[asterisk-users] Asterisk not sending RTP
Nir Simionovich
nirs at exchange.atelis.net
Sun Sep 3 07:28:48 MST 2006
well, here is the full SIP debug:
Sep 3 10:05:59 DEBUG[6139] manager.c: Manager received command
'Originate'
Sep 3 10:05:59 DEBUG[6139] chan_sip.c: Setting NAT on RTP to 0
Sep 3 10:05:59 DEBUG[6139] chan_sip.c: Outgoing Call for 972544482826
Sep 3 10:05:59 VERBOSE[6139] logger.c: We're at 192.117.233.176 port
18372
Sep 3 10:05:59 VERBOSE[6139] logger.c: Adding codec 0x4 (ulaw) to SDP
Sep 3 10:05:59 VERBOSE[6139] logger.c: Adding codec 0x8 (alaw) to SDP
Sep 3 10:05:59 VERBOSE[6139] logger.c: 13 headers, 9 lines
Sep 3 10:05:59 VERBOSE[6139] logger.c: Reliably Transmitting (no NAT) to
62.219.61.73:5060:
INVITE sip:972544482826 at 62.219.61.73 SIP/2.0
Via: SIP/2.0/UDP 192.117.233.176:5060;branch=z9hG4bK33e91b50;rport
From: "972544482826" <sip:972544482826 at 192.117.233.176>;tag=as5514479f
To: <sip:972544482826 at 62.219.61.73>
Contact: <sip:972544482826 at 192.117.233.176>
Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 03 Sep 2006 09:05:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 188
v=0
o=root 6093 6093 IN IP4 192.117.233.176
s=session
c=IN IP4 192.117.233.176
t=0 0
m=audio 18372 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
---
Sep 3 10:05:59 VERBOSE[6124] logger.c:
<-- SIP read from 62.219.61.73:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport
=5060
From: "972544482826" <sip:972544482826 at 192.117.233.176>;tag=as5514479f
To: <sip:972544482826 at 62.219.61.73>
Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176
CSeq: 102 INVITE
Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (6 headers 0 lines)Sep 3
10:05:59 VERBOSE[6124] logger.c: --- (6 headers 0 li
nes)---
Sep 3 10:05:59 DEBUG[6124] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on '4a8dc38a2198b8fb0
f40b63f7680671e at 192.117.233.176' Request 102: Found
Sep 3 10:05:59 VERBOSE[6124] logger.c:
<-- SIP read from 62.219.61.73:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport
=5060
From: "972544482826" <sip:972544482826 at 192.117.233.176>;tag=as5514479f
To: <sip:972544482826 at 62.219.61.73>;tag=2607
Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176
CSeq: 102 INVITE
Supported: timer,100rel
Content-Length: 0
Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (8 headers 0 lines)Sep 3
10:05:59 VERBOSE[6124] logger.c: --- (8 headers 0 li
nes)---
Sep 3 10:05:59 DEBUG[6124] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on '4a8dc38a2198b8fb0
f40b63f7680671e at 192.117.233.176' Request 102: Found
Sep 3 10:06:00 VERBOSE[6124] logger.c:
<-- SIP read from 62.219.61.73:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport
=5060
From: "972544482826" <sip:972544482826 at 192.117.233.176>;tag=as5514479f
To: <sip:972544482826 at 62.219.61.73>;tag=2607
Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176
CSeq: 102 INVITE
Content-Type: application/sdp
Supported: timer,100rel
Content-Length: 123
v=0
o=MG4000|2.0 14726 14726 IN IP4 62.219.61.73
s=-
c=IN IP4 62.219.61.73
t=0 0
m=audio 51644 RTP/AVP 0
a=ptime:10
Sep 3 10:06:00 VERBOSE[6124] logger.c: --- (9 headers 7 lines)Sep 3
10:06:00 VERBOSE[6124] logger.c: --- (9 headers 7 li
nes)---
