Re: [asterisk-users] sip error logging
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Re: [asterisk-users] Jabber / facebook chat?
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Re: [asterisk-users] Jabber / GTalk / hints
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Re: [asterisk-users] dialplan is not finding my number asterisk1.8.3
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Re: [asterisk-users] CDR MYSQL missing field data
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Re: [asterisk-users] Act! Integration
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Re: [asterisk-users] Asterisk PRI back-to-back connect
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Re: [asterisk-users] How to use Atxfer in AMI
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Re: [asterisk-users] Multi-Tenant Hosted PBX system with Resellerfunctionality
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Re: [asterisk-users] Play different voice-mail messages based oncertain conditions
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Re: [asterisk-users] Call are established, but voices can't be heard
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Re: [asterisk-users] Testing from where number is...
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Re: [asterisk-users] SIP Provider Recommendation in US
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Re: [asterisk-users] FAX on PRI to MFCR2
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Re: [asterisk-users] Avoided deadlock Error(solved)
The proble is dialplan I configure fine -- Sent from my BlackBerry® VoIP, Windows/Linux Administration and Network Management US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -Original Message- From: Stefan Schmidt s...@sil.at Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 24 Nov 2010 22:59:56 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Avoided deadlock Error Am 24.11.2010 13:48, schrieb Bayardo Sanchez: My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem is : Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x861f6d8', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85a6420', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85bc510', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85f9e68', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85e1db0', 9 retries! this error comes only when I call spain saturated my CLI with the message error hello, as tilghman noticed 1.2 is EOL, but i still use it too and i see a bunch of this messages on different servers and they dont cause any problem at all. if you have some problems with this (except the warning message) you should upgrade. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avoided deadlock Error
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem is : Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x861f6d8', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85a6420', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85bc510', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85f9e68', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85e1db0', 9 retries! this error comes only when I call spain saturated my CLI with the message error -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avoided deadlock Error
Othe problem is small time my hdd is full of recording -- Sent from my BlackBerry® VoIP, Windows/Linux Administration and Network Management US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -Original Message- From: Stefan Schmidt s...@sil.at Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 24 Nov 2010 22:59:56 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Avoided deadlock Error Am 24.11.2010 13:48, schrieb Bayardo Sanchez: My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem is : Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x861f6d8', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85a6420', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85bc510', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85f9e68', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85e1db0', 9 retries! this error comes only when I call spain saturated my CLI with the message error hello, as tilghman noticed 1.2 is EOL, but i still use it too and i see a bunch of this messages on different servers and they dont cause any problem at all. if you have some problems with this (except the warning message) you should upgrade. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk sip attack
