Re: [Asterisk-Users] a beginner's SIP question .. (further!)

[Dan]

Hi,   Check the available codecs at the both ends. Keep in mind that Asterisk can convert only between G.711 and GSM (without any extra codec installed. like G.729). Keep in mind too that the Microsoft GSM codec (the one used for example in Netmeeting) is not compatible with the one used by Asterisk or other products. like X-Lite software SIP phone.   BR, Dan   ----- Original Message ----- From: Dave Alan Caruana To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 3:09 PM Subject: Re: [Asterisk-Users] a beginner's SIP question .. (further!) more about the same problem ... i've been playing around and got to this error message which seems relevant ..   *CLI> dial 1303     -- Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]") in new stack     -- Called [EMAIL PROTECTED]     -- SIP/216.52.153.207-1fb9 answered OSS/dsp  << Console call has been answered >> NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 received NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 received NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 received NOTICE[1232188736]: File rtp.c, Line 326 (ast_rtp_read): Unknown RTP codec 19 received Killed am I right in thinking i need a different codec to connect to the sip host I want to connect to? where do codecs come from?   many 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