[Asterisk-Dev] SIP Extrange Problem
Sergio Serrano Revuelto
sergio.serrano at avanzada7.com
Fri Feb 27 03:18:09 MST 2004
Hi again,
I attach more information about this problem and I pasted new
/var/log/asterisk/debug . I have changed return -2 in rtp.c like said to
me Lawrence.
Thanks
Srsergio
--Feb 27 10:20:08 DEBUG[6151]: Setting NAT on RTP to -1
Feb 27 10:20:09 DEBUG[6151]: Stopping retransmission on
'e79dcda14db3d67e at 192.168.1.160' of Response 14698: Found
Feb 27 10:20:09 DEBUG[6151]: Setting NAT on RTP to -1
Feb 27 10:20:09 DEBUG[6151]: Check for res for 705
Feb 27 10:20:09 DEBUG[6151]: Call from user '705' is 1 out of 0
Feb 27 10:20:09 DEBUG[6151]: build_route: Contact hop:
<sip:705 at 192.168.1.160>
Feb 27 10:20:09 DEBUG[36882]: SIMPLE DIAL (NO URL)
Feb 27 10:20:09 DEBUG[36882]: Setting NAT on RTP to 0
Feb 27 10:20:09 DEBUG[36882]: Outgoing Call for 707
Feb 27 10:20:09 DEBUG[36882]: Call from user '707' is 1 out of 0
Feb 27 10:20:09 DEBUG[36882]: Prodding channel 'SIP/705-8d0f'
Feb 27 10:20:09 DEBUG[6151]: (Provisional) Stopping retransmission (but
retaining packet) on '75499f6e2157af091c2149502fe17a66 at 192.168.0.207'
Request 102: Found
Feb 27 10:20:09 DEBUG[6151]: (Provisional) Stopping retransmission (but
retaining packet) on '75499f6e2157af091c2149502fe17a66 at 192.168.0.207'
Request 102: Found
Feb 27 10:20:09 DEBUG[36882]: RTP NAT: Using address 82.223.12.13:5004
Feb 27 10:20:10 DEBUG[6151]: Stopping retransmission on
'1458dbe26ab870562ac896293899ee4d at 217.11.115.168' of Request 102: Found
Feb 27 10:20:14 DEBUG[6151]: Stopping retransmission on
'6e78575a52619bd4223386d15d41fceb at 192.168.0.207' of Request 102: Found
Feb 27 10:20:17 DEBUG[6151]: Acked pending invite 102
Feb 27 10:20:17 DEBUG[6151]: Stopping retransmission on
'75499f6e2157af091c2149502fe17a66 at 192.168.0.207' of Request 102: Found
Feb 27 10:20:17 DEBUG[6151]: build_route: Contact hop:
<sip:707 at 192.168.0.157:52478>
Feb 27 10:20:17 DEBUG[6151]: Stopping retransmission on
'e79dcda14db3d67e at 192.168.1.160' of Response 14699: Found
Feb 27 10:20:17 DEBUG[36882]: Ooh, format changed from UNKN to G729A
Feb 27 10:20:17 DEBUG[36882]: Ooh, format changed from UNKN to G729A
Feb 27 10:20:31 DEBUG[6151]: Stopping retransmission on
'585039651ce045a940a6a47a223a3a1a at 192.168.0.207' of Request 102: Found
Feb 27 10:20:36 DEBUG[36882]: Difference is 688, ms is 106
Feb 27 10:21:03 DEBUG[6151]: Stopping retransmission on
'76aa3b427a14b6fe59d5c5f14aefafb5 at 192.168.0.207' of Request 102: Found
Feb 27 10:21:03 DEBUG[6151]: Destroying call
'76aa3b427a14b6fe59d5c5f14aefafb5 at 192.168.0.207'
Feb 27 10:21:04 DEBUG[6151]: Stopping retransmission on
'6c68d95548e78cbf78d6a4866538ea75 at 192.168.0.207' of Request 102: Found
Feb 27 10:21:04 DEBUG[6151]: Destroying call
'6c68d95548e78cbf78d6a4866538ea75 at 192.168.0.207'
Feb 27 10:21:05 DEBUG[6151]: Stopping retransmission on
'76544c7e730f305d4b8a14e766a12ad8 at 192.168.0.207' of Request 102: Found
Feb 27 10:21:05 DEBUG[6151]: Destroying call
'76544c7e730f305d4b8a14e766a12ad8 at 192.168.0.207'
Feb 27 10:21:06 DEBUG[6151]: Stopping retransmission on
'75499f6e2157af091c2149502fe17a66 at 192.168.0.207' of Response 31669:
Found
Feb 27 10:21:11 DEBUG[36882]: Didn't get a frame from channel:
SIP/705-8d0f
Feb 27 10:21:11 DEBUG[36882]: Bridge stops bridging channels
SIP/705-8d0f and SIP/707-b886
Feb 27 10:21:11 DEBUG[36882]: find_user(707) - decrement inUse counter
Feb 27 10:21:11 DEBUG[36882]: cdr_mysql: inserting a CDR record.
