[asterisk-bugs] [Asterisk 0015688]: The "port" parameter for an outbound provider is not being respected

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15688 
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Reported By:                falves11
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15688
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2 
SVN Revision (number only!): 211391 
Request Review:              
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Date Submitted:             2009-08-10 13:32 CDT
Last Modified:              2009-08-20 14:29 CDT
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Summary:                    The "port" parameter for an outbound provider is not
being respected
Description: 
Audio is at 208.78.161.230 port 25574                                      
                                  
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 216.115.20.41:5061:
INVITE sip:12564286161 at sphone.vopr.vonage.net:5061 SIP/2.0
Via: SIP/2.0/UDP 208.78.161.230:5060;branch=z9hG4bK0e8eb88f;rport
Max-Forwards: 70
From: "13106017395"
<sip:13106xxxxx at sphone.vopr.vonage.net>;tag=as0f1f2595
To: <sip:12564286161 at sphone.vopr.vonage.net:5061>
Contact: <sip:13106xxxxx at 208.78.161.230>
Call-ID: 340e75926a10ebcd40e5c7b62127cf47 at sphone.vopr.vonage.net
CSeq: 102 INVITE
User-Agent: X-PRO release 1103g
Remote-Party-ID: "13106xxxxx"
<sip:13106017395 at sphone.vopr.vonage.net>;privacy=off;screen=yes
Date: Mon, 10 Aug 2009 18:17:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 302

v=0
o=root 57910148 57910148 IN IP4 208.78.161.230
s=X-PRO Vonage
c=IN IP4 208.78.161.230
t=0 0
m=audio 25574 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


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---------------------------------------------------------------------- 
 (0109371) lmadsen (administrator) - 2009-08-20 14:29
 https://issues.asterisk.org/view.php?id=15688#c109371 
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Also, you should know by now that we need to see the entire SIP capture
(debug), the history, the dialplan involved to reproduce the issue, and the
relevant portions of sip.conf to reproduce the issue. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-08-20 14:29 lmadsen        Note Added: 0109371                          
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