[Asterisk-Users] Problem Sending to Cisco 3660 Sip Endpoint
david winter
dwinter at planet-telecom.com
Sat Sep 25 12:55:22 MST 2004
All,
I am trying to do a dial to a cisco3660 endpoint. see the below
extensions.conf, sip.conf, and output to see my problem. Thanks in
advance for any input. In the debug look for the WARNING lines. thanks!
exten => 5149053538,1,Answer
exten => 5149053538,2,Wait,2
exten => 5149053538,3,Playback(you-sound-cute)
exten => 5149053538,4,Dial(SIP/61393881910 at melbourne,5)
exten => 5149053538,105,Hangup
[general]
disallow=all
allow=ulaw
allow=alaw
allow=g729
[melbourne]
type=friend
defaultip=xxx.xxx.xxx.xxx
context=demo
[montreal]
type=friend
context=demo
defaultip=yyy.yyy.yyy.yyy
*CLI>
Sip read:
INVITE sip:5149053538 at xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKD1E
From: <sip:8138174204 at yyy.yyy.yyy.yyy>;tag=F9E311A8-246C
To: <sip:5149053538 at xxx.xxx.xxx.xxx>
Date: Sat, 25 Sep 2004 19:51:18 GMT
Call-ID: 1A028391-E6311D9-BB32A8C6-CEB81596 at yyy.yyy.yyy.yyy
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 436292417-241373657-3182559241-3907589232
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID:
<sip:8138174204 at yyy.yyy.yyy.yyy>;party=calling;screen=yes;privacy=off
Timestamp: 1096141878
Contact: <sip:8138174204 at yyy.yyy.yyy.yyy:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 194
v=0
o=CiscoSystemsSIP-GW-UserAgent 1631 5118 IN IP4 yyy.yyy.yyy.yyy
s=SIP Call
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 19366 RTP/AVP 0
c=IN IP4 yyy.yyy.yyy.yyy
a=rtpmap:0 PCMU/8000
a=ptime:20
20 headers, 9 lines
Using latest request as basis request
Sending to yyy.yyy.yyy.yyy : 5060 (non-NAT)
Found RTP audio format 0
Peer audio RTP is at port yyy.yyy.yyy.yyy:19366
Found description format PCMU
Capabilities: us - 0x10c(ULAW|ALAW|G729A), peer -
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined -
0x0(EMPTY)
Found no matching peer or user for 'yyy.yyy.yyy.yyy:58107'
Looking for 5149053538 in default
list_route: hop: <sip:8138174204 at yyy.yyy.yyy.yyy:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKD1E
From: <sip:8138174204 at yyy.yyy.yyy.yyy>;tag=F9E311A8-246C
To: <sip:5149053538 at xxx.xxx.xxx.xxx>;tag=as2f5e7572
Call-ID: 1A028391-E6311D9-BB32A8C6-CEB81596 at yyy.yyy.yyy.yyy
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5149053538 at xxx.xxx.xxx.xxx>
Content-Length: 0
to yyy.yyy.yyy.yyy:5060
-- Executing Answer("SIP/yyy.yyy.yyy.yyy-08141378", "") in new stack
We're at xxx.xxx.xxx.xxx port 12034
Answering with preferred capability 0x4(ULAW)
Answering with preferred capability 0x8(ALAW)
Answering with preferred capability 0x100(G729A)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKD1E
From: <sip:8138174204 at yyy.yyy.yyy.yyy>;tag=F9E311A8-246C
To: <sip:5149053538 at xxx.xxx.xxx.xxx>;tag=as2f5e7572
Call-ID: 1A028391-E6311D9-BB32A8C6-CEB81596 at yyy.yyy.yyy.yyy
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5149053538 at xxx.xxx.xxx.xxx>
Content-Type: application/sdp
Content-Length: 210
v=0
o=root 19664 19664 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 12034 RTP/AVP 0 8 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
to yyy.yyy.yyy.yyy:5060
-- Executing Wait("SIP/yyy.yyy.yyy.yyy-08141378", "2") in new stack
Sip read:
ACK sip:5149053538 at xxx.xxx.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK1D34
