Does Asterisk alter the Headers of INVITE Message

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Asterisk Users 11 Comments

Hi all,
I am trying to send an extra header in SIP INVITE Message , i.e (email=”me@me.com”) but when I check the Message at the target that header is not there
So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message?
Thank u

11 thoughts on - Does Asterisk alter the Headers of INVITE Message

  • Please guys anybody knows How can I send a unique token to the Receiver at the Invite call? Is that possible?

    ________________________________________
    Sent: Sunday, November 27, 2011 11:47 AM

    Hi all,
    I am trying to send an extra header in SIP INVITE Message , i.e (email=”me@me.com”) but when I check the Message at the target that header is not there
    So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message?
    Thank u

  • I tried that with my SIP Cleint but the custom Header is not reaching the cleint … Does the asketrisk delete that?
    ________________________________________
    Sent: Sunday, November 27, 2011 3:30 PM

    Custom SIP headers are a common way to do that. Try SIPAddHeader().

  • Yes, see attached …
    Proxy server alter my “Test” custom header and delete it, Is there a way to include it in message sent from SIP Proxy to target?
    ________________________________________
    Sent: Sunday, November 27, 2011 4:19 PM

    Are you sure? Have you taken a packet capture to confirm?

  • That would be a proxy configuration issue, wouldn’t it?

    In principle, the proxy should be passing these messages through
    unmodified, unless you have an explicit configuration directive that
    instructs it to remove headers from the INVITE.

  • thats my main question if u can see “Does Asterisk alter the Headers of INVITE Message”
    I am using ASterisk NOW proxy …. I didnt configure it to delete anything , Can u tell me how I can change it to pass that parameters?
    thanx
    ________________________________________
    Sent: Sunday, November 27, 2011 4:41 PM

    That would be a proxy configuration issue, wouldn’t it?

    In principle, the proxy should be passing these messages through
    unmodified, unless you have an explicit configuration directive that
    instructs it to remove headers from the INVITE.

  • Any body knows how I can configure Asterisk SIP to pass all Header Parameters?
    ________________________________________
    Sent: Sunday, November 27, 2011 4:50 PM

    thats my main question if u can see “Does Asterisk alter the Headers of INVITE Message”
    I am using ASterisk NOW proxy …. I didnt configure it to delete anything , Can u tell me how I can change it to pass that parameters?
    thanx
    ________________________________________
    Sent: Sunday, November 27, 2011 4:41 PM

    That would be a proxy configuration issue, wouldn’t it?

    In principle, the proxy should be passing these messages through
    unmodified, unless you have an explicit configuration directive that
    instructs it to remove headers from the INVITE.

  • Well, It doesn’t forward the INVITE at all, as asterisk is NOT a proxy. It
    creates a totally new INVITE when you issue the Dial application, with its
    own set of headers.

    Now, you can pass the Test header with something like this (taken from
    memory…):

    SipAddHeader(Test: ${SIP_HEADER(Test)})

    Do that prior to the call to the Dial application, and you will see your
    header in the outgoing INVITE. Of course this means that your dial plan need
    to know which headers to pass.

    // T

  • thank you for ur solution, I did this in dail plan yesterday … it took me 5 hours to find that solution , I wish u replied to me earlier but thanx 🙂
    ________________________________________
    Sent: Monday, November 28, 2011 12:43 AM

    Well, It doesn’t forward the INVITE at all, as asterisk is NOT a proxy. It
    creates a totally new INVITE when you issue the Dial application, with its
    own set of headers.

    Now, you can pass the Test header with something like this (taken from
    memory…):

    SipAddHeader(Test: ${SIP_HEADER(Test)})

    Do that prior to the call to the Dial application, and you will see your
    header in the outgoing INVITE. Of course this means that your dial plan need
    to know which headers to pass.

    // T