This week I was experiencing attacks sip log into my accounts were more than 1,000 requests for records Sip accounts in less than an hour THROUGH deny the ip of my router access list in cisco and my asterisk server to go through the iptables drop ip attacker is a way for an account with another ip can not log into my asterisk server to add some command in my sip.conf for deny register account sip in my asterisk? -- Sent from my BlackBerry® VoIP, Windows/Linux Administration and Network Management US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callcenter open source program
I'm working in callcenter I using vicidial is cool -- Sent from my BlackBerry® VoIP, Windows/Linux Administration and Network Management -Original Message- From: wassim darwich wassimdarwi...@yahoo.com Date: Sun, 7 Mar 2010 06:21:34 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Callcenter open source program -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] kill sip user
I have a user but I need to give that user only kill and disable all connection cut calls what is the command in the CLIC -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
are using some kind of router? On Sat, Sep 26, 2009 at 8:20 AM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Hello I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] i have a error in ivr
i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug 14 08:15:22 WARNING[25931]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/74.63.41.218-b6036ae0 for procall3 -- Executing Playback(SIP/74.63.41.218-b6036ae0, procall3) in new stack Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug 14 08:15:22 WARNING[25931]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/74.63.41.218-b6036ae0 for procall3 -- Executing Queue(SIP/74.63.41.218-b6036ae0, procall|n|||) in new stack -- Started music on hold, class 'default', on SIP/74.63.41.218-b6036ae0 -- Called SIP/101 -- Called SIP/103 -- SIP/103-09142868 is ringing -- SIP/101-090fb6a0 is ringing -- Stopped music on hold on SIP/74.63.41.218-b6036ae0 == Spawn extension (trunkinbound, 651085, 3) exited non-zero on 'SIP/74.63.41.218-b6036ae0' other erro is when i call to my tollfree number is rining 2 extencion the 101 and 103 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] i have a error in ivr
the audio is in format wav i save in Format PCM attributte 8,000 KHz; 8bit; Mono 7kb/s my extension.conf is the next: exten = 651085,1,Playback(procall3) exten = 651085,n,Playback(procall3) exten = 651085,n,Queue(procall|n|||) exten = 651085,n,Playback(voicemail-invitation) exten = 651085,n,VoiceMail,111 exten = 651085,n,Hangup [procall] exten = s,1,Set(TIMEOUT(digit)=7) ; exten = s,2,Set(TIMEOUT(response)=10) exten = s,3,Set(CHANNEL(language)=en) ; define como idioma predefinido el ingles y usas las voces en este idioma exten = s,4,BackGround(custom/procall3) ; presenta en menu en ingles exten = s,5,WaitExten() ; Espera que el llamante presione una tecla exten = 100,1,Dial(SIP/100,45,r) exten = 101,2,Dial(SIP/101,45,r) exten = 101,3,Dial(SIP/103,45,r) exten = i,1,Playback(invalid) exten = i,2,Playback(goodbye) exten = i,3,hangup exten = t,1,goto(procall,s,1) exten = h,1,Hangup and my queues.conf is: [procall] music=default strategy=ringall timeout=15 retry = 5 monitor-format = wav monitor-join = yes joinempty = yes member = SIP/100 member = SIP/101 member = SIP/103 tnx for u help On Fri, Aug 14, 2009 at 9:06 AM, Danny Nicholas da...@debsinc.com wrote: Playback is expecting a frequency of 8000. use sox to correct. As for 101/103, that is how the dialplan is written, not an error per se. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Friday, August 14, 2009 9:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] i have a error in ivr i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug 14 08:15:22 WARNING[25931]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/74.63.41.218-b6036ae0 for procall3 -- Executing Playback(SIP/74.63.41.218-b6036ae0, procall3) in new stack Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug 14 08:15:22 WARNING[25931]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/74.63.41.218-b6036ae0 for procall3 -- Executing Queue(SIP/74.63.41.218-b6036ae0, procall|n|||) in new stack -- Started music on hold, class 'default', on SIP/74.63.41.218-b6036ae0 -- Called SIP/101 -- Called SIP/103 -- SIP/103-09142868 is ringing -- SIP/101-090fb6a0 is ringing -- Stopped music on hold on SIP/74.63.41.218-b6036ae0 == Spawn extension (trunkinbound, 651085, 3) exited non-zero on 'SIP/74.63.41.218-b6036ae0' other erro is when i call to my tollfree number is rining 2 extencion the 101 and 103 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you
