Re: [asterisk-users] DID is not working (call is not routing)
Hello Chandra, What about Teliax´s service? Is it recommended? How´s their call quality? Thanks in advance... On 10/10/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi William,My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your valuable time for me. Regards,Chandra.William Piper [EMAIL PROTECTED] wrote: Your server seems to be doing exactly what you are telling it to do: -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that The number you have dialed is not inservice. Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf(SIP/216.89.79.2 -09e1d020, 0?from-trunk||1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack -- Executing Wait(SIP/216.89.79.2-09e1d020, 2) in new stack -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion(SIP/216.89.79.2-09e1d020, ) in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp(SIP/216.89.79.2-09e1d020, Hangup) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, DID=s) in new stack -- Executing Goto(SIP/216.89.79.2-09e1d020, s|1) in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf(SIP/216.89.79.2-09e1d020, 0?from-trunk|s|1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Stay in the know. Pulse on the new Yahoo.com. Check it out. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] DID is not working (call is not routing)
Hi Libera,We have an account with Teliax from 7 months. Teliax's service is very good and giving excellent customer support also. But, I observed the below things from Teliax's people.1) Let us assume that you have configured your Teliax account settings with XLite or any other sofphone directly without using Trixbox or Asterisk. After that, if you are facing any problem, they are solving.2) If you configure Teliax account settings with Asterisk or Trixbox, they are facing trouble to solve some technical problems from Trixbox or Asterisk point of view3) Voice quality is very good.Thank you.Regards,Chandra."R.R Libera" [EMAIL PROTECTED] wrote: Hello Chandra, What about Teliax´s service? Is it recommended? How´s their call quality? Thanks in
advance... On 10/10/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi William,My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your valuable time for me. Regards,Chandra.William Piper [EMAIL PROTECTED] wrote:Your server seems to be doing exactly what you are telling it
to do: -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created
Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf("SIP/216.89.79.2 -09e1d020", "0?from-trunk||1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing
'ss-noservice' (language 'en') -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 3) exited
non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Stay in the know. Pulse on the new Yahoo.com. Check it out. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] DID is not working (call is not routing)
Hi William,Thank you for response. Sorry. I forgot to say that am configuring using Trixbox. Can you tell me the solution? Thank you.Regards,Chandra,William Piper [EMAIL PROTECTED] wrote: Your server seems to be doing exactly what you are telling it to do: -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy
Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf("SIP/216.89.79.2 -09e1d020", "0?from-trunk||1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing
Answer("SIP/216.89.79.2-09e1d020", "") in new stack -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack -- Executing
Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.
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Re: [asterisk-users] DID is not working (call is not routing)
No idea, I've never used Trixbox. I believe they have a support forum though... bp On 10/9/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi William,Thank you for response. Sorry. I forgot to say that am configuring using Trixbox. Can you tell me the solution? Thank you. Regards,Chandra, William Piper [EMAIL PROTECTED] wrote: Your server seems to be doing exactly what you are telling it to do: -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that The number you have dialed is not inservice. Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf(SIP/216.89.79.2 -09e1d020, 0?from-trunk||1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack -- Executing Wait(SIP/216.89.79.2-09e1d020, 2) in new stack -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion(SIP/216.89.79.2-09e1d020, ) in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp(SIP/216.89.79.2-09e1d020, Hangup) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, DID=s) in new stack -- Executing Goto(SIP/216.89.79.2-09e1d020, s|1) in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf(SIP/216.89.79.2-09e1d020, 0?from-trunk|s|1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call rates. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] DID is not working (call is not routing)
Hi William,My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your valuable time for me. Regards,Chandra.William Piper [EMAIL PROTECTED] wrote: Your server seems to be doing exactly what you are telling it to do: -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf("SIP/216.89.79.2 -09e1d020", "0?from-trunk||1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC.
-- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack --
Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.
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[asterisk-users] DID is not working (call is not routing)
Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully.When I am making a call to my DID number from outside, its telling that "The number you have dialed is not inservice". Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk||1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing Answer("SIP/216.89.79.2-09e1d020", "") in new stack -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack -- Executing
Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp("SIP/216.89.79.2-09e1d020", "Hangup") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack -- Executing Goto("SIP/216.89.79.2-09e1d020", "s|1") in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "0?from-trunk|s|1") in new stack -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing
Answer("SIP/216.89.79.2-09e1d020", "") in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020'When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra.
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Re: [asterisk-users] DID is not working (call is not routing)
Your server seems to be doing exactly what you are telling it to do: -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that The number you have dialed is not inservice. Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf(SIP/216.89.79.2 -09e1d020, 0?from-trunk||1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack -- Executing Wait(SIP/216.89.79.2-09e1d020, 2) in new stack -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion(SIP/216.89.79.2-09e1d020, ) in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp(SIP/216.89.79.2-09e1d020, Hangup) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, DID=s) in new stack -- Executing Goto(SIP/216.89.79.2-09e1d020, s|1) in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf(SIP/216.89.79.2-09e1d020, 0?from-trunk|s|1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
