[Asterisk-Users] a beginner's SIP question ..

[Dave Alan Caruana]

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

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