[asterisk-users] res_config_mysql.c: MySQL RealTime: Failed to connect database server ..
Hi, I don't explain very well what my problem, but i can't make calls. i analise my log full and i found two errors Jul 3 19:02:08 ERROR[4670] res_config_mysql.c: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. Jul 3 19:02:08 VERBOSE[4670] logger.c: -- Added extension 'exit-FAILED' priority 1 to macro-vm Jul 3 19:02:08 VERBOSE[4670] logger.c: -- Added extension 'exit-FAILED' priority 2 to macro-vm Jul 3 19:02:08 VERBOSE[4670] logger.c: -- Added extension 'exit-SUCCESS' priority 1 to macro-vm my complete log full, in anex. thanks -- Carlos Jerónimo ___ --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] No sound, problem is not a NAT
HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: Connected to Asterisk SVN-branch-1.2-r68526 currently running on hernandezz-laptop (pid = 6970) Verbosity is at least 10 -- Executing Macro(SIP/5000-081da408, user-callerid|) in new stack -- Executing NoOp(SIP/5000-081da408, user-callerid: device 5000) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?report) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?start) in new stack -- Executing Set(SIP/5000-081da408, REALCALLERIDNUM=5000) in new stack -- Executing NoOp(SIP/5000-081da408, REALCALLERIDNUM is 5000) in new stack -- Executing Set(SIP/5000-081da408, AMPUSER=5000) in new stack -- Executing Set(SIP/5000-081da408, AMPUSERCIDNAME=hnmvbn) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?report) in new stack -- Executing Set(SIP/5000-081da408, CALLERID(all)=hnmvbn 5000) in new stack -- Executing Set(SIP/5000-081da408, REALCALLERIDNUM=5000) in new stack -- Executing NoOp(SIP/5000-081da408, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?continue) in new stack -- Executing Set(SIP/5000-081da408, __TTL=64) in new stack -- Executing GotoIf(SIP/5000-081da408, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/5000-081da408, Using CallerID hnmvbn 5000) in new stack -- Executing Wait(SIP/5000-081da408, 2) in new stack -- Executing Macro(SIP/5000-081da408, systemrecording|dorecord) in new stack -- Executing Goto(SIP/5000-081da408, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/5000-081da408, /tmp/5000-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') hernandezz-laptop*CLI Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call problem...
please? -- Carlos Jerónimo ___ --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] call problem...
Hi, tahnks for your answer. but i haved install with command apt-get install asterisk, but i don't have package asterisk-addons. if i download asterisk-addons by digium site, run well with asterisk debian pakages?? thanks 2007/6/8, Tzafrir Cohen [EMAIL PROTECTED]: On Fri, Jun 08, 2007 at 02:52:39PM +0100, Carlos Jerónimo wrote: Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: [EMAIL PROTECTED]:/home/hernandezz# asterisk -rvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i noticed the problem could be the module format_mp3.so not being loaded because not exists in my PC. in modules.conf i comment the line load = format_mp3.so and now it's works.This module is necessary? It can be used for playing mp3 files. It is part of asterisk-addons. You may need to reinstall / upgrade asterisk-addons for the current version of Asterisk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreePbx/asterisk/openser
Hi, i use asterisk with freePbx for all configurations. Now i want use a openser with asterisk but also with freepbx. I pretend use the asterisk whit freepbx, but autentications for users in openSer.. it's possible?? thanks -- Carlos Jerónimo ___ --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] internal sounds of asterisk / freePBX
Hi chris. The result it is the same, no sound. -- Executing Answer(SIP/7010-081f6f68, ) in new stack -- Executing Playback(SIP/7010-081f6f68, beep) in new stack -- Playing 'beep' (language 'en') more sugestion? 2007/4/18, Christopher Aloi [EMAIL PROTECTED]: Try getting rid of all those macros etc.. so you can see what's going on, something simple like: exten = 500,1,Answer() exten = 500,n,Playback(beep) exten = 500,n,Hangup() Then dial 500 from your soft phone and see what happens. On 4/17/07, EWV2 [EMAIL PROTECTED] wrote: The codecs are correct, so you are having other type of problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --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 ___ --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 -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] internal sounds of asterisk / freePBX
