[asterisk-bugs] [JIRA] (ASTERISK-27643) SIP INVITEs to non-5060 port still use 5060 (Outbound Calls failing)

Tue Jan 30 15:07:13 CST 2018

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27643?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=241916#comment-241916 ] 

Asterisk Team commented on ASTERISK-27643:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> SIP INVITEs to non-5060 port still use 5060 (Outbound Calls failing)
> --------------------------------------------------------------------
>
>                 Key: ASTERISK-27643
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27643
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.7.0
>            Reporter: nate55
>            Severity: Critical
>
> I'm working with a SIP provider that requires me to communicate with them over Port 5070, instead of 5060. My listening port is still 5060 (using chan_sip), with the SIP trunk configuration having port=5070, outboundproxyport=5070, and 5070 in my registration string. I'm able to register successfully, and I can see SIP Options being sent to 5070 and getting responses there, as well.
> The problem is when I try to place an outbound call, the first SIP invite says it's sending over port 5070, and then the subsequent invites (the retries) say they are going over port 5060. I don't know if this is the expected behavior. I would think that the retries shouldn't just ignore the port specified in the configuration. 
> However, after countless hours of testing with the SIP Provider, we have determined that even the initial invite is NOT being sent over port 5070, despite the CLI saying "Reliably Transmitting (NAT) to {SIP_HOST_IP}:5070". I was able to confirm this by running a tcpdump on port 5070, and seeing that no packets were captured when I placed the call. Running the same command with port 5060, showed the invites that were supposed to be going out on 5070.
> SIP INVITES BELOW:
> Reliably Transmitting (NAT) to {SIP_HOST_IP}:5070:
> INVITE sip:{CalledNumber}@{SIP_HOST_IP}:5070 SIP/2.0
> Via: SIP/2.0/UDP {MY_EXTERNAL_IP}:5060;branch=z9hG4bK0e006357;rport
> Max-Forwards: 70
> From: <sip:{BTN_USERNAME}@{SIP_HOST_IP}>;tag=as3a3d5a2e
> To: <sip:{CalledNumber}@{SIP_HOST_IP}:5070>
> Contact: <sip:{BTN_USERNAME}@{MY_EXTERNAL_IP}:5060>
> Call-ID: 5e8e26990132748c6b5b3d083a6db2cf@{SIP_HOST_IP}
> CSeq: 102 INVITE
> User-Agent: FPBX-12.0.1rc28(11.7.0)
> Date: Tue, 30 Jan 2018 20:25:46 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Remote-Party-ID: "3476479119" <sip:3476479119@{SIP_HOST_IP}>;party=calling;privacy=off;screen=no
> Content-Type: application/sdp
> Content-Length: 235
> v=0
> o=root 2027027818 2027027818 IN IP4 {MY_EXTERNAL_IP}
> s=Asterisk PBX 11.7.0
> c=IN IP4 {MY_EXTERNAL_IP}
> t=0 0
> m=audio 17916 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> Retransmitting #1 (NAT) to {SIP_HOST_IP}:5060:
> INVITE sip:{CalledNumber}@{SIP_HOST_IP}:5070 SIP/2.0
> Via: SIP/2.0/UDP {MY_EXTERNAL_IP}:5060;branch=z9hG4bK0e006357;rport
> Max-Forwards: 70
> From: <sip:{BTN_USERNAME}@{SIP_HOST_IP}>;tag=as3a3d5a2e
> To: <sip:{CalledNumber}@{SIP_HOST_IP}:5070>
> Contact: <sip:{BTN_USERNAME}@{MY_EXTERNAL_IP}:5060>
> Call-ID: 5e8e26990132748c6b5b3d083a6db2cf@{SIP_HOST_IP}
> CSeq: 102 INVITE
> User-Agent: FPBX-12.0.1rc28(11.7.0)
> Date: Tue, 30 Jan 2018 20:25:46 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Remote-Party-ID: "3476479119" <sip:3476479119@{SIP_HOST_IP}>;party=calling;privacy=off;screen=no
> Content-Type: application/sdp
> Content-Length: 235
> v=0
> o=root 2027027818 2027027818 IN IP4 {MY_EXTERNAL_IP}
> s=Asterisk PBX 11.7.0
> c=IN IP4 {MY_EXTERNAL_IP}
> t=0 0
> m=audio 17916 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> ---
> Retransmitting #2 (NAT) to {SIP_HOST_IP}:5060:
> INVITE sip:{CalledNumber}@{SIP_HOST_IP}:5070 SIP/2.0
> Via: SIP/2.0/UDP {MY_EXTERNAL_IP}:5060;branch=z9hG4bK0e006357;rport
> Max-Forwards: 70
> From: <sip:{BTN_USERNAME}@{SIP_HOST_IP}>;tag=as3a3d5a2e
> To: <sip:{CalledNumber}@{SIP_HOST_IP}:5070>
> Contact: <sip:{BTN_USERNAME}@{MY_EXTERNAL_IP}:5060>
> Call-ID: 5e8e26990132748c6b5b3d083a6db2cf@{SIP_HOST_IP}
> CSeq: 102 INVITE
> User-Agent: FPBX-12.0.1rc28(11.7.0)
> Date: Tue, 30 Jan 2018 20:25:46 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Remote-Party-ID: "3476479119" <sip:3476479119@{SIP_HOST_IP}>;party=calling;privacy=off;screen=no
> Content-Type: application/sdp
> Content-Length: 235
> v=0
> o=root 2027027818 2027027818 IN IP4 {MY_EXTERNAL_IP}
> s=Asterisk PBX 11.7.0
> c=IN IP4 {MY_EXTERNAL_IP}
> t=0 0
> m=audio 17916 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv

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