| Summary: | ASTERISK-14625: The "port" parameter for an outbound provider is not being respected | ||
| Reporter: | Private Name (falves11) | Labels: | |
| Date Opened: | 2009-08-10 13:32:38 | Date Closed: | 2011-06-07 14:08:02 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| 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@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@sphone.vopr.vonage.net>;tag=as0f1f2595 To: <sip:12564286161@sphone.vopr.vonage.net:5061> Contact: <sip:13106xxxxx@208.78.161.230> Call-ID: 340e75926a10ebcd40e5c7b62127cf47@sphone.vopr.vonage.net CSeq: 102 INVITE User-Agent: X-PRO release 1103g Remote-Party-ID: "13106xxxxx" <sip:13106017395@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 ****** ADDITIONAL INFORMATION ****** As you can see, Asterisk is dialing to port 5061, while in muy peer definition I used port=10000. There is no "5061" word in my whole sip.conf | ||
| Comments: | By: Private Name (falves11) 2009-08-10 13:46:12 Note: I also have a "register=" line with the Vonage information, and it also uses port 10000. So where is it getting the "5061"?. I can give access to my development server to a Bug marshal for quick verification. There is nothing else going on. By: Leif Madsen (lmadsen) 2009-08-20 14:28:28 Please do not continue to mark every issue you open as a blocking issue. By: Leif Madsen (lmadsen) 2009-08-20 14:29:32 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. By: Private Name (falves11) 2009-08-20 14:52:32 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. By: Olle Johansson (oej) 2009-09-03 14:07:32 Please read the bug guidelines and provide us with the information Leif asked for. We need to see your configuration to be able to help. The information you have provided here is not enough. By: Private Name (falves11) 2009-09-03 14:12:20 Please close the case. By: Olle Johansson (oej) 2009-09-03 14:17:05 Closed on reporter's request. | ||