[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:                     feedback
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 14:52 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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 (0109377) falves11 (reporter) - 2009-08-20 14:52
 https://issues.asterisk.org/view.php?id=15688#c109377 
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There is no dialplan, strictu-sensu, away from

Dial(SIP/peer/${EXTEN})
Hangup()

in the peer definition there is a "port=1000" line, which is not being
respected, as you may see in the SIP trace. 
Please let me know if you can log into my server or you can setup a simple
test bed to replicate the issue. It is a blocking issue because some
carriers, like Vonage, only allow ports for inbound that are not 5060 or
5061. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-08-20 14:52 falves11       Note Added: 0109377                          
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