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Thread: One way voice with beta SIP forwarding?

  1. #1
    Join Date
    Mar 2010
    Posts
    4

    Default One way voice with beta SIP forwarding?

    I'm a VOIP Express customer forwarding my old home phone number to my PBX (Asterisk 1.6). When I use the beta SIP support in the forwarding settings of vPanel, I only can hear the callee's voice (from PBX to the caller) and the caller's voice (from caller through VOIPo to PBX) is missing.

    All of my other VOIP providers are working fine with the PBX with voice working in both directions, so I doubt it is a firewall issue.

    Any advice? Is anyone else successfully using the VOIP Express forwarding to a SIP account from an Asterisk 1.6 PBX, or even to another SIP BYOD device?

    Thanks!

  2. #2
    Join Date
    Feb 2007
    Posts
    801

    Default Re: One way voice with beta SIP forwarding?

    Since you're forwarding to a SIP account, it could still be a firewall issue. Have you tried forwarding to a provider you have set up as an incoming trunk on your PBX? You may not want it that way permanently, but it might help you troubleshoot the issue. If you don't have a SIP account you want to use, consider signing up for a FREE (IP-to-IP only) account with CallCentric and see if calls forwarded to that SIP account have two-way audio.

  3. #3
    Join Date
    Mar 2010
    Posts
    4

    Default Re: One way voice with beta SIP forwarding?

    I found the problem: the incoming forwarded Express calls are coming from a server (4.68.250.14 other than sip.voipwelcome.com so I had to canreinvite=no to the [general] section of my asterisk sip.conf in addition to the sip.voipwelcome.com section.

    I now have two way voice via SIP on the forwarded calls.

    Thanks!

  4. #4
    Join Date
    Feb 2007
    Location
    Irvine CA
    Posts
    1,542,128,043

    Default Re: One way voice with beta SIP forwarding?

    Quote Originally Posted by Stormwind View Post
    I found the problem: the incoming forwarded Express calls are coming from a server (4.68.250.14 other than sip.voipwelcome.com so I had to canreinvite=no to the [general] section of my asterisk sip.conf in addition to the sip.voipwelcome.com section.

    I now have two way voice via SIP on the forwarded calls.

    Thanks!
    Glad to hear you're up and running.

    That's one of the differences in our setup vs many others. The audio typically doesn't go through us and is handled directly by the carrier the call is terminating to.
    Timothy Dick
    Founder/CEO
    VOIPo.com

    Interact with VOIPo: Twitter, Facebook

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