Re: [asterisk-users] i have a error in ivr
yes procall3 is in /var/lib/asterisk/sounds/procall3/wav erase these: exten = i,1,Playback(invalid) exten = i,2,Playback(goodbye) exten = i,3,hangup exten = t,1,goto(procall,s,1) exten = h,1,Hangup ? On Fri, Aug 14, 2009 at 9:56 AM, Danny Nicholas da...@debsinc.com wrote: Procall3 is /var/lib/asterisk/sounds/procall3.wav? IMO, procall should look like this: [procall] exten = s,1,Set(TIMEOUT(digit)=7) ; exten = s,2,Set(TIMEOUT(response)=10) exten = s,3,Set(CHANNEL(language)=en) ; define como idioma predefinido el ingles y usas las voces en este idioma exten = s,4,BackGround(custom/procall3) ; presenta en menu en ingles exten = s,5,WaitExten() ; Espera que el llamante presione una tecla exten = 100,1,Dial(SIP/100,45,r) exten = 100,2,Dial(SIP/101,45,r) exten = 100,3,Dial(SIP/103,45,r) exten = i,1,Playback(invalid) exten = i,2,Playback(goodbye) exten = i,3,hangup exten = t,1,goto(procall,s,1) exten = h,1,Hangup -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Friday, August 14, 2009 10:20 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] i have a error in ivr the audio is in format wav i save in Format PCM attributte 8,000 KHz; 8bit; Mono 7kb/s my extension.conf is the next: exten = 651085,1,Playback(procall3) exten = 651085,n,Playback(procall3) exten = 651085,n,Queue(procall|n|||) exten = 651085,n,Playback(voicemail-invitation) exten = 651085,n,VoiceMail,111 exten = 651085,n,Hangup [procall] exten = s,1,Set(TIMEOUT(digit)=7) ; exten = s,2,Set(TIMEOUT(response)=10) exten = s,3,Set(CHANNEL(language)=en) ; define como idioma predefinido el ingles y usas las voces en este idioma exten = s,4,BackGround(custom/procall3) ; presenta en menu en ingles exten = s,5,WaitExten() ; Espera que el llamante presione una tecla exten = 100,1,Dial(SIP/100,45,r) exten = 101,2,Dial(SIP/101,45,r) exten = 101,3,Dial(SIP/103,45,r) exten = i,1,Playback(invalid) exten = i,2,Playback(goodbye) exten = i,3,hangup exten = t,1,goto(procall,s,1) exten = h,1,Hangup and my queues.conf is: [procall] music=default strategy=ringall timeout=15 retry = 5 monitor-format = wav monitor-join = yes joinempty = yes member = SIP/100 member = SIP/101 member = SIP/103 tnx for u help On Fri, Aug 14, 2009 at 9:06 AM, Danny Nicholas da...@debsinc.com wrote: Playback is expecting a frequency of 8000. use sox to correct. As for 101/103, that is how the dialplan is written, not an error per se. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Friday, August 14, 2009 9:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] i have a error in ivr i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug 14 08:15:22 WARNING[25931]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/74.63.41.218-b6036ae0 for procall3 -- Executing Playback(SIP/74.63.41.218-b6036ae0, procall3) in new stack Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug 14 08:15:22 WARNING[25931]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/74.63.41.218-b6036ae0 for procall3 -- Executing Queue(SIP/74.63.41.218-b6036ae0, procall|n|||) in new stack -- Started music on hold, class 'default', on SIP/74.63.41.218-b6036ae0 -- Called SIP/101 -- Called SIP/103 -- SIP/103-09142868 is ringing -- SIP/101-090fb6a0 is ringing -- Stopped music on hold on SIP/74.63.41.218-b6036ae0 == Spawn extension (trunkinbound, 651085, 3) exited non-zero on 'SIP/74.63.41.218-b6036ae0' other erro is when i call to my tollfree number is rining 2 extencion the 101 and 103 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which
[asterisk-users] inbound filed
i create inbound confi my confi is: [incoming] exten= 1246463,,1,Dial(SIP/8003,60,rT) exten= 6463,1,Dial(SIP/8003,60,rT) exten= 1246463,,n,Wait(5) exten= 1246463,,n,Hangup but y calling and send this error in my CLI: [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inbound filed
nothing send the error: [Apr 15 10:30:54] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found On Wed, Apr 15, 2009 at 10:11 AM, Jared Smith jsm...@digium.com wrote: On Wed, 2009-04-15 at 09:59 -0600, Bayardo Sanchez wrote: i create inbound confi my confi is: [incoming] exten= 1246463,,1,Dial(SIP/8003,60,rT) exten= 6463,1,Dial(SIP/8003,60,rT) exten= 1246463,,n,Wait(5) exten= 1246463,,n,Hangup but y calling and send this error in my CLI: [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. It appears that you have some extra commas in your configuration. Try: [incoming] exten= 1246463,1,Dial(SIP/8003,60,rT) exten= 1246463,n,Wait(5) exten= 1246463,n,Hangup exten= 6463,1,Dial(SIP/8003,60,rT) -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inbound filed
