Re: [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: [EMAIL PROTECTED] 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/[EMAIL PROTECTED],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: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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/[EMAIL PROTECTED]") in new stack     -- Called [EMAIL PROTECTED]     -- 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] for seqno 1 (Response)     -- Executing Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new stack     -- Called [EMAIL PROTECTED]     -- 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] for seqno 102 (Request)     -- Executing Dial("SIP/217.168.168.49:5060", "SIP/[EMAIL PROTECTED]") in new stack     -- Called [EMAIL PROTECTED]     -- 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] for seqno 102 (Request) ----- Original Message ----- From: Dan To: [EMAIL PROTECTED] 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: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]   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/[EMAIL PROTECTED])   When from my softphone I dial sip:[EMAIL PROTECTED]   on the console I get :     -- Executing Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new stack     -- Called [EMAIL PROTECTED]     -- 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