Re: [Asterisk-Users] flash button on asterisk + legacy pbx system
Hi Michael, I have a TDM400P on an Asterisk box with: 1) a FXO connected to the old pbx and 2) a FXS connected to a normal analog phone 3) the analog phone is a Telecom Sirio, (the most common in Italy) If I knew how to check asterisk send/receive this non-digits signals it can be easier to understand if everything is going wrong, even for the future...I do not think installing Asterisk keeping an old PBX is so uncommon. TIA Giorgio Incantalupo. Michael Collins wrote: you say Flash asterisk command send a flash signal to old pbx so that it sees that command as coming from an analog phone. But since Flash is not a digit, how can I catch it from within asterisk? How can I tell asterisk (es inside extensions.conf) to do something whene receive it from a phone? Giorgio, Could you please give us some more information about your setup? Two really important questions: #1 - how are you connected from Asterisk to the PBX? (FXO/FXS, T1, etc.?) #2 - how are you connected from Asterisk to the telephone? (SIP, FXS, etc.?) #3 - What kind of telephone are you using? Knowing this will help us figure out what is going on. -MC ___ --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
[Asterisk-Users] Voicemail options
Currently we have (with our NEC phone system) the options in voicemail to have a message say press 2 to go to my mobile phone Can this be done in asterisk without setting up an IVR for each user ? Has anyone got a voicemail dialplan that can do this ? Thanks -- Kevin Withnall ILB Computing PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081 http://kevin.withnall.com/ smime.p7s Description: S/MIME cryptographic signature ___ --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] Voicemail options
Asterisk has an option to have an out (by pressing '0') and you could use that to jump out of voicemail and off to someones mobile. Maybe a dbget to grab the mobile phone for the user would be a neat way to go. -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au ph: 03 8320 8100 On Tue, 2006-07-04 at 16:33 +1000, Kevin Withnall wrote: Currently we have (with our NEC phone system) the options in voicemail to have a message say press 2 to go to my mobile phone Can this be done in asterisk without setting up an IVR for each user ? Has anyone got a voicemail dialplan that can do this ? Thanks -- Kevin Withnall ILB Computing PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081 http://kevin.withnall.com/ ___ --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
[Asterisk-Users] Running 40 active calls (too m uch för CPU?)
Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio tracks on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --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] Nokia E61
Devraj Mukherjee wrote: Hello world, Any success stories of getting a Nokia E61 to work with Asterisk server? Interested to hear before we buy them for work :) I don't know about e61, but I connected an e60 up yesterday that wasn't any hassle. Even the stories about poor quality with WPA + G.729 seemed to be false. ___ --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] Voicemail options
Thanks for that, it works like a charm :-) -- Kevin Withnall ILB Computing PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081 http://kevin.withnall.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Tuesday, 4 July 2006 4:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail options Asterisk has an option to have an out (by pressing '0') and you could use that to jump out of voicemail and off to someones mobile. Maybe a dbget to grab the mobile phone for the user would be a neat way to go. -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au ph: 03 8320 8100 On Tue, 2006-07-04 at 16:33 +1000, Kevin Withnall wrote: Currently we have (with our NEC phone system) the options in voicemail to have a message say press 2 to go to my mobile phone Can this be done in asterisk without setting up an IVR for each user ? Has anyone got a voicemail dialplan that can do this ? Thanks -- Kevin Withnall ILB Computing PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081 http://kevin.withnall.com/ ___ --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
[Asterisk-Users] PSTN Incoming Route
Greetings, I have installed a new FXO card but even though there's no incoming route, it answers the line after 2 to 3 rings. If I do create an incoming route, the same happens, but it never rings the ring group or extension I enter. It's almost as if the card acts as a modem. The caller hears nothing, just silence. I have a VoIP incoming route which works perfect. Any comments will be greatly appreciated... Many thanks, P. ___ --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
SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)
I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Running 40 active calls (too much för CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio tracks on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --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
[Asterisk-Users] Qsig-Link * to Meridian 81c