i call my tollfree number and send the call to my extension 8003 On Wed, Apr 15, 2009 at 10:51 AM, Brandon B. bran...@brellsystems.comwrote: You call call to extension '246463' will not match 'exten = 1246463'. On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: i create inbound confi my confi is: [incoming] exten= 1246463,,1,Dial(SIP/8003,60,rT) exten= 6463,1,Dial(SIP/8003,60,rT) exten= 1246463,,n,Wait(5) exten= 1246463,,n,Hangup but y calling and send this error in my CLI: [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inbound filed
nothing [Apr 15 11:24:15] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. On Wed, Apr 15, 2009 at 11:07 AM, Brandon B. bran...@brellsystems.comwrote: Try this: [incoming] exten= 246463,1,Dial(SIP/8003,60,rT) exten= 246463,n,Wait(5) exten= 246463,n,Hangup exten= 6463,1,Dial(SIP/8003,60,rT) On Wed, Apr 15, 2009 at 11:00 AM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: i call my tollfree number and send the call to my extension 8003 On Wed, Apr 15, 2009 at 10:51 AM, Brandon B. bran...@brellsystems.comwrote: You call call to extension '246463' will not match 'exten = 1246463'. On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: i create inbound confi my confi is: [incoming] exten= 1246463,,1,Dial(SIP/8003,60,rT) exten= 6463,1,Dial(SIP/8003,60,rT) exten= 1246463,,n,Wait(5) exten= 1246463,,n,Hangup but y calling and send this error in my CLI: [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] inbound filed
tollfree calls was working fine but stopped working without any reason On Wed, Apr 15, 2009 at 11:51 AM, Brandon B. bran...@brellsystems.comwrote: Is your system configured to send the dialed calls to the [incoming] context? On Wed, Apr 15, 2009 at 11:22 AM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: nothing [Apr 15 11:24:15] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. On Wed, Apr 15, 2009 at 11:07 AM, Brandon B. bran...@brellsystems.comwrote: Try this: [incoming] exten= 246463,1,Dial(SIP/8003,60,rT) exten= 246463,n,Wait(5) exten= 246463,n,Hangup exten= 6463,1,Dial(SIP/8003,60,rT) On Wed, Apr 15, 2009 at 11:00 AM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: i call my tollfree number and send the call to my extension 8003 On Wed, Apr 15, 2009 at 10:51 AM, Brandon B. bran...@brellsystems.comwrote: You call call to extension '246463' will not match 'exten = 1246463'. On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: i create inbound confi my confi is: [incoming] exten= 1246463,,1,Dial(SIP/8003,60,rT) exten= 6463,1,Dial(SIP/8003,60,rT) exten= 1246463,,n,Wait(5) exten= 1246463,,n,Hangup but y calling and send this error in my CLI: [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support - Linux E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action
[asterisk-users] Help Inbound number
i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Inbound number
in my extension.conf i set : [default] exten = 1246463,1,Answer(SIP/8003) On Mon, Mar 16, 2009 at 12:06 PM, Pascal Bruno tipas...@gmail.com wrote: Do you have an extension set for 246463 in your extensions.conf? On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Inbound number
nothing the problem persitem On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee gera...@gmail.com wrote: are you sure calls from this provider are going to context 'default' ? sip.conf [procall] type=peer username=XX secret=XX context=default 2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Inbound number
The inbound was working well suddenly stopped working I want all calls made to the number should answer the extension 8003 On Mon, Mar 16, 2009 at 3:49 PM, Danny Nicholas da...@debsinc.com wrote: Just to read this right – you are trying to take an inbound call from 888xxx and transfer it to your sip extension 8003? If so, Are you able to make internal calls to 8003? Can you transfer other calls to 8003 (exten = s,1,Dial(SIP/8003) ) ? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Monday, March 16, 2009 4:38 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Help Inbound number nothing the problem persitem On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee gera...@gmail.com wrote: are you sure calls from this provider are going to context 'default' ? sip.conf [procall] type=peer username=XX secret=XX context=default 2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is http://downloads.digium.com/pub/ down???
