search for: ast_get_srv

...plenty) OS: open SUSE 11 Asterisk Version: 1.4.2 Asterisk GUI Version: 2.0 The system was completely set up using the Asterisk GUI with a couple tweaks in users.conf that via:talk wants. Here is what happens: 1. Asterisk verifies connection to the server and we get this. (CLI output) -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net...

...ere. -- Saved useragent "wengo/v1/wengophoneng/wengo/rev12359/trunk/" for peer 2001 -- Executing [100 at my-phones:1] VoiceMail("SIP/2001-08225788", "2000") in new stack -- <SIP/2001-08225788> Playing 'vm-intro' (language 'en') -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 -- <SIP/2001-08225788> Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/tmp/OayHq7 format: wav, 0x821...

...s_ninit' /usr/src/asterisk-cvs/enum.c:307: undefined reference to `__res_nsearch' /usr/src/asterisk-cvs/enum.c:325: undefined reference to `__res_nclose' enum.o: In function `parse_naptr': /usr/src/asterisk-cvs/enum.c:157: undefined reference to `__dn_expand' srv.o: In function `ast_get_srv': /usr/src/asterisk-cvs/srv.c:279: undefined reference to `__res_ninit' /usr/src/asterisk-cvs/srv.c:282: undefined reference to `__res_nsearch' /usr/src/asterisk-cvs/srv.c:297: undefined reference to `__res_nclose' srv.o: In function `parse_srv': /usr/src/asterisk-cvs/srv.c:136:...

...2209-000026d3>AGI Tx >> 200 result=0 <SIP/2209-000026d3>AGI Rx << EXEC Dial SIP/ 00918885268942 at sip.trunk.gradwell.com,60,r -- AGI Script Executing Application: (Dial) Options: (SIP/ 00918885268942 at sip.trunk.gradwell.com,60,r) == Using SIP RTP CoS mark 5 > ast_get_srv: SRV lookup for '_sip._udp.sip.trunk.gradwell.com' mapped to host v-sip-trunk-out-f1.gradwell.net, port 5060 -- Called 00918885268942 at sip.trunk.gradwell.com [Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463 handle_response_invite: Received response: "Forbidden" from '&...

...er Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 -- Accepting AUTHENTICATED call from 127.0.0.1: > requested format = alaw, > requested prefs = (), > actual format = alaw, > host prefs = (alaw), > prior...

...hing else that shows up, but is there a way to make all the dns messages go away? Ira > doing dnsmgr_lookup for 'gw5.telasip.com' > doing dnsmgr_lookup for 'sipconnect.ipcomms.net' > doing dnsmgr_lookup for 'proxy.ideasip.com' > ast_get_srv: SRV lookup for '_sip._UDP.proxy.ideasip.com' mapped to host proxy.ideasip.com, port 5060

...("Zap/1-1", "we are at fax") in new stack -- Executing [s at fax:3] Dial("Zap/1-1", "ZAP/2") in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 Thanks for helping out. I really appreciate it. Thanks, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/...

...he connection to ares.sip-happens.com was being refused there was no roll-over to sometimes.sip-happens. Here's what asterisk did: -- Executing [s at macro-enumdial:23] Dial("SIP/anonymous-b5e02fd0", "SIP/18771234567 at tollfree.sip-happens.com.||") in new stack -- ast_get_srv: SRV lookup for '_sip._udp.tollfree.sip-happens.com.' mapped to host ares.sip-happens.com, port 5070 -- Called 18771234567 at tollfree.sip-happens.com. [Oct 17 10:15:46] NOTICE[4973]: chan_sip.c:2920 auto_congest: Auto-congesting SIP/tollfree.sip-happens.com.-081ddc28 -- SIP/tollfre...