no i don't have any card. 2007/4/18, Leonardo Kamache (Gmail) [EMAIL PROTECTED]: Did you have any E1/T1 cards in your server? On 4/18/07, shadowym [EMAIL PROTECTED] wrote: CallWeaver is the new name for OpenPBX -Original Message- From: Carlos Jerónimo [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] internal sounds of asterisk / freePBX i use xlite and kphone in a diferent pc's. i can phone well. the problem is internal asterisk sounds. I think i not use Call Weaver, what is call weaver, i search at google but i'm was confused. i hope more help's. thanks 2007/4/17, Andrew Joakimsen [EMAIL PROTECTED]: If that's what your phone is setup. Are you even using a SIP phone? What does the PEER context contain? Also, while Asterisk 1.2 and CALL WEAVER are basically the same (besides that fact that CALL WEAVER is trying to fully support faxing and Asterisk/Digium refuse to support correctly faxing) they do not share sound files. So if you are indeed using CALL WEAVER and their sounds you shouldn't be asking about that here. On 4/17/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update
[asterisk-users] internal sounds of asterisk / freePBX
Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --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] internal sounds of asterisk / freePBX
HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --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 ___ --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 -- Carlos Jerónimo ___ --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] internal sounds of asterisk / freePBX
i use xlite and kphone in a diferent pc's. i can phone well. the problem is internal asterisk sounds. I think i not use Call Weaver, what is call weaver, i search at google but i'm was confused. i hope more help's. thanks 2007/4/17, Andrew Joakimsen [EMAIL PROTECTED]: If that's what your phone is setup. Are you even using a SIP phone? What does the PEER context contain? Also, while Asterisk 1.2 and CALL WEAVER are basically the same (besides that fact that CALL WEAVER is trying to fully support faxing and Asterisk/Digium refuse to support correctly faxing) they do not share sound files. So if you are indeed using CALL WEAVER and their sounds you shouldn't be asking about that here. On 4/17/07, Carlos Jerónimo [EMAIL PROTECTED] wrote: HI, my sip.conf /codecs disallow=all allow=ulaw allow=alaw this codcs is correct? thanks 2007/4/17, EWV2 [EMAIL PROTECTED]: It sounds like a codec problem. What codec are you using? If you are using g723.1 or g729 passthru you will not be able to hear nothing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Jerónimo Sent: Tuesday, April 17, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] internal sounds of asterisk / freePBX Sorry but i can't register in the freepbx forum, so this is my solutons for resolve my trouble. HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: == Spawn extension (macro-systemrecording, h, 1) exited non-zero on 'SIP/7010-081d7288' -- Executing Macro(SIP/7010-0819b350, user-callerid|) in new stack -- Executing NoOp(SIP/7010-0819b350, user-callerid: device 7010) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?start) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, REALCALLERIDNUM is 7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSER=7010) in new stack -- Executing Set(SIP/7010-0819b350, AMPUSERCIDNAME=Portaria) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?report) in new stack -- Executing Set(SIP/7010-0819b350, CALLERID(all)=Portaria 7010) in new stack -- Executing Set(SIP/7010-0819b350, REALCALLERIDNUM=7010) in new stack -- Executing NoOp(SIP/7010-0819b350, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/7010-0819b350, 0?continue) in new stack -- Executing Set(SIP/7010-0819b350, _TTL=64) in new stack -- Executing GotoIf(SIP/7010-0819b350, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/7010-0819b350, Using CallerID Portaria 7010) in new stack -- Executing Wait(SIP/7010-0819b350, 2) in new stack -- Executing Macro(SIP/7010-0819b350, systemrecording|dorecord) in new stack -- Executing Goto(SIP/7010-0819b350, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/7010-0819b350, /tmp/7010-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --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 ___ --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 -- Carlos Jerónimo ___ --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 ___ --Bandwidth and Colocation provided
[asterisk-users] Freeworddialup, no inbound calls
Hi, i registered freeworlddialup, and follow steps by the ebook trixbox made easy. i can make outbound calls, in my network and out (i can call a gizmo number), and echo test, but i can not receive incomming calls, out of network extensions, for example in the page freeworlddialup, i go to the link Call me in my.FWD and nothing. . i read any posts here, but any solution for my problem. when i run command: sip show registry in asterisk not apear anything registred. What the problem? is the problem? in my extensions nat=yes. my configurations: allow=ulaw auth=md5 callerid=Carlos Jeronimo‹835557› disallow=all host=iax2.fwdnet.net qualify=yes secret=xxx type=peer username=835557 Incoming Settings: allow=ulaw auth=rsa context=from-pstn disallow=all inkeys=freeworlddialup secret=xxx type=user 835557:[EMAIL PROTECTED] any sugestion? regards -- Carlos Jerónimo ___ --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] Couldn't load variables.txt?aldope=xxxxx