i any problem conecct ready On Sat, Jan 31, 2009 at 7:37 AM, Jonn Taylor jo...@taylortelephone.comwrote: Anyone else having problems connecting to http://downloads.digium.com/pub/ ?? Jonn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
24 chanels On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com wrote: What is your call-limit set to in sip.conf? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Tuesday, January 27, 2009 9:12 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] I need help The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I need help
only 3 On Tue, Jan 27, 2009 at 10:03 AM, Danny Nicholas da...@debsinc.com wrote: Is it the same 3 or the first 3? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Tuesday, January 27, 2009 9:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] I need help 24 chanels On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com wrote: What is your call-limit set to in sip.conf? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Tuesday, January 27, 2009 9:12 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] I need help The problem is I have 20 agents calling all but 3 of them get this error when calling On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote: Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' if you have made no changes to your asterisk configuration since it last worked Contact your service provider as they have an issue more likely though is you have a bad dial command. /telepathy -- Kind Regards Max Brooks - Developer Legatio Technologies Limited Phone: 01793 520 506 www.legatio.com, www.ftax.co.uk Legatio is part of the Callcredit Information Group: www.skipton.com, www.callcredit.co.uk, www.eurodirect.co.uk Legatio Technologies Limited, One Park Lane, Leeds, West Yorkshire, LS3 1EP Registered in England and Wales No. 4519902 - Powered by Legatio.com This message is confidential. It may not be disclosed to, or used by, anyone other than the addressee(s). If you receive this message in error, please advise us immediately using the email address i...@legatio.com. Internet e-mail is not necessarily secure. Legatio will not accept responsibility for alterations or additions to any e-mail message or attached documents that occur after transmission. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer
[asterisk-users] I need help
i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] soft phone
i am used eyebeam is a good softphone On Sun, Jan 25, 2009 at 12:55 PM, David fire ddf...@gmail.com wrote: hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need Help
i need a program for monitorin my bandwitch of my asterisk server -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] recording failed
I have a problem when I call a good record but I make a call to return to the same number I erased the previous record, and I replaced with the last call -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recordin call in asterisk
I need help need recording all call for my pbx but i am a novato in asterisk
my confi for record is:
exten=_NX,n,Set(CALLFILENAME=CLIENTE-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID})
exten = _NX,n,MixMonitor(${CALLFILENAME}.gsm,m)
exten = _NX,n,Dial(${TRUNK_CLIENTE}/${EXTEN})
--
Bayardo Sánchez García
Web Developer - Internet Portals
Linux User: #418392
Ubuntu User #14171
America Central - Managua, NI (505) 249-2853 - 4886876
IM msn messenger: bjsanch...@hotmail.com
Skype: bayardo.sanchez
This email is intended solely for the person or organization to which it is
addressed. It may contain privileged and confidential information. If you
are not the intended recipient, you are prohibited from copying, disclosing
or distributing this email or its contents (as it may be unlawful for you to
do so) or taking any action in reliance on it. If you have received this
email by mistake, please delete it. All e-mail sent to this address will be
received by B.S. Solution e-mail system and is subject to archiving and
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___
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