > ? > Channel SIP/QuadNortel-09a4c0e0 was answered. > ? > [Khfemsrv*CLI> -- Executing > [smvoice_callprogress@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@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@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@161.49.142.250' in 6400 ms (Method: > INVITE) > ? > [Khfemsrv*CLI> quit > set_destination: Parsing > <sip:7113;phone-context=cdp.udp@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@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@161.49.142.250>;tag=as4e5a553d > To: <sip:7113@192.168.45.129>;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a > Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@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@161.49.142.250'. Giving up. > ? > [Khfemsrv*CLI> quit > Really destroying SIP dialog > '1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250' Method: INVITE > ? > [Khfemsrv*CLI> quit > Executing last minute cleanups > Asterisk cleanly ending (0). > >