Feb 27 10:21:11 DEBUG[36882]: cdr_mysql: SQL command as follows: INSERT
INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,l
astdata,duration,billsec,disposition,amaflags,accountcode) VALUES
('2004-02-27 10:21:11','\"705\" <705>','705','707','voip-h323',
'SIP/705-8d0
f','SIP/707-b886','Dial','SIP/707|30|tTrm',62,54,'ANSWERED',3,'')
Feb 27 10:21:11 DEBUG[36882]: find_user(705) - decrement inUse counter
Feb 27 10:21:11 DEBUG[6151]: Stopping retransmission on
'75499f6e2157af091c2149502fe17a66 at 192.168.0.207' of Request 103: Found
Feb 27 10:21:11 DEBUG[6151]: Destroying call
'75499f6e2157af091c2149502fe17a66 at 192.168.0.207'
Feb 27 10:21:11 DEBUG[6151]: Destroying call
'e79dcda14db3d67e at 192.168.1.160'
Feb 27 10:21:12 DEBUG[6151]: Stopping retransmission on
'0a82ec3c0f71d234651ab02f29764caf at 217.11.115.168' of Request 102: Found
Feb 27 10:21:12 DEBUG[6151]: Destroying call
'0a82ec3c0f71d234651ab02f29764caf at 217.11.115.168'
Feb 27 10:21:14 DEBUG[6151]: Stopping retransmission on
'2959a92225d797da41e8cdbb29ad6b9a at 192.168.0.207' of Request 102: Found
Feb 27 10:21:14 DEBUG[6151]: Destroying call
'2959a92225d797da41e8cdbb29ad6b9a at 192.168.0.207'
Feb 27 10:21:20 DEBUG[6151]: Scheduled a timeout # 22572
Feb 27 10:21:20 DEBUG[6151]: Stopping retransmission on
'40026379660cb07b4c20c4d81f37e3c0 at 192.168.0.207' of Request 148: Found
Feb 27 10:21:20 DEBUG[6151]: Stopping retransmission on
'40026379660cb07b4c20c4d81f37e3c0 at 192.168.0.207' of Request 149: Found
-----Mensaje original-----
De: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] En nombre de Sergio Serrano
Revuelto
Enviado el: viernes, 27 de febrero de 2004 10:10
Para: asterisk-dev at lists.digium.com
Asunto: RE: [Asterisk-Dev] SIP Extrange Problem
I try sip debug, but nothing happen.
Extension 705 is behind a NAT. Extension 707 is in Intranet.
Problem is always with 707 Extension.