From: <sip:8138174204 at yyy.yyy.yyy.yyy>;tag=F9E311A8-246C
To: <sip:5149053538 at xxx.xxx.xxx.xxx>;tag=as2f5e7572
Date: Sat, 25 Sep 2004 19:51:18 GMT
Call-ID: 1A028391-E6311D9-BB32A8C6-CEB81596 at yyy.yyy.yyy.yyy
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0
9 headers, 0 lines
-- Executing Playback("SIP/yyy.yyy.yyy.yyy-08141378",
"you-sound-cute") in new stack
-- Playing 'you-sound-cute' (language 'en')
-- Executing Dial("SIP/yyy.yyy.yyy.yyy-08141378",
"SIP/61393881910 at melbourne|5") in new stack
We're at xxx.xxx.xxx.xxx port 14742
Answering/Requesting with root capability 4
Answering with preferred capability 0x8(ALAW)
Answering with preferred capability 0x100(G729A)
Answering with non-codec capability 0x1(G723)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:61393881910 at xxx.xxx.xxx.xxx:0 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3c032cdd
From: "8138174204" <sip:8138174204 at xxx.xxx.xxx.xxx>;tag=as24022d46
To: <sip:61393881910 at xxx.xxx.xxx.xxx:0>
Contact: <sip:8138174204 at xxx.xxx.xxx.xxx>
Call-ID: 6581583d4e5383e203162557304975a4 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Sep 2004 19:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 19664 19664 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 14742 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to xxx.xxx.xxx.xxx:0
Sep 25 15:47:21 WARNING[1110272944]: chan_sip.c:598 __sip_xmit: sip_xmit
of 0x81487dc (len 755) to xxx.xxx.xxx.xxx returned -1: Invalid argument
-- Called 61393881910 at melbourne
Retransmitting #1 (no NAT):
INVITE sip:61393881910 at xxx.xxx.xxx.xxx:0 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3c032cdd
From: "8138174204" <sip:8138174204 at xxx.xxx.xxx.xxx>;tag=as24022d46
To: <sip:61393881910 at xxx.xxx.xxx.xxx:0>
Contact: <sip:8138174204 at xxx.xxx.xxx.xxx>
Call-ID: 6581583d4e5383e203162557304975a4 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Sep 2004 19:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 19664 19664 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 14742 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to xxx.xxx.xxx.xxx:0
Sep 25 15:47:22 WARNING[1087986608]: chan_sip.c:598 __sip_xmit: sip_xmit
of 0x81487dc (len 755) to xxx.xxx.xxx.xxx returned -1: Invalid argument
Retransmitting #2 (no NAT):
INVITE sip:61393881910 at xxx.xxx.xxx.xxx:0 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3c032cdd
From: "8138174204" <sip:8138174204 at xxx.xxx.xxx.xxx>;tag=as24022d46
To: <sip:61393881910 at xxx.xxx.xxx.xxx:0>
Contact: <sip:8138174204 at xxx.xxx.xxx.xxx>
Call-ID: 6581583d4e5383e203162557304975a4 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Sep 2004 19:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 266
v=0
o=root 19664 19664 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 14742 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to xxx.xxx.xxx.xxx:0
Sep 25 15:47:23 WARNING[1087986608]: chan_sip.c:598 __sip_xmit: sip_xmit
of 0x81487dc (len 755) to xxx.xxx.xxx.xxx returned -1: Invalid argument
--
David Winter
Senior Network Engineer
Planet-Telecom, Inc.
Tampa FL
(813)901-5182 Office
(813)864-3162 Direct
(813)817-4204 Mobile
(813)881-9762 Fax
------------------------------------------
AIM: mobofool
ICQ: 3563403
MSN: dwinter at vt.edu
Y!: vt_fool
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