It's works. a lot of regards. thanks for all. 2007/3/30, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, I suggest you to use /init/op_panel_debian.sh script inside oppanel tar file. Put it inside /etc/init.d and then as root type: *update-rc.d op_panel defaults * to setup the script for boot. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgi thanks for all, it works. Your opinion is correct, my op_server was not run, and i run him. I'm using Ubuntu Dapper and i want run the op_server when the machine starts, and i add a line in the file rc.local like this: cd /var/www/html/panel/; su -c /var/www/html/panel/op_server.pl but this not works when i start the machine. Please say me the changes i will have put in this line. thanks 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, if you have not op_panel.pid in /var/run/asterisk this means the panel server is not working. I do not know where freepbx puts oppanel files (usually they are in /usr/local but not always). Just find them and exec the file *op_server.pl* in stand alone mode (just type ./op_server.pl inside its directory) so you can see it is working (you'll see a lot of messages). If you cannot find the oppanel dir this means it is not installed. You could download it from www.asternic.org and install it following the instructions on the site. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio. when i type: ps -A -F | grep panel: # [EMAIL PROTECTED]:/home/hernandezz# ps -A -F | grep panel 1000 5378 1 0 8596 15480 0 11:26 ?00:00:02 gnome-panel --sm-client-id default1 1000 5433 1 0 5903 10900 0 11:26 ?00:00:00 /usr/lib/gnome-panel/clock-applet --oaf-activate-iid=OAFIID:GNOME_ClockApplet_Factory --oaf-ior-fd=37 root 7363 6544 0 723 812 0 12:19 pts/000:00:00 grep panel # when i type: [EMAIL PROTECTED]:/home/hernandezz# /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p # Can't open perl script /usr/local/op_panel/op_server.pl: No such file or directory [EMAIL PROTECTED]:/home/hernandezz# # what's the problem..? sorry i try search but i'm not a linux expert. thanks. 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, type: *ps -A -F | grep panel* You should see something like: root 14851 1 0 2700 8164 0 11:01 ?00:00:01 /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p /var/run/asterisk/op_panel.pid This means that tha panel process is running. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio, sorry but how do this? how i verify the server it's running, and if not runnig how i put this running. Thanks 2007/3/28, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? ___ --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 ___ --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 ___ --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 ___ --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 -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error in FreePBX
Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg Im-sorryan-error-has-occured and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some NAT configuration parameters, it didnt worked aswell. As this ever happened to anyone before? Any hints are very appreciated. Thank you very much -- Carlos Jerónimo ___ --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] Couldn't load variables.txt?aldope=xxxxx
Hi Giorgio. when i type: ps -A -F | grep panel: # [EMAIL PROTECTED]:/home/hernandezz# ps -A -F | grep panel 1000 5378 1 0 8596 15480 0 11:26 ?00:00:02 gnome-panel --sm-client-id default1 1000 5433 1 0 5903 10900 0 11:26 ?00:00:00 /usr/lib/gnome-panel/clock-applet --oaf-activate-iid=OAFIID:GNOME_ClockApplet_Factory --oaf-ior-fd=37 root 7363 6544 0 723 812 0 12:19 pts/000:00:00 grep panel # when i type: [EMAIL PROTECTED]:/home/hernandezz# /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p # Can't open perl script /usr/local/op_panel/op_server.pl: No such file or directory [EMAIL PROTECTED]:/home/hernandezz# # what's the problem..? sorry i try search but i'm not a linux expert. thanks. 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, type: *ps -A -F | grep panel* You should see something like: root 14851 1 0 2700 8164 0 11:01 ?00:00:01 /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p /var/run/asterisk/op_panel.pid This means that tha panel process is running. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio, sorry but how do this? how i verify the server it's running, and if not runnig how i put this running. Thanks 2007/3/28, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? ___ --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 ___ --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 -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error in FreePBX
