[Asterisk-Users] Help with Sip call problems - Whats not working?
Wes Marderness
wmarderness at sigmabit.com
Mon Feb 9 09:37:56 MST 2004
What does your extensions.conf look like? Did you answer() the call first ?
wes
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Chris Lee
Sent: Monday, February 09, 2004 6:01 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Help with Sip call problems - Whats not
working?
When I press a key (8) on the phone, it should play a few bits of audio
and go to voicemail for testing. I dont get any sound back, and it
appears the call is progressing without me.
Here is the console output with sip debug:
Sip read:
INVITE sip:8 at 10.10.10.3 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
Contact: <sip:p3000 at 10.10.10.2>
Max-Forwards: 70
User-Agent: DrayTek UA-1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Type: application/sdp
Content-Length: 287
v=0
o=p3000 5972727 56415 IN IP4 10.10.10.2
s=SIP Call
c=IN IP4 10.10.10.2
t=0 0
m=audio 10096 RTP/AVP 0 8 18 4 2 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 g723/8000
a=rtpmap:2 g726/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
12 headers, 13 lines
Using latest request as basis request
Sending to 10.10.10.2 : 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 description format pcmu
Found description format pcma
Found description format G729
Found description format g723
Found description format g726
Found description format telephone-event
Capabilities: us - 14, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 8 in wellingborough-road
list_route: hop: <sip:p3000 at 10.10.10.2>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Length: 0
to 10.10.10.2:5060
-- Executing BackGround("SIP/p3000-1186",
"sounds/carried-away-by-monkeys") in new stack
We're at 10.10.10.3 port 17190
Answering with preferred capability 2
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 17878 17878 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 17190 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 10.10.10.2:5060
-- Playing 'sounds/carried-away-by-monkeys' (language 'en')
babybell*CLI>
Sip read:
INVITE sip:8 at 10.10.10.3 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
Contact: <sip:p3000 at 10.10.10.2>
Max-Forwards: 70
User-Agent: DrayTek UA-1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Type: application/sdp
Content-Length: 287
v=0
o=p3000 5972727 56415 IN IP4 10.10.10.2
s=SIP Call
c=IN IP4 10.10.10.2
t=0 0
m=audio 10096 RTP/AVP 0 8 18 4 2 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 g723/8000
a=rtpmap:2 g726/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
12 headers, 13 lines
Ignoring this request
We're at 10.10.10.3 port 17190
Answering with preferred capability 2
Answering with preferred capability 4
Answering with preferred capability 8
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 17878 17879 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 17190 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 10.10.10.2:5060
babybell*CLI>
Sip read:
ACK sip:8 at 10.10.10.3 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-AAE-26994
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: DrayTek UA-1.0
Content-Length: 0
9 headers, 0 lines
Retransmitting #1 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 17878 17878 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 17190 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 10.10.10.2:5060
Retransmitting #2 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 17878 17878 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 17190 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 10.10.10.2:5060
-- Executing BackGround("SIP/p3000-1186", "sounds/lots-o-monkeys")
in new stack
-- Playing 'sounds/lots-o-monkeys' (language 'en')
Retransmitting #3 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 17878 17878 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 17190 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 10.10.10.2:5060
Retransmitting #4 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 17878 17878 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 17190 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 10.10.10.2:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841
From: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
To: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8 at 10.10.10.3>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 17878 17878 IN IP4 10.10.10.3
s=session
c=IN IP4 10.10.10.3
t=0 0
m=audio 17190 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 10.10.10.2:5060
Feb 9 09:47:15 WARNING[81926]: chan_sip.c:471 retrans_pkt: Maximum
retries exceeded on call akZ-25626 at 10.10.10.2 for seqno 1 (Response)
== Spawn extension (wellingborough-road, 8, 2) exited non-zero on
'SIP/p3000-1186'
-- Executing Hangup("SIP/p3000-1186", "") in new stack
== Spawn extension (wellingborough-road, h, 1) exited non-zero on
'SIP/p3000-1186'
set_destination: Parsing <sip:p3000 at 10.10.10.2> for address/port to send to
set_destination: set destination to 10.10.10.2, port 5060
Reliably Transmitting:
BYE sip:p3000 at 10.10.10.2 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK3faaa31d
From: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
To: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
Contact: <sip:8 at 10.10.10.3>
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 10.10.10.2:5060
babybell*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK3faaa31d
From: <sip:8 at 10.10.10.3>;tag=as3bf9fee8
To: p3000 <sip:p3000 at 10.10.10.3:5060>;tag=TdR-16808
Call-ID: akZ-25626 at 10.10.10.2
CSeq: 102 BYE
Content-Length: 0
7 headers, 0 lines
Message is BYE
#######################
Calls originating at FXO and going to this extension work fine. Calls
originating at this extension are a problem.
Any help would be great
Regards
Chris Lee
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