This is a call coming in through the ISDN to 7940's.
Answering with non-codec capability 1 - Is that the problem?
SIP Debugging Enabled
We're at 10.1.0.11 port 18406
Answering/Requesting with root capability 8
Answering/Requesting with preferred capability 4
Answering/Requesting with preferred capability 8
Answering with non-codec capability 1 <<<<<<-------------
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:20@10.1.0.119 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as03605c88
To: <sip:20@10.1.0.119>
Contact: <sip:asterisk@10.1.0.11>
Call-ID: 0a0561966f285f534c5d9fc26cf56414@10.1.0.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 16 Apr 2004 19:21:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 14316 14316 IN IP4 10.1.0.11
=sessionI>
c=IN IP4 10.1.0.11
t=0 0
m=audio 18406 RTP/AVP 8 0 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 10.1.0.119:5060
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as03605c88
To: <sip:20@10.1.0.119>
Call-ID: 0a0561966f285f534c5d9fc26cf56414@10.1.0.11
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:20@10.1.0.119:5060>
Content-Length: 0
9 headers, 0 lines
pbx01*CLI>
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK6f5d4357
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as03605c88
To: <sip:20@10.1.0.119>;tag=000e3857223c0238011930bc-566c64f8
Call-ID: 0a0561966f285f534c5d9fc26cf56414@10.1.0.11
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:20@10.1.0.119:5060>
Content-Length: 0
9 headers, 0 lines
We're at 10.1.0.11 port 18198
Answering/Requesting with root capability 8
Answering/Requesting with preferred capability 4
Answering/Requesting with preferred capability 8
Answering with non-codec capability 1
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:26@10.1.0.120 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as286f917d
To: <sip:26@10.1.0.120>
Contact: <sip:asterisk@10.1.0.11>
Call-ID: 64a5e840038490150da1fb64206238d6@10.1.0.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 16 Apr 2004 19:21:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 14316 14316 IN IP4 10.1.0.11
=sessionI>
c=IN IP4 10.1.0.11
t=0 0
m=audio 18198 RTP/AVP 8 0 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 10.1.0.120:5060
pbx01*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as286f917d
To: <sip:26@10.1.0.120>
Call-ID: 64a5e840038490150da1fb64206238d6@10.1.0.11
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:26@10.1.0.120:5060>
Content-Length: 0
9 headers, 0 lines
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK39bce4f9
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as286f917d
To: <sip:26@10.1.0.120>;tag=000f23ad6e25021c1c1f7e2d-532b7f03
Call-ID: 64a5e840038490150da1fb64206238d6@10.1.0.11
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:26@10.1.0.120:5060>
Content-Length: 0
9 headers, 0 lines
We're at 10.1.0.11 port 29654
Answering/Requesting with root capability 8
Answering/Requesting with preferred capability 4
Answering/Requesting with preferred capability 8
Answering with non-codec capability 1
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:22@10.1.0.125 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as19596a6b
To: <sip:22@10.1.0.125>
Contact: <sip:asterisk@10.1.0.11>
Call-ID: 2e641a15795a90b630e31ed87ac1b9d7@10.1.0.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 16 Apr 2004 19:21:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 258
v=0
o=root 14316 14316 IN IP4 10.1.0.11
=sessionI> sip debug
c=IN IP4 10.1.0.11
t=0 0
m=audio 29654 RTP/AVP 8 0 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 10.1.0.125:5060
pbx01*CLI> sip debug
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as19596a6b
To: <sip:22@10.1.0.125>
Call-ID: 2e641a15795a90b630e31ed87ac1b9d7@10.1.0.11
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:22@10.1.0.125:5060>
Content-Length: 0
9 headers, 0 lines
pbx01*CLI> sip debug
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.0.11:5060;branch=z9hG4bK78395ced
From: "asterisk" <sip:asterisk@10.1.0.11>;tag=as19596a6b
To: <sip:22@10.1.0.125>;tag=000f23ac489f00c519541b4d-016de7d7
Call-ID: 2e641a15795a90b630e31ed87ac1b9d7@10.1.0.11
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:22@10.1.0.125:5060>
Content-Length: 0
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tracy R Reed
Sent: 16 April 2004 19:20
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Cisco 7940 no audio
On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake
thusly:> When we receive or make a call to the outside - they can hear us, but
we> cant hear them.
I have had this problem several times and so far no resolution. However
for me it has always been with IAX. I have been told that IAX is
supposed
to be NAT-safe but that does not seem to be the case for me. For
example:
SIP (grandstream, snom) ->Asterisk->NAT->Asterisk->SIP (grandstream,
snom)
He can hear me but I can't hear him.
In another case I had:
IAXclient (soft phone)->NAT->Asterisk->Snom
And I could hear him but he could not hear me. Same phone system and
settings as above.
However as soon as I switched the first users phone to talk directly to
my
Asterisk box with SIP it worked perfectly. And when I switched the user
in
the second case to a SIP based soft phone it also worked just fine. SIP
has worked better through NAT than IAX (with nat=yes in sip.conf) which
is
bizarre and contrary to what I have read where IAX should be NAT-safe
and
SIP not.
I have dreams of a world fully converted to IPv6 where NAT no longer
exists. Alas, it is but a dream.
--
Tracy Reed The attachment is a digital signature.
http://copilotconsulting.com More info:
http://copilotconsulting.com/sig