-08197f70, exit-FAILED|1) in new stack -- Goto (macro-vm,exit-FAILED,1) -- Executing Playback(SIP/6009-08197f70, im-sorryan-error-has-occured) in new stack -- Playing 'im-sorry' (language 'en') -- Playing 'an-error-has-occured' (language 'en') -- Executing Hangup(SIP/6009-08197f70, ) in new stack == Spawn extension (macro-vm, exit-FAILED, 2) exited non-zero on 'SIP/6009-08197f70' in macro 'vm' == Spawn extension (macro-vm, exit-FAILED, 2) exited non-zero on 'SIP/6009-08197f70' in macro 'exten-vm' == Spawn extension (macro-vm, exit-FAILED, 2) exited non-zero on 'SIP/6009-08197f70' ieeta-proj-04*CLI thanks Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error in FreePBX
Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx foruns this week, and my login is inactive yet. In the mail i receive this msg: Welcome to FreePBX Forums Forums Please keep this email for your records. Your account information is as follows: Your account is currently inactive, the administrator of the board will need to activate it before you can log in. You will receive another email when this has occured. ** because this i post this here. Regards 2007/3/29, Steve Murphy [EMAIL PROTECTED]: On Thu, 2007-03-29 at 13:26 +0200, Remco Barendse wrote: On Thu, 29 Mar 2007, Carlos Jerónimo wrote: Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg Im-sorryan-error-has-occured and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some NAT configuration parameters, it didnt worked aswell. As this ever happened to anyone before? Any hints are very appreciated. Thank you very much I have the same problem, it seems to occur when an extension is busy here. All my extensions are on local lan with phones having ip addresses in a private range without NAT or anything so that is not the problem. Sounds like an error in the dial pan FreePBX generated. My suggestion: try a FreePBX mailing list first; the problem *is* more likely to be in their stuff. murf -- Steve Murphy Software Developer Digium ___ --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 -- Carlos Jerónimo ___ --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] Couldn't load variables.txt?aldope=xxxxx
Hi Giorgi thanks for all, it works. Your opinion is correct, my op_server was not run, and i run him. I'm using Ubuntu Dapper and i want run the op_server when the machine starts, and i add a line in the file rc.local like this: cd /var/www/html/panel/; su -c /var/www/html/panel/op_server.pl but this not works when i start the machine. Please say me the changes i will have put in this line. thanks 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, if you have not op_panel.pid in /var/run/asterisk this means the panel server is not working. I do not know where freepbx puts oppanel files (usually they are in /usr/local but not always). Just find them and exec the file *op_server.pl* in stand alone mode (just type ./op_server.pl inside its directory) so you can see it is working (you'll see a lot of messages). If you cannot find the oppanel dir this means it is not installed. You could download it from www.asternic.org and install it following the instructions on the site. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio. when i type: ps -A -F | grep panel: # [EMAIL PROTECTED]:/home/hernandezz# ps -A -F | grep panel 1000 5378 1 0 8596 15480 0 11:26 ?00:00:02 gnome-panel --sm-client-id default1 1000 5433 1 0 5903 10900 0 11:26 ?00:00:00 /usr/lib/gnome-panel/clock-applet --oaf-activate-iid=OAFIID:GNOME_ClockApplet_Factory --oaf-ior-fd=37 root 7363 6544 0 723 812 0 12:19 pts/000:00:00 grep panel # when i type: [EMAIL PROTECTED]:/home/hernandezz# /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p # Can't open perl script /usr/local/op_panel/op_server.pl: No such file or directory [EMAIL PROTECTED]:/home/hernandezz# # what's the problem..? sorry i try search but i'm not a linux expert. thanks. 2007/3/29, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, type: *ps -A -F | grep panel* You should see something like: root 14851 1 0 2700 8164 0 11:01 ?00:00:01 /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p /var/run/asterisk/op_panel.pid This means that tha panel process is running. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgio, sorry but how do this? how i verify the server it's running, and if not runnig how i put this running. Thanks 2007/3/28, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? ___ --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 ___ --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 ___ --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 -- Carlos Jerónimo ___ --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] Couldn't load variables.txt?aldope=xxxxx
Hi Giorgio, sorry but how do this? how i verify the server it's running, and if not runnig how i put this running. Thanks 2007/3/28, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? ___ --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 -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Couldn't load variables.txt?aldope=xxxxx
HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? -- Carlos Jerónimo ___ --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] ses ActiveDirectory and also Ldap and Kerberos.