Hi all, I have a Qsig link over a TE210P card between my asterisk box and a meridian 81c which worked very well. My problem is that no name is transmitted in both directions. I always get messages like. !! Unknown IE 49 (cs5, Unknown Information Element) !! Unknown IE 50 (cs5, Unknown Information Element) My iax/sip clients contains the callerid like callerid=name nr My Zapata.conf section is like ;span 2 TE210P Card 0 switchtype=qsig signalling=pri_cpe pridialplan=private prilocaldialplan=private nfs=megacom usecallerid=yes overlapdial=yes usecallingpres=yes priindication=outofband facilityenable = yes callerid=asreceived context=MainMenu group=2 resetinterval = 1 channel=32-46,48-62 any ideas ? thanks in advance rgds marcus ___ --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] Nokia E61
Hi, configuration for E61 is the same as E60. As for the codec, G729 works between E60/61 phones (G729 passthru). At 03:44 PM 7/4/2006, you wrote: Devraj Mukherjee wrote: Hello world, Any success stories of getting a Nokia E61 to work with Asterisk server? Interested to hear before we buy them for work :) I don't know about e61, but I connected an e60 up yesterday that wasn't any hassle. Even the stories about poor quality with WPA + G.729 seemed to be false. ___ --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] PSTN Incoming Route
Check the default context defined in zapata.conf which is where incoming calls will go to. It may be going to a context that you are not aware of. - Daniel On Jul 4, 2006, at 3:46 AM, Pierre du Plessis wrote: Greetings, I have installed a new FXO card but even though there's no incoming route, it answers the line after 2 to 3 rings. If I do create an incoming route, the same happens, but it never rings the ring group or extension I enter. It's almost as if the card acts as a modem. The caller hears nothing, just silence. I have a VoIP incoming route which works perfect. Any comments will be greatly appreciated... Many thanks, P. ___ --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] How to configure NOKIA N70 with Asterisk?
Asterisk has some issues with the Nokia SIP client. I've started to add some small changes to svn trunk to support call hold with the Nokia, as well as behave a bit better in regards to ilbc encoding, even though that should still be avoided. I've had a lot of issues with the Nokia loosing the registration and WLAN access while I'm still in the office. Anyone that have any remedies for that? /Olle --- * Asterisk Training http://edvina.net/training/ * Next: Asterisk SIP Masterclass, Chicago | Asterisk Bootcamp on the Beach, Spain ___ --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] SIP debug logging
i am trying to sort out an issue with my SIP provider (I can make outgoing calls but am not recieving calls) and have been trying to use sip debug from the CLI. I am after a way to get these debug messages into a file (I find it easier to go over a file than having to deal with all the re-register messages flying by the screen). I have tried setting the debug field in logger.conf, but this doesn't seem to be doing what I want. Is there an easy way to have these messages going into a file? I find that the easiest way is to connect to asterisk like this: asterisk -rn | tee /tmp/asterisk-debug The n will remove some ascii color codes. You might want to add a g as well to force core files if Asterisk crashes. /o --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] Putting a call recording into a mailbox
Hi, I was wondering, after recording a call (through either the monitor()-application or automon), is there a way to put the recorded file into a user's mailbox? So far, we just send out the file as an email attachment, but having it in my mailbox would just be so much more convenient... (^_^) Marc Rohlfing ___ --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] Putting a call recording into a mailbox
On Tue, 2006-07-04 at 10:17 +0200, Marc Rohlfing wrote: Hi, I was wondering, after recording a call (through either the monitor()-application or automon), is there a way to put the recorded file into a user's mailbox? So far, we just send out the file as an email attachment, but having it in my mailbox would just be so much more convenient... (^_^) I don't understand what you are asking: what's the difference between sending out the email as an attachment so it ends up in a user's mailbox versus having it in the user's mailbox. Aren't they the same? Can you share the script that automatically mails the recording to the user? Regards, Patrick ___ --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] Putting a call recording into a mailbox
They may chose to access the file via their phone. By delivering it to their mailbox they have the option of either phone access or email access (assuming voicemail is delivered to their email server). Just a guess. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Tuesday, 4 July 2006 4:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Putting a call recording into a mailbox On Tue, 2006-07-04 at 10:17 +0200, Marc Rohlfing wrote: Hi, I was wondering, after recording a call (through either the monitor()-application or automon), is there a way to put the recorded file into a user's mailbox? So far, we just send out the file as an email attachment, but having it in my mailbox would just be so much more convenient... (^_^) I don't understand what you are asking: what's the difference between sending out the email as an attachment so it ends up in a user's mailbox versus having it in the user's mailbox. Aren't they the same? Can you share the script that automatically mails the recording to the user? Regards, Patrick ___ --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
AW: [Asterisk-Users] Putting a call recording into a mailbox