I attach /var/log/asterisk/debug and messages
Feb 27 09:46:23 DEBUG[6151]: Call from user '705' is 1 out of 0 Feb 27
09:46:23 DEBUG[6151]: build_route: Contact hop: <sip:705 at 192.168.1.160>
Feb 27 09:46:23 DEBUG[24595]: SIMPLE DIAL (NO URL) Feb 27 09:46:23
DEBUG[24595]: Setting NAT on RTP to 0 Feb 27 09:46:23 DEBUG[24595]:
Outgoing Call for 707 Feb 27 09:46:23 DEBUG[24595]: Call from user '707'
is 1 out of 0 Feb 27 09:46:23 DEBUG[24595]: Prodding channel
'SIP/705-083d' Feb 27 09:46:23 DEBUG[6151]: (Provisional) Stopping
retransmission (but retaining packet) on
'186e62c321bde78f11ee6f385778061f at 192.168.0.207'
Request 102: Found
Feb 27 09:46:23 DEBUG[6151]: (Provisional) Stopping retransmission (but
retaining packet) on '186e62c321bde78f11ee6f385778061f at 192.168.0.207'
Request 102: Found
Feb 27 09:46:23 DEBUG[24595]: RTP NAT: Using address 82.223.7.185:5004
Feb 27 09:46:26 DEBUG[6151]: Acked pending invite 102 Feb 27 09:46:26
DEBUG[6151]: Stopping retransmission on
'186e62c321bde78f11ee6f385778061f at 192.168.0.207' of Request 102: Found
Feb 27 09:46:26 DEBUG[6151]: build_route: Contact hop:
<sip:707 at 192.168.0.157:52478> Feb 27 09:46:26 DEBUG[6151]: Stopping
retransmission on '39d77c53f34ee760 at 192.168.1.160' of Response 28042:
Found Feb 27 09:46:26 DEBUG[24595]: Ooh, format changed from UNKN to
G729A Feb 27 09:46:26 DEBUG[24595]: Ooh, format changed from UNKN to
G729A Feb 27 09:46:50 DEBUG[24595]: Didn't get a frame from channel:
SIP/705-083d Feb 27 09:46:50 DEBUG[24595]: Bridge stops bridging
channels SIP/705-083d and SIP/707-ba16 Feb 27 09:46:50 DEBUG[24595]:
find_user(707) - decrement outUse counter Feb 27 09:46:50 DEBUG[24595]:
cdr_mysql: inserting a CDR record. Feb 27 09:46:50 DEBUG[24595]:
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode) VALUES ('2004-02-27
09:46:50','\"705\" <705>','705','707','voip-h323',
'SIP/705-083d','SIP/707-ba16','Dial','SIP/707|30|tTrm',27,24,'ANSWERED',
3,'')
Feb 27 09:46:50 DEBUG[24595]: find_user(705) - decrement inUse counter
Any idea?
srsergio
-----Mensaje original-----
De: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] En nombre de Conroy,
Lawrence (SMTP) Enviado el: viernes, 27 de febrero de 2004 1:02
Para: asterisk-dev at lists.digium.com
Asunto: Re: [Asterisk-Dev] SIP Extrange Problem
Hi Sergio, Folks,
"after a few seconds" is interesting.
I wonder if this is a re-INVITE issue?
First, switch on "sip debug" in the asterisk console, and
see if there are sip messages approximately when the audio dies.
One quick way to *TEST* this is to:
* set the phones in sip.conf to canreinvite=no,
AND
* edit rtp.c to always return -2, (a couple of lines before the
existing conditional -2 return).
That way, * should always stay in the call and should NOT try to drop
out by sending a re-INVITE to the
phones after the initial call setup.
Then retry the calls to see if the problem has "gone away".
If so, then the re-INVITE issue has bitten you. It can cause strange
effects just like you're seeing (especially with some kinds of NAT/ALG
between the phones). The problem is in the NAT/ALG, but it's triggered
by the re-INVITE behaviour of *.
all the best,
Lawrence
On 26 Feb 2004, at 10:07 am, Sergio Serrano Revuelto wrote:
> Hi all,
> For a few days we have a very extrange problem. We have an
> intranet with Budgetone and others SIP Phones. In the extranet We
> Have Budgetone Phones. The whole system was working well between the
> extranet and the intranet until a few days ago.
> When we try to speak with a Budgetone of the intranet, we can speak
> during a few seconds but after a time the audio is cut in the sense of
> intranet-extranet.
> The problem is not only it, but if a budgetone of the intranet speaks
> with another phone of the intranet the same thing happens. After a
> time of conversation the audio is cut in the sense of the budgetone to
> another phone. I see the next meesage in debug:
>
> Feb 26 10:50:04 DEBUG[50193]: Didn't get a frame from channel:
> SIP/707-996a I have checked the files of configuration. It does not
> appear at all any more in the files of logs and I do not know that to
> do. Can it be a problem of the internal network? of the switches? Is
> there any bug in the budgetones?
>
>
> Any idea?
