Re: [asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
Indeed. The problem was the ). thanks to all who helped me debug this...my eyes are not so young anymore... On 2/3/07, jacobso1 [EMAIL PROTECTED] wrote: hi, i think the problem is here : exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to) | replace with exten = _321[0123],n,Dial(SIP/${EXTEN},30,to) note, i removed the parenthesis ')' after the {EXTEN} this should do regards, jacobson --- Scarlet ONE - Combine ADSL with unlimited fixed phone and save 400 euros http://www.scarlet.be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
The following strange conditions is happening while I try to dial a SIP user from another SIp user. SIP to Zap dialing is fine, as all 4 users can call PSTN. I'm using Asterisk SVN-branch-1.2-r51359M Example: extension 3210 calls extension 3213. They are all registered properly: chrom01*CLI sip show peers Name/username HostDyn Nat ACL Port Status 3213/3213 192.168.0.112D 5060 Unmonitored 3212/3212 192.168.0.112D 5060 Unmonitored 3211/3211 192.168.0.112D 5060 Unmonitored 3210/3210 192.168.0.112D 5060 Unmonitored 4 sip peers [4 online , 0 offline] -- Executing Ringing(SIP/3210-084eaa80, ) in new stack -- Executing AGI(SIP/3210-084eaa80, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/3210-084eaa80, SIP/3213)|30|to) in new stack Feb 3 12:42:25 WARNING[10368]: chan_sip.c:1994 create_addr: No such host: 3213) Feb 3 12:42:25 NOTICE[10368]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) **sip.conf*** ** i have 4 extensions, 3210,3211,3212 and 3213. they are all defined in sip.conf with the following parameters (just change 3212 for the next extension and so on). [3212] username=3212 secret=3212 type=friend context=default nat=no canreinvite=no [EMAIL PROTECTED] disallow=all allow=ulaw host=dynamic language=en dtmfmode=inband My dial plan is like this: The AGI is doing nothing more than simple call logging to MySQL **extensions.conf** ** exten = _321[0123],1,Ringing exten = _321[0123],n,AGI(agi://127.0.0.1:4577/call_log) exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to) exten = _321[0123],n,Voicemail,u${EXTEN} exten = _321[0123],n,Hangup comments? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
The fact that all of the phones have the same 'host' is not a good sign. Also - turn 'qualify' on. It really helps with phone status. PaulH On Sat, 2007-02-03 at 12:50 -0500, Erick Perez wrote: The following strange conditions is happening while I try to dial a SIP user from another SIp user. SIP to Zap dialing is fine, as all 4 users can call PSTN. I'm using Asterisk SVN-branch-1.2-r51359M Example: extension 3210 calls extension 3213. They are all registered properly: chrom01*CLI sip show peers Name/username HostDyn Nat ACL Port Status 3213/3213 192.168.0.112D 5060 Unmonitored 3212/3212 192.168.0.112D 5060 Unmonitored 3211/3211 192.168.0.112D 5060 Unmonitored 3210/3210 192.168.0.112D 5060 Unmonitored 4 sip peers [4 online , 0 offline] -- Executing Ringing(SIP/3210-084eaa80, ) in new stack -- Executing AGI(SIP/3210-084eaa80, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/3210-084eaa80, SIP/3213)|30|to) in new stack Feb 3 12:42:25 WARNING[10368]: chan_sip.c:1994 create_addr: No such host: 3213) Feb 3 12:42:25 NOTICE[10368]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) **sip.conf*** ** i have 4 extensions, 3210,3211,3212 and 3213. they are all defined in sip.conf with the following parameters (just change 3212 for the next extension and so on). [3212] username=3212 secret=3212 type=friend context=default nat=no canreinvite=no [EMAIL PROTECTED] disallow=all allow=ulaw host=dynamic language=en dtmfmode=inband My dial plan is like this: The AGI is doing nothing more than simple call logging to MySQL **extensions.conf** ** exten = _321[0123],1,Ringing exten = _321[0123],n,AGI(agi://127.0.0.1:4577/call_log) exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to) exten = _321[0123],n,Voicemail,u${EXTEN} exten = _321[0123],n,Hangup comments? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
No such host: 3213) Look for an extra closing parenthesis in your Dial command: Dial(SIP/3210-084eaa80, SIP/3213)|30|to) It should be SIP/3213 rather than SIP/3213). --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
hi, i think the problem is here : exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to) | replace with exten = _321[0123],n,Dial(SIP/${EXTEN},30,to) note, i removed the parenthesis ')' after the {EXTEN} this should do regards, jacobson --- Scarlet ONE - Combine ADSL with unlimited fixed phone and save 400 euros http://www.scarlet.be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users