Hi, I don't understand what you are asking: what's the difference between sending out the email as an attachment so it ends up in a user's mailbox versus having it in the user's mailbox. Aren't they the same? Oops, my bad: I'm talking about the user's *voicemail* box here - should have been more precise there... Marc Rohlfing ___ --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] trixbox 1.1 download
I have trixbox 1.0 how I can update it to 1.1 or from where I can download trixbox 1.1 Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] trixbox 1.1 download
On 8/3/06, Khaled Chehab [EMAIL PROTECTED] wrote: I have trixbox 1.0 how I can update it to 1.1 or from where I can download trixbox 1.1 Have you tried the trixbox-update.sh script? Mike Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Putting a call recording into a mailbox
This is a good idea, good use of technology. You should be able to do this by looking at the way voicemails are already being stored, just add the file in and make the txt file with the relevant information. Look in for hints: /var/spool/asterisk/voicemail/context/mailbox/ On 7/4/06, Marc Rohlfing [EMAIL PROTECTED] wrote: Hi, I don't understand what you are asking: what's the difference between sending out the email as an attachment so it ends up in a user's mailbox versus having it in the user's mailbox. Aren't they the same? Oops, my bad: I'm talking about the user's *voicemail* box here - should have been more precise there... Marc Rohlfing ___ --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] SIP debug logging
On Tue, Jul 04, 2006 at 10:00:30AM +0200, Olle E Johansson wrote: i am trying to sort out an issue with my SIP provider (I can make outgoing calls but am not recieving calls) and have been trying to use sip debug from the CLI. I am after a way to get these debug messages into a file (I find it easier to go over a file than having to deal with all the re-register messages flying by the screen). I have tried setting the debug field in logger.conf, but this doesn't seem to be doing what I want. Is there an easy way to have these messages going into a file? I find that the easiest way is to connect to asterisk like this: asterisk -rn | tee /tmp/asterisk-debug What's wrong with: tail -f /var/log/asterisk/full (after setting the desired verbosity and debug level) This also removes the need for -n -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] trixbox 1.1 download
Khaled Chehab wrote: //I have trixbox 1.0 how I can update it to 1.1 or from where I can download trixbox 1.1 // Do you want to get an answer? Can you read at all? Read the replies to your previous questions. 1. It is July not August. So fix your date. 2. Remove your disclaimer. It doesn't make any sense to add a disclaimer when you write to a public mailing list. * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * Kai ___ --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
SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)
Hello again, I read this interesting article about the TE405P card. How do I check what firmware version my card has? http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html ... And how do I update it if it's an old one? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Running 40 active calls (too much för CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio tracks on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --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] can't dial Scotland ...
Mark, While this is a possibility, what I'm really looking for is some help in where to start debugging this problem.CheersOn 7/3/06, Mark Phillips
[EMAIL PROTECTED] wrote:Perhaps the BT crew are all on a drunken rampage along Sochiehall
Street?On Mon, 2006-07-03 at 15:14 +0100, Colin MacMillan wrote: Hello, For some reason I can't call Scotland from London ... The details: Asterisk v.
1.2.9.1 ISDN2 Interface - Junghanns card with BRIstuff 0.3.0-PRE-1q Extensions.conf (context SIP-PHONES) exten=_X.,1,Dial(Zap/g1/${EXTEN},60) When I call this number - 01417778979 (this is a building company and
the number should work fine) - a woman's voice from BT announces - 'call cannot be completed as dialed, please check the number and try again'. I have only had this problem with calls to Glasgow, no other telephone
number is having a problem, local, national, or international. Any help appreciated Colin ___ --Bandwidth and Colocation provided by
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Re: [Asterisk-Users] Need help with Junghanns Quadbri
I think you'll need to set the jumpers on the card in order to specify the NT ports. Jean-Louis curty wrote: Hi everybody I hope that somebody can help me with the following I have 2 quadbri cards 2 - 1t0 cards 1 pabx alcatel 4200 I would like to connect my asterisk to the alcatel , I installed bristuff 0.3.0-1p , loaded the zaphfc driver in NT mode configured zaptel and zapata , it works great. then I removed the 1 t0 card, added the quadbri loaded qozap : insmod qozap.ko ports=15 ( 4 ports in NT ) adjusted the zaptel zapata, specified the right signalling, right context ran ztcfg -vv ( 12 channels configured ) started asterisk, I get layer1 down message on the 4 ports, leds remain red what ever I do in my conf , I am not able to get a reaction from the card ( I tried with my two quadbri, on 2 different pc's ) what can I check ? thanks jl ___ --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] How to configure NOKIA N70 with Asterisk?
