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/[EMAIL PROTECTED],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:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKD1E
From: <sip:[EMAIL PROTECTED]>;tag=F9E311A8-246C
To: <sip:[EMAIL PROTECTED]>
Date: Sat, 25 Sep 2004 19:51:18 GMT
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;party=calling;screen=yes;privacy=off
Timestamp: 1096141878
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKD1E
From: <sip:[EMAIL PROTECTED]>;tag=F9E311A8-246C
To: <sip:[EMAIL PROTECTED]>;tag=as2f5e7572
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]>;tag=F9E311A8-246C To: <sip:[EMAIL PROTECTED]>;tag=as2f5e7572 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK1D34 From: <sip:[EMAIL PROTECTED]>;tag=F9E311A8-246C To: <sip:[EMAIL PROTECTED]>;tag=as2f5e7572 Date: Sat, 25 Sep 2004 19:51:18 GMT Call-ID: [EMAIL PROTECTED] 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/[EMAIL PROTECTED]|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:[EMAIL PROTECTED]:0 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3c032cdd
From: "8138174204" <sip:[EMAIL PROTECTED]>;tag=as24022d46
To: <sip:[EMAIL PROTECTED]:0>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 [EMAIL PROTECTED]
Retransmitting #1 (no NAT):
INVITE sip:[EMAIL PROTECTED]:0 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3c032cdd
From: "8138174204" <sip:[EMAIL PROTECTED]>;tag=as24022d46
To: <sip:[EMAIL PROTECTED]:0>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:0 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3c032cdd
From: "8138174204" <sip:[EMAIL PROTECTED]>;tag=as24022d46
To: <sip:[EMAIL PROTECTED]:0>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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: [EMAIL PROTECTED]
Y!: vt_fool
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