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

[Dave Alan Caruana]

sorry i'm sending so many emails, I always think of something
exactly after i've pressed Send .. please be patient with me :)

I also have OH323 installed, supposedly correctly, and the same
gateway I want to connect to on SIP also supports H323, however
i do not know what the dial command line for H323 is .. i'm trying

exten => 1304,1,Dial(OH323/216.52.153.206) ;ring

but I actually want to dial extension 723 on the remote end,
so this is surely not right.. current messages i'm getting
from Asterisk are these :

*CLI> dial 1304
    -- Executing Dial("OSS/dsp", "OH323/216.52.153.206") in new stack
*CLI>   0:03.623                   H323 Cleaner H323    Connection ip$localhost/9771 terminated.
ERROR[1232188736]: File chan_oh323.c, Line 610 (oh323_call): H323:0: Could not call 216.52.153.206.
    -- Couldn't call 216.52.153.206
    -- Hungup 'H323:0'
  == Everyone is busy at this time

help *very* welcome ;)

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

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