On 4 Jul 2006, at 09:58, Olle E Johansson wrote: I've had a lot of issues with the Nokia loosing the registration and WLAN access while I'm still in the office. Anyone that have any remedies for that? Yep, that's my main issue as well. I doubts it's a configuration issue since there isn't all that much to configure. Maybe a software upgrade on the phone will help - apparently there has been one small upgrade since the version on my phone (1.0610.02.15), although for a different issue. jens ___ --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] Avaya 4610sw SIP setup problem
Hi, You can call me by my first name (Silviu) :)) I have made the changes to the settings file, I have removed the LDAP-related settings - nothing changes... The file is still taken into account, as other changes affect the phone, but the SIP fields stay desperately blank... I don't think I'll wait for the next firmware release, I'm currently evaluating several Siemens optiPoint phones (SIP) which look good so far. I have to get things moving, the customer won't wait forever for the Avaya phones to work.c However I'm a bit disappointed to leave things as they are, I have a feeling of ... failure? I guess I'll still try some thing or another in my (inexistent) spare time. Thanks for your help, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 04 July 2006 03:57To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Herchi,I want you to re-read my last e-mail very carefully. Your response does not mention at all my guess that the three SP_DIRSRVR variables may be giving you trouble. I'm still interested in knowing what happens if you remove them from your settings file. Also, I have heard a rumour that there will be a new firmware release on July 10th. Actually, I just clicked the feedback button on their web page for the firmware download and asked. They responded on the first business day (unusual for Avaya), indicating 7/10 is the approximate release date. So there you have my source Let me know On 7/3/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi, I had edited out all lines starting with a #, which is ot right, as the marker for comments is##... See below for the entire file. I just tried the configuration throughDHCP, by setting the 176 option to point to the right TFTP server and also to the right SIP proxy. The Avaya boot test application is not complaining, but the phones ... do I need to say it? *sigh* SET DOMAIN "company.com" SET DNSSRVR "204.140.111.43"SET PHNCC "352"SET PHNDPLENGTH "4"SET PHNIC "00"SET PHNOL "0"SET SYSLANG "English"SET APPSTAT "1"SET RESTORESTAT "1"SET AGCHAND "0"SET AGCHEAD "0"SET AGCSPKR "0"SET SNTPSRVR "204.140.111.200"SET DSTOFFSET "1"SET DSTSTART "1SunApr2L"SET DSTSTOP "LSunOct2L"SET GMTOFFSET "-5:00"SET DATESEPARATOR "/"SET DATETIMEFORMAT "3" SET SIPDOMAIN "slt05.company.agn" SET SIPPROXYSRVR "204.140.111.219"SET SIPPORT "5070" SET SIPREGISTRAR "204.140.111.219" SET DIALPLAN "[234]xxx|55"SET DIALWAIT "3"SET MUSICSRVR ""SET MWISRVR ""SET PHNNUMOFSA "3"SET REGISTERWAIT "120" SET SP_DIRSRVR "10.1.1.1"SET SP_DIRSRVRPORT "389"SET SP_DIRTOPDN "ou=People,o=avaya.com"IF $MODEL4 SEQ 4602 goto SETTINGS4602IF $MODEL4 SEQ 4610 goto SETTINGS4610IF $MODEL4 SEQ 4620 goto SETTINGS4620IF $MODEL4 SEQ 4621 goto SETTINGS4621IF $MODEL4 SEQ 4622 goto SETTINGS4622IF $MODEL4 SEQ 4625 goto SETTINGS4625IF $MODEL4 SEQ 4630 goto SETTINGS4630goto END # SETTINGS4602goto END# SETTINGS4610 SET WMLHOME " http://support.avaya.com/elmodocs2/avayaip/4620/home.wml" SET WMLPROXY "204.140.111.246" SET WMLPORT "3128"goto END # SETTINGS4620goto END# SETTINGS4621goto END# SETTINGS4622goto END# SETTINGS4625goto END# SETTINGS4630 SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htmSET PHNEMERGNUM 112goto END # END From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom Lynn Sent: 01 July 2006 18:18 To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Is the text shown below the ENTIRE file? It looks like all of the settings for the individial phone models are missing. I'm not sure what the consequences of branching to the 4610 section will be if it doesn't exist. Also, I don't use the SP_DIRSRVR values. What happens if those three entries are removed? SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com I can't find these three entries anywhere in my 46xx settings file.I also cannot find them in the lan admin guide from the manufacturer.They seem to be somewhat like the ldap options for the 4630 phone, but those didn't have a leading SP_ prefix on the variable name. Why don't you comment them out and see what happens?Tom Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET
RE: [Asterisk-Users] can't dial Scotland ...
