[Asterisk-Users] a beginner's SIP question .. (further!)

[Dave Alan Caruana]

more about the same problem ...
i've been playing around and got to this error message which seems relevant ..

*CLI> dial 1303
    -- Executing Dial("OSS/dsp", "SIP/723 at 216.52.153.207") in new stack
    -- Called 723 at 216.52.153.207
    -- SIP/216.52.153.207-1fb9 answered OSS/dsp
 << Console call has been answered >>
NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 received
NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 received
NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 received
NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 received
Killed

am I right in thinking i need a different codec to connect to the sip host I want to
connect to? where do codecs come from?

many cheers
Dave

  ----- Original Message ----- 
  From: Dan 
  To: asterisk-users at lists.digium.com 
  Sent: Friday, May 30, 2003 7:50 PM
  Subject: Re: [Asterisk-Users] a beginner's SIP question ..


  Hi Dave,

  If you have registered the SIP phone with Asterisk, then you must have a line like:

  exten => 555,1,dial(SIP/723 at 216,52,153.207)

  in extensions.conf file

  Then call 555 from the SIP phone to access the destination.

  BR,
  Dan
    ----- Original Message ----- 
    From: Dave Alan Caruana 
    To: asterisk-users at lists.digium.com 
    Sent: Friday, May 30, 2003 6:21 PM
    Subject: Re: [Asterisk-Users] a beginner's SIP question ..


    I have included a dump of the debug info ...
    what I am trying to do is route a call from sipphone 217.168.168.49
    through asterisk 217.168.168.51 onto a gateway 723 at 216.52.153.207
    If i dial direct from the sip phone to the gateway it works fine .. so 
    I do not think there is any incompatibility there.
    Calls don't go through though ...

    please help!!!

    cheers
    Dave


    *CLI>     -- Executing Dial("SIP/217.168.168.49:5060", "SIP/723 at 216.52.153.207") in new stack
        -- Called 723 at 216.52.153.207
        -- SIP/216.52.153.207-eca2 answered SIP/217.168.168.49:5060
        -- Attempting native bridge of SIP/217.168.168.49:5060 and SIP/216.52.153.207-eca2
    WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call call-1054307890-9 at 217.168.168.49 for seqno 1 (Response)
      == Spawn extension (default, 1303, 1) exited non-zero on 'SIP/217.168.168.49:5060'
    WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call call-1054307890-9 at 217.168.168.49 for seqno 1 (Response)
        -- Executing Dial("SIP/217.168.168.49:5060", "SIP/723 at 216.52.153.207") in new stack
        -- Called 723 at 216.52.153.207
        -- SIP/216.52.153.207-1418 answered SIP/217.168.168.49:5060
        -- Attempting native bridge of SIP/217.168.168.49:5060 and SIP/216.52.153.207-1418
    WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call call-1054307890-9 at 217.168.168.49 for seqno 1 (Response)
      == Spawn extension (default, 1303, 1) exited non-zero on 'SIP/217.168.168.49:5060'
    WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call call-1054307890-9 at 217.168.168.49 for seqno 102 (Request)
        -- Executing Dial("SIP/217.168.168.49:5060", "SIP/723 at 216.52.153.207") in new stack
        -- Called 723 at 216.52.153.207
        -- SIP/216.52.153.207-11ed answered SIP/217.168.168.49:5060
        -- Attempting native bridge of SIP/217.168.168.49:5060 and SIP/216.52.153.207-11ed
    WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call call-1054307890-9 at 217.168.168.49 for seqno 1 (Response)
      == Spawn extension (default, 1303, 1) exited non-zero on 'SIP/217.168.168.49:5060'
    WARNING[1125329600]: File chan_sip.c, Line 404 (retrans_pkt): Maximum retries exceeded on call call-1054307890-9 at 217.168.168.49 for seqno 102 (Request)

      ----- Original Message ----- 
      From: Dan 
      To: asterisk-users at lists.digium.com 
      Sent: Thursday, May 29, 2003 8:15 PM
      Subject: Re: [Asterisk-Users] a beginner's SIP question ..


      Hi,

      Check to have a common set of codecs.
      If X-Lite is used and at the other end is a phone without GSM support, then it doesn't work.
      Try to disable GSM on the soft phone (if X-Lite).

      BR,
      Dan


        ----- Original Message ----- 
        From: Dave Alan Caruana 
        To: asterisk-users at lists.digium.com 
        Sent: Thursday, May 29, 2003 9:01 PM
        Subject: [Asterisk-Users] a beginner's SIP question ..


        I am trying to get asterisk to dial this address :
        sip:723 at 216.52.153.207

        Using a softphone on my PC (217.168.168.49)
        it dials immediately and I get a voice prompt ..

        I have configured an extension, 1303 on asterisk,
        modifying the demo configuration :

        exten => 1303,1,Dial(SIP/723 at 216.52.153.207)

        When from my softphone I dial
        sip:1303 at 217.168.168.51

        on the console I get :
            -- Executing Dial("SIP/sipphone-97b6", "SIP/723 at 216.52.153.207") in new stack
            -- Called 723 at 216.52.153.207
            -- SIP/216.52.153.207-7c3b answered SIP/sipphone-97b6
            -- Attempting native bridge of SIP/sipphone-97b6 and SIP/216.52.153.207-7c3b

        but on my headset all I get is silence .. the call doesn't drop though.

        What am I doing wrong ?

        many thanks,
        Dave

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030603/4b927d54/attachment.htm