>
>
>
> Thanks,
>
> srsergio
>
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-------------- next part --------------
INVITE sip:707 at 217.11.115.168 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.160;branch=z9hG4bK6d1d55e21d956cc8
From: "705" <sip:705 at 217.11.115.168>;tag=ec8465b30e800be8
To: <sip:707 at 217.11.115.168>
Contact: <sip:705 at 192.168.1.160>
Call-ID: 81857e68888de0d4 at 192.168.1.160
CSeq: 15033 INVITE
User-Agent: Grandstream BT100 1.0.4.46
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 327
v=0
o=705 8000 8000 IN IP4 192.168.1.160
s=SIP Call
c=IN IP4 192.168.1.160
t=0 0
m=audio 5004 RTP/AVP 18 8 0 4 2 15 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
12 headers, 15 lines
Using latest request as basis request
Sending to 192.168.1.160 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format ULAW
Found audio format GSM
Found audio format UNKN
Found audio format UNKN
Found description format G729
Found description format PCMA
Found description format PCMU
Found description format G723
Found description format G726-32
Found description format G728
Found description format telephone-event
Capabilities: us - 264, them - 285/0, combined - 264
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.160;branch=z9hG4bK6d1d55e21d956cc8;received=82.223.12.13
From: "705" <sip:705 at 217.11.115.168>;tag=ec8465b30e800be8
To: <sip:707 at 217.11.115.168>;tag=as36a804b8
Call-ID: 81857e68888de0d4 at 192.168.1.160
CSeq: 15033 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:707 at 217.11.115.168>
Proxy-Authenticate: Digest realm="asterisk", nonce="70c81733"
Content-Length: 0
to 82.223.12.13:5060
CAC-IP*CLI>
Sip read:
ACK sip:707 at 217.11.115.168 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.160;branch=z9hG4bK6d1d55e21d956cc8
From: "705" <sip:705 at 217.11.115.168>;tag=ec8465b30e800be8
To: <sip:707 at 217.11.115.168>;tag=as36a804b8
Contact: <sip:705 at 192.168.1.160>
Call-ID: 81857e68888de0d4 at 192.168.1.160
CSeq: 15033 ACK
User-Agent: Grandstream BT100 1.0.4.46
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
11 headers, 0 lines
CAC-IP*CLI>
Sip read:
INVITE sip:707 at 217.11.115.168 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.160;branch=z9hG4bK3c78c7e4601f3d78
From: "705" <sip:705 at 217.11.115.168>;tag=ec8465b30e800be8
To: <sip:707 at 217.11.115.168>
Contact: <sip:705 at 192.168.1.160>
Proxy-Authorization: DIGEST username="705", realm="asterisk", algorithm=MD5, uri="sip:707 at 217.11.115.168", nonce="70c81733", response="42138aecd0e7ca49cdacf4df648503d2"
Call-ID: 81857e68888de0d4 at 192.168.1.160
CSeq: 15034 INVITE
User-Agent: Grandstream BT100 1.0.4.46
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 327
v=0
o=705 8000 8000 IN IP4 192.168.1.160
s=SIP Call
c=IN IP4 192.168.1.160
t=0 0
m=audio 5004 RTP/AVP 18 8 0 4 2 15 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
13 headers, 15 lines
Using latest request as basis request
Sending to 192.168.1.160 : 5060 (NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format ULAW
Found audio format GSM
Found audio format UNKN
Found audio format UNKN
Found description format G729
Found description format PCMA
Found description format PCMU
Found description format G723
Found description format G726-32
Found description format G728
Found description format telephone-event
Capabilities: us - 264, them - 285/0, combined - 264
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 707 in outgoing
list_route: hop: <sip:705 at 192.168.1.160>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.160;branch=z9hG4bK3c78c7e4601f3d78;received=82.223.12.13
From: "705" <sip:705 at 217.11.115.168>;tag=ec8465b30e800be8
To: <sip:707 at 217.11.115.168>;tag=as67f0a00a
Call-ID: 81857e68888de0d4 at 192.168.1.160
CSeq: 15034 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:707 at 217.11.115.168>
Content-Length: 0
to 82.223.12.13:5060
-- Executing Goto("SIP/705-7bb4", "voip-h323|707|1") in new stack
-- Goto (voip-h323,707,1)
-- Executing AGI("SIP/705-7bb4", "AceptaLlamada.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/AceptaLlamada.php
