[Asterisk-Users] a beginner's SIP question ..
Michael Manousos
manousos at inaccessnetworks.com
Tue Jun 3 07:21:39 MST 2003
Hi,
Dave Alan Caruana wrote:
> 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,
First, make sure to specify a codec type, in oh323.conf, that is
supported by the gateway.
If a gatekeeper is used and the gateway and Asterisk are
registered on this gatekeeper, then you should do:
exten => 1304,1,Dial(OH323/723)
If there is no gatekeeper involved, do:
exten => 1304,1,Dial(OH323/723 at 216.52.153.206)
> 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
Michael.
>
> ----- Original Message -----
> *From:* Dan <mailto:dtoma at fx.ro>
> *To:* asterisk-users at lists.digium.com
> <mailto: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
> <mailto: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 <mailto:david at melita.net>
> *To:* asterisk-users at lists.digium.com
> <mailto: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 <mailto: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 <mailto:SIP/723 at 216.52.153.207>") in new
> stack
> -- Called 723 at 216.52.153.207 <mailto: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
> <mailto: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
> <mailto: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 <mailto:SIP/723 at 216.52.153.207>") in new
> stack
> -- Called 723 at 216.52.153.207 <mailto: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
> <mailto: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
> <mailto: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 <mailto:SIP/723 at 216.52.153.207>") in new
> stack
> -- Called 723 at 216.52.153.207 <mailto: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
> <mailto: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
> <mailto:call-1054307890-9 at 217.168.168.49> for seqno 102 (Request)
>
> ----- Original Message -----
> *From:* Dan <mailto:dtoma at fx.ro>
> *To:* asterisk-users at lists.digium.com
> <mailto: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 <mailto:david at melita.net>
> *To:* asterisk-users at lists.digium.com
> <mailto: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
> <mailto: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
> <mailto:SIP/723 at 216.52.153.207>") in new stack
> -- Called 723 at 216.52.153.207 <mailto: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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