Sep 3 10:06:00 DEBUG[6124] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on '4a8dc38a2198b8fb0
f40b63f7680671e at 192.117.233.176' Request 102: Found
Sep 3 10:06:00 VERBOSE[6124] logger.c: Found RTP audio format 0
Sep 3 10:06:00 VERBOSE[6124] logger.c: Peer audio RTP is at port
62.219.61.73:51644
Sep 3 10:06:00 DEBUG[6124] chan_sip.c: Peer audio RTP is at port
62.219.61.73:51644
Sep 3 10:06:00 VERBOSE[6124] logger.c: Capabilities: us - 0xc
(ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), c
ombined - 0x4 (ulaw)
Sep 3 10:06:00 VERBOSE[6124] logger.c: Non-codec capabilities: us - 0x1
(telephone-event), peer - 0x0 (nothing), combined
- 0x0 (nothing)
Sep 3 10:06:01 VERBOSE[6124] logger.c:
<-- SIP read from 62.219.61.73:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport
=5060
From: "972544482826" <sip:972544482826 at 192.117.233.176>;tag=as5514479f
To: <sip:972544482826 at 62.219.61.73>;tag=2607
Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176
CSeq: 102 INVITE
Content-Type: application/sdp
Contact: <sip:972544482826 at 62.219.61.73:5060;transport=udp>;user=phone
Supported: timer,100rel
Content-Length: 123
v=0
o=MG4000|2.0 14726 14726 IN IP4 62.219.61.73
s=-
c=IN IP4 62.219.61.73
t=0 0
m=audio 51644 RTP/AVP 0
a=ptime:10
Sep 3 10:06:01 VERBOSE[6124] logger.c: --- (10 headers 7 lines)Sep 3
10:06:01 VERBOSE[6124] logger.c: --- (10 headers 7
lines)---
Sep 3 10:06:01 DEBUG[6124] chan_sip.c: Acked pending invite 102
Sep 3 10:06:01 DEBUG[6124] chan_sip.c: Stopping retransmission on
'4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176' of R
equest 102: Match Found
Sep 3 10:06:01 VERBOSE[6124] logger.c: Found RTP audio format 0
Sep 3 10:06:01 VERBOSE[6124] logger.c: Peer audio RTP is at port
62.219.61.73:51644
Sep 3 10:06:01 DEBUG[6124] chan_sip.c: Peer audio RTP is at port
62.219.61.73:51644
Sep 3 10:06:01 VERBOSE[6124] logger.c: Capabilities: us - 0xc
(ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), c
ombined - 0x4 (ulaw)
Sep 3 10:06:01 VERBOSE[6124] logger.c: Non-codec capabilities: us - 0x1
(telephone-event), peer - 0x0 (nothing), combined
- 0x0 (nothing)
Sep 3 10:06:01 DEBUG[6124] chan_sip.c: build_route: Contact hop:
<sip:972544482826 at 62.219.61.73:5060;transport=udp>;user=
phone
Sep 3 10:06:01 VERBOSE[6124] logger.c: list_route: hop:
<sip:972544482826 at 62.219.61.73:5060;transport=udp>
Sep 3 10:06:01 VERBOSE[6124] logger.c: set_destination: Parsing
<sip:972544482826 at 62.219.61.73:5060;transport=udp> for ad
dress/port to send to
Sep 3 10:06:01 VERBOSE[6124] logger.c: set_destination: set destination
to 62.219.61.73, port 5060
Sep 3 10:06:01 VERBOSE[6124] logger.c: Transmitting (no NAT) to
62.219.61.73:5060:
ACK sip:972544482826 at 62.219.61.73:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.117.233.176:5060;branch=z9hG4bK052932bd;rport
From: "972544482826" <sip:972544482826 at 192.117.233.176>;tag=as5514479f
To: <sip:972544482826 at 62.219.61.73>;tag=2607
Contact: <sip:972544482826 at 192.117.233.176>
Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Sep 3 10:06:01 DEBUG[6097] channel.c: Avoiding initial deadlock for
'SIP/BezeqInt-007c1920'
Sep 3 10:06:01 VERBOSE[6139] logger.c: > Channel
SIP/BezeqInt-007c1920 was answered.