Ok thanks.. in the future i post hear the results. 2007/3/22, dave cantera [EMAIL PROTECTED]: carlos, this is coming from a linux admin perspective but here is something to get started... active directory transfers info to-from windows domain controllers via the network. there are probably api frameworks available as open source, although they may be incomplete. I have seen another vendor create a product that when you logged into the windows box, the startup script had a program that also logged into a linux box... you would have to write that binary yourself... once you did it on windows, you would have to port it to linux. I would look to the samba project as a start... they do windows file sharing and probably login to widows controllers to get authentication.. haven't used it extensively though in years... I know that it is doable, would take a lot of research if you never used samba... let me google something and see if I can get any more info... ok, you might start here... http://www.google.com/search?hl=enq=%22active+directory%22+%22single+signon%22+sambabtnG=Google+Search good luck, I would be interested to see what you come up with... please repost the results here! daveC Carlos Jerónimo wrote: Hi i'm student and my final project is related to Voip. I have Asterisk almost fully configured. The next step is to accept login of users, that data is in Universitys database which uses ActiveDirectory and also Ldap and Kerberos. It's possible? I don't want authentications in sip.conf, but in other remote database. The problem is i don't have ideas how to start with. I would appreciate some ideas be -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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 -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ses ActiveDirectory and also Ldap and Kerberos.
Hi i'm student and my final project is related to Voip. I have Asterisk almost fully configured. The next step is to accept login of users, that data is in Universitys database which uses ActiveDirectory and also Ldap and Kerberos. It's possible? I don't want authentications in sip.conf, but in other remote database. The problem is i don't have ideas how to start with. I would appreciate some ideas be -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error, install freePbx
Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package pear.php.net/db Cannot initialize 'db' , invalid or missing package files Package db is not valid install failed Why this error? help me, please. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error, install freePbx
thanks, and sorry because the mailing list nothing to do with pear. but it'sa for install freePBX. pear install DB not works. more any sugestion? 2007/3/20, dima [EMAIL PROTECTED]: perhaps you should try pear install DB However note that this mailing list has nothing to do with pear. Hi, i try install FreePbx by tuturial in http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Pasoview_comment_id=13443 but i have this error when i try install freepbx: #pear install db No releases available for package pear.php.net/db Cannot initialize 'db' , invalid or missing package files Package db is not valid install failed Why this error? help me, please. ___ --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 -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Autentication
Hi, i have a doubt about autentication in asterisk. it's possible to integration the asterisk with the other server for autentication, for example kerberos, ou other? i want to implement asterisk in a department of university, but it's necessary autentication by students, login and password for example. thanks. -- Carlos Jerónimo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice mail server
Hi, how i have to do for receive a email with a alert from my voice mail? My doubt is what I put in serveremail in file voicemail.conf. I think is a email server, but can be see anyone? I searching one in the internet? Thanks and sory my english ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MusiconHold
Hi, i configured Musiconhold and Works, but the sound is very low. I haved put the volume in the max, but is equal. I tested to my voice, and the sound is also low. exten=8000,1,Wait(2) exten=8000,2,Record(menu:gsm) exten=8000,3,Wait(2) exten=8000,4,Playback(menu) exten=8000,5,Hangup() when the musicaonhold is play e recieved this warning. exten = 6000,1,MusicOnHold() Executing MusicOnHold(SIP/2000-f7d9, pessoal) in new stack -- Started music on hold, class 'pessoal', on SIP/2000-f7d9 Feb 16 15:45:14 WARNING[8318]: interface.c:215 decodeMP3: Junk at the beginning of frame Please I need a suggestion, I NOT HAVE FXO, only two network card Thanks and sory my english ** My configuration: Extensions.conf exten = 6000,1,MusicOnHold() Zapata.conf musiconhold=default context=default musiconhold.conf [default] directory=/var/lib/asterisk/mohmp3/pessoal/ mode=files random= yes ___ --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] Voice mail server
Thanks Philipp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: sexta-feira, 23 de Fevereiro de 2007 22:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice mail server Carlos Jerónimo wrote: Hi, how i have to do for receive a email with a alert from my voice mail? You need a working installation of sendmail on your server. Then you append the email address of the users to the mailbox definitions in voicemail.conf like this: 1234 = 1234,Some User,[EMAIL PROTECTED] (See http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) My doubt is what I put in serveremail in file voicemail.conf. I think is a email server, but can be see anyone? I searching one in the internet? No, that's not a mail server. Just use the default serveremail=asterisk Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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 ___ --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] Voice mail server
Thanks Philipp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: sexta-feira, 23 de Fevereiro de 2007 22:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice mail server Carlos Jerónimo wrote: Hi, how i have to do for receive a email with a alert from my voice mail? You need a working installation of sendmail on your server. Then you append the email address of the users to the mailbox definitions in voicemail.conf like this: 1234 = 1234,Some User,[EMAIL PROTECTED] (See http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf) My doubt is what I put in serveremail in file voicemail.conf. I think is a email server, but can be see anyone? I searching one in the internet? No, that's not a mail server. Just use the default serveremail=asterisk Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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 ___ --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