-- AGI Script AceptaLlamada.php completed, returning 0
-- Executing Dial("SIP/705-7bb4", "SIP/707|30|tTrm") in new stack
We're at 192.168.0.207 port 5216
Answering with preferred capability 256
Answering with preferred capability 8
Answering with non-codec capability 1
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:707 at 192.168.0.157:52478 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK34d0afd8
From: "705" <sip:705 at AVANZADA7>;tag=as31ec6245
To: <sip:707 at 192.168.0.157:52478>
Contact: <sip:705 at 192.168.0.207>
Call-ID: 3630716b1a422b3d25287adb5184c69a at 192.168.0.207
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 27 Feb 2004 09:41:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 214
v=0
o=root 7036 7036 IN IP4 192.168.0.207
s=session
c=IN IP4 192.168.0.207
t=0 0
m=audio 5216 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(no NAT) to 192.168.0.157:52478
-- Called 707
We're at 217.11.115.168 port 6952
Answering with preferred capability 256
Answering with preferred capability 8
Answering with non-codec capability 1
Transmitting (NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.160;branch=z9hG4bK3c78c7e4601f3d78;received=82.223.12.13
From: "705" <sip:705 at 217.11.115.168>;tag=ec8465b30e800be8
To: <sip:707 at 217.11.115.168>;tag=as67f0a00a
Call-ID: 81857e68888de0d4 at 192.168.1.160
CSeq: 15034 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:707 at 217.11.115.168>
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 7036 7036 IN IP4 217.11.115.168
s=session
c=IN IP4 217.11.115.168
t=0 0
m=audio 6952 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 82.223.12.13:5060
CAC-IP*CLI>
Sip read:
SIP/2.0 100 trying
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK34d0afd8
From: "705" <sip:705 at AVANZADA7>;tag=as31ec6245
To: <sip:707 at 192.168.0.157:52478>
Call-ID: 3630716b1a422b3d25287adb5184c69a at 192.168.0.207
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.4.46
Content-Length: 0
8 headers, 0 lines
CAC-IP*CLI>
Sip read:
SIP/2.0 180 ringing
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK34d0afd8
From: "705" <sip:705 at AVANZADA7>;tag=as31ec6245
To: <sip:707 at 192.168.0.157:52478>;tag=eefef56c3a79f4d3
Call-ID: 3630716b1a422b3d25287adb5184c69a at 192.168.0.207
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.4.46
Content-Length: 0
8 headers, 0 lines
-- SIP/707-f1d0 is ringing
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.168.0.157:52478 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK12c0c182
From: "asterisk" <sip:asterisk at AVANZADA7>;tag=as5f2edc41
To: <sip:192.168.0.157:52478>
Contact: <sip:asterisk at 192.168.0.207>
Call-ID: 54647aad0f549c577e544d35757e9ff8 at 192.168.0.207
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 27 Feb 2004 09:41:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 192.168.0.157:52478
11 headers, 0 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK12c0c182
From: "asterisk" <sip:asterisk at AVANZADA7>;tag=as5f2edc41
To: <sip:192.168.0.157:52478>;tag=b95680682b6ccef1
Call-ID: 54647aad0f549c577e544d35757e9ff8 at 192.168.0.207
CSeq: 102 OPTIONS
User-Agent: Grandstream BT100 1.0.4.46
Contact: <sip:707 at 192.168.0.157:52478>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK34d0afd8
From: "705" <sip:705 at AVANZADA7>;tag=as31ec6245
To: <sip:707 at 192.168.0.157:52478>;tag=eefef56c3a79f4d3
Call-ID: 3630716b1a422b3d25287adb5184c69a at 192.168.0.207
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.4.46
Contact: <sip:707 at 192.168.0.157:52478>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 147
v=0
o=707 8000 8000 IN IP4 192.168.0.157
s=SIP Call>
c=IN IP4 192.168.0.157
t=0 0
m=audio 40442 RTP/AVP 18
a=rtpmap:18 G729/8000
a=ptime:20
11 headers, 8 lines
Found audio format UNKN
Found description format G729
Capabilities: us - 264, them - 256/0, combined - 256
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:707 at 192.168.0.157:52478>
set_destination: Parsing <sip:707 at 192.168.0.157:52478> for address/port to send to
