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

[Dan]

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
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