Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two.
Calls between the phones conected to the SIP GW and the ones conected to the MGCP GW goes OK ONLY if I call from the MGCP to the SIP. Phones at MGCP can call without problems to the PSTN (voice quality isn't very good, with silence times, but it can be supported!). But phones at SIP can't do any call! The problem is that when I pick up the callee phone, I don't hear nothing and the call goes off inbetween 4 or 5 seconds. And the caller (SIP) doesn't realise I have picked up, because It's still hearing the calling tone.When the call goes off, the caller hear the congestion tone. I don't know what is the problem!!!! The other problem is that I can't achive to transfer calls. When I dial #, it doesn't happen anything!! And the callerID doesn't work either... My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no [sip] type=friend callerid="sip" <2222> username=sip host=188.208.12.37 accountcode=sip My extensions.conf exten => 2222,1,dial,SIP/[EMAIL PROTECTED]|60|rTt exten => 2222,2,Hangup I also have done a SIP debug and I'm sneding an extract of what I have found. I can't understand why the "out of SIP" messages go to an IP so strange!!! (229...) I can't find this IP anywhre in my system...Any ideas? Hope someone can help!! Thanks in advance! michelle PD:188.208.12.237 is the asterisk IP (...) s=session c=IN IP4 188.208.12.237 t=0 0 =audio 13532 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 229.159.241.112:5060 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-3a5246f7- 8c6b606-10eb From: ;tag=0-13c4-3a5246f7-8c6b604-c3a To: ;tag=as52ed0a6a Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: Content-Type: application/sdp Content-Length: 135 v=0 o=root 11673 11673 IN IP4 188.208.12.237 s=session c=IN IP4 188.208.12.237 t=0 0 =audio 13532 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 229.159.241.112:5060 -- Hungup 'IAX2[test]/1' == Spawn extension (default, 3333, 1) exited non-zero on 'SIP/229.159.241.112:5 060' set_destination: Parsing for address/port to send t o set_destination: set destination to 188.208.12.37, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 188.208.12.237:5060;branch=z9hG4bK6723148d From: ;tag=as52ed0a6a To: ;tag=0-13c4-3a5246f7-8c6b604-c3a Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 188.208.12.37:5060 Sip read: SIP/2.0 200 OK From: To: ;tag=0-13c4-3a5246f7-8c6b604-c3a Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE Via: SIP/2.0/UDP 188.208.12.237:5060 ;received=188.208.12.237 ;branch=z9hG4bK67231 48d Content-Length:0 7 headers, 0 lines Message is BYE ----- Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, M�dem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