Sep 3 10:06:01 VERBOSE[6147] logger.c: -- Executing
NoOp("SIP/BezeqInt-007c1920", "in MusicOnHoldTest") in new stack
Sep 3 10:06:01 VERBOSE[6147] logger.c: -- Executing
Answer("SIP/BezeqInt-007c1920", "") in new stack
Sep 3 10:06:01 VERBOSE[6147] logger.c: -- Executing
Playback("SIP/BezeqInt-007c1920", "AEN2") in new stack
Sep 3 10:06:01 DEBUG[6147] channel.c: Scheduling timer at 160 sample
intervals
Sep 3 10:06:01 VERBOSE[6147] logger.c: -- Playing 'AEN2' (language
'en')
Sep 3 10:06:38 VERBOSE[6124] logger.c:
<-- SIP read from 62.219.61.73:5060:
BYE sip:972544482826 at 192.117.233.176 SIP/2.0
Via: SIP/2.0/UDP
62.219.61.73:5060;branch=z9hG4bK0g1mob302o71ekc331o0sd0000g00.1
Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176
From: <sip:972544482826 at 62.219.61.73>;tag=2607
To: "972544482826" <sip:972544482826 at 192.117.233.176>;tag=as5514479f
CSeq: 1 BYE
Supported: timer,100rel
Max-Forwards: 69
Content-Length: 0
Sep 3 10:06:38 VERBOSE[6124] logger.c: --- (9 headers 0 lines)Sep 3
10:06:38 VERBOSE[6124] logger.c: --- (9 headers 0 li
nes)---
Sep 3 10:06:38 VERBOSE[6124] logger.c: Sending to 62.219.61.73 : 5060
(non-NAT)
Sep 3 10:06:38 VERBOSE[6124] logger.c: Transmitting (no NAT) to
62.219.61.73:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
62.219.61.73:5060;branch=z9hG4bK0g1mob302o71ekc331o0sd0000g00.1;received=6
2.219.61.73
From: <sip:972544482826 at 62.219.61.73>;tag=2607
To: "972544482826" <sip:972544482826 at 192.117.233.176>;tag=as5514479f
Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e at 192.117.233.176
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:972544482826 at 192.117.233.176>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
Sep 3 10:06:38 DEBUG[6147] channel.c: Scheduling timer at 0 sample
intervals
Sep 3 10:06:38 VERBOSE[6147] logger.c: == Spawn extension
(MusicOnHoldTest, 972544482826, 3) exited non-zero on 'SIP/Be
zeqInt-007c1920'
Sep 3 10:06:38 DEBUG[6147] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Sep 3 10:06:38 DEBUG[6147] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr (calldate,clid,src,dst,d
context,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,a
maflags,accountcode,uniqueid,userfield) VALUES (
'2006-09-03
10:06:01','972544482826','972544482826','972544482826','MusicOnHoldTest',
'SIP/BezeqInt-007c1920','','Playback
','AEN2',37,37,'ANSWERED',3,'','1157274359.7','')
Sep 3 10:06:38 DEBUG[6147] pbx.c: Function result is '972544482826'
Sep 3 10:06:38 DEBUG[6147] pbx.c: Function result is '972544482826'
Any idea would be highly appreciated...
Nir S
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jeremy
McNamara
Sent: Sunday, September 03, 2006 4:09 PM
To: nirs at atelis.net; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Asterisk not sending RTP
Nir Simionovich wrote:
> Any ideas anyone ?
Do you have a compatible codec?
What does the SDP show?
Is sip.conf binding to a valid IP address?
Jeremy McNamara
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