set_destination: set destination to 192.168.0.157, port 52478
Transmitting:
ACK sip:707 at 192.168.0.157:52478 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK34d0afd8
From: "705" <sip:705 at AVANZADA7>;tag=as31ec6245
To: <sip:707 at 192.168.0.157:52478>;tag=eefef56c3a79f4d3
Contact: <sip:705 at 192.168.0.207>
Call-ID: 3630716b1a422b3d25287adb5184c69a at 192.168.0.207
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.0.157:52478
-- SIP/707-f1d0 answered SIP/705-7bb4
We're at 217.11.115.168 port 6952
Answering with preferred capability 256
Answering with preferred capability 8
Answering with non-codec capability 1
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.160;branch=z9hG4bK3c78c7e4601f3d78;received=82.223.12.13
From: "705" <sip:705 at 217.11.115.168>;tag=ec8465b30e800be8
To: <sip:707 at 217.11.115.168>;tag=as67f0a00a
Call-ID: 81857e68888de0d4 at 192.168.1.160
CSeq: 15034 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:707 at 217.11.115.168>
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 7036 7037 IN IP4 217.11.115.168
s=session
c=IN IP4 217.11.115.168
t=0 0
m=audio 6952 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 82.223.12.13:5060
-- Attempting native bridge of SIP/705-7bb4 and SIP/707-f1d0
CAC-IP*CLI>
Sip read:
ACK sip:707 at 217.11.115.168 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.160;branch=z9hG4bK71d2bacd6cc1af20
From: "705" <sip:705 at 217.11.115.168>;tag=ec8465b30e800be8
To: <sip:707 at 217.11.115.168>;tag=as67f0a00a
Contact: <sip:705 at 192.168.1.160>
Proxy-Authorization: DIGEST username="705", realm="asterisk", algorithm=MD5, uri="sip:707 at 217.11.115.168", nonce="70c81733", response="31b573fa976e9aea436fb431c3a0a138"
Call-ID: 81857e68888de0d4 at 192.168.1.160
CSeq: 15034 ACK
User-Agent: Grandstream BT100 1.0.4.46
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
12 headers, 0 lines
11 headers, 0 linesug
Reliably Transmitting:
OPTIONS sip:82.223.12.13 SIP/2.0
Via: SIP/2.0/UDP 217.11.115.168:5060;branch=z9hG4bK1b3f5364
From: "asterisk" <sip:asterisk at AVANZADA7>;tag=as58900006
To: <sip:82.223.12.13>
Contact: <sip:asterisk at 217.11.115.168>
Call-ID: 0041ad747a472fe93787b9420af16a7b at 217.11.115.168
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 27 Feb 2004 09:42:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 82.223.12.13:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.11.115.168:5060;branch=z9hG4bK1b3f5364
From: "asterisk" <sip:asterisk at AVANZADA7>;tag=as58900006
To: <sip:82.223.12.13>;tag=e3942a33a0e01bc5
Call-ID: 0041ad747a472fe93787b9420af16a7b at 217.11.115.168
CSeq: 102 OPTIONS
User-Agent: Grandstream BT100 1.0.4.46
Contact: <sip:705 at 192.168.1.160>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
11 headers, 0 lines
-------------- next part --------------
705 Asterisk 707
|----------------Invite(707)----------------------->| |
| RTP Port=5004 | |
| | |
|<-----------------407-------------------------------| |
| | |
|------------------------ACK----------------------->| |
| | |
|----------------Invite(707)----------------------->| |
| RTP Port=5004 | |
| | |
|<-------------------Trying-------------------------| |
| |---------------------Invite(707)------------------>|
| | RTP Port=5216 |
|<-------183 Session Progress-------------- | |
| RTP Port=6952| |
| |<-------------------Trying------------------------- |
| | |
| |<-------------------Ringing----------------------- |
| | |
| | |
| |------------------------Options------------------>|
| | |
| |<--------------------200 OK---------------------- |
| | |
| |<--------------------200 OK---------------------- |
| | RTP Port=40442|
| | |
| |----------------------ACK------------------------->|
| | |
|<----------------------200 OK-------------------- | |
| RTP Port=6952 | |
| | |
|------------------------ACK----------------------->| |
| | |
|<-------------------OPTIONS------------------- | |
| | |
| | |
|---------------------200 OK---------------------> | |
| | |
| | |
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