Do you
know for sure that it's BT's announcement, and not one from your dial
plan?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Colin
MacMillanSent: 04 July 2006 10:21To: [EMAIL PROTECTED];
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] can't dial Scotland ...Mark, While this
is a possibility, what I'm really looking for is some help in where to start
debugging this problem.Cheers
On 7/3/06, Mark
Phillips [EMAIL PROTECTED]
wrote:
Perhaps
the BT crew are all on a drunken rampage along Sochiehall
Street?On Mon, 2006-07-03 at 15:14 +0100, Colin
MacMillan wrote: Hello, For some reason I can't call
Scotland from London ... The details: Asterisk v. 1.2.9.1 ISDN2 Interface - Junghanns card
with BRIstuff 0.3.0-PRE-1q Extensions.conf (context
SIP-PHONES) exten=_X.,1,Dial(Zap/g1/${EXTEN},60)
When I call this number - 01417778979 (this is a building company and
the number should work fine) - a woman's voice from BT announces
- 'call cannot be completed as dialed, please check the number and
try again'. I have only had this problem with calls
to Glasgow, no other telephone number is having a problem, local,
national, or international. Any help appreciated
Colin ___
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Re: [Asterisk-Users] Need help with Junghanns Quadbri
On 5/31/06, Jean-Louis curty [EMAIL PROTECTED] wrote: I does nothing special, no output, nor error, same.. .:-( you should at least get any output from ztcfg, but aside from that, like Tommaso said, you must also set the correct jumpers. cheers ___ --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] Integrate asterisk with Database
Vidura,you would want to use some kind of IVR + php-agi to do the database operations (of course there are 10 other combinations - like Ruby - on -rails and RAGI). Quick suggestion: if you've played with asterisk before, I recommend that you look at voip-info.org for php-agi links and snapvine.com if you want to use Ruby/RAGIif you would like professional help, i suggest you post it on asterisk-biz list or contact me off-list rajeevOn 7/3/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Marcin Lukasik wrote: Have you even _tried_ to create your dialplan?And to make it worse, he copied this drivel to the Developers lists.--Chris Mason(264) 497-5670 Fax: (264) 497-8463 Int:(305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271Cell: 264-235-5670Yahoo IM: [EMAIL PROTECTED]--This message has been scanned for viruses and dangerous content by MailScanner, and isbelieved to be clean.--This message has been scanned for viruses anddangerous content by MailScanner, and isbelieved to be clean.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with Junghanns Quadbri
thanks to all of you, I fixed my problem by changing the cable !jl 2006/7/4, stoffell [EMAIL PROTECTED]: On 5/31/06, Jean-Louis curty [EMAIL PROTECTED] wrote: I does nothing special, no output, nor error, same.. .:-(you should at least get any output from ztcfg, but aside from that,like Tommaso said, you must also set the correct jumpers.cheers___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-Addons compile problem (cdr_addon_mysql.c)
Hi Ken,I did a clean install of FreePBX n Asterisk with asterisk-addons, sounds. Have you downloaded perl and perl-CPAN? Check this link http://aussievoip.com.au/wiki/index.php?page=freePBX-2.1beta1Install and scroll down to the section about mysql_addon-KimOn 6/28/06, Ken Chan [EMAIL PROTECTED] wrote:Hello,I am trying to install Asterisk-Addon and got the following problem: -cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.occ -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/local/mysql/include-I/usr/local/include/mysql-c -o cdr_addon_mysql.o cdr_addon_mysql.c cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz-L/usr/local/mysql/lib -L/usr/local/lib/mysql-L/usr/local/mysql/lib/mysql/usr/lib/gcc-lib/powerpc-linux/3.2/../../../../powerpc-linux/bin/ld: cdr_addon_mysql.so: undefined versioned symbol name _restfpr_22_x@@libmysqlclient_15 /usr/lib/gcc-lib/powerpc-linux/3.2/../../../../powerpc-linux/bin/ld: failed to set dynamic section sizes: Bad valuecollect2: ld returned 1 exit statusmake: *** [cdr_addon_mysql.so] Error 1rm app_saycountpl.o -I am using Addons-1.2.3.I spent almost 2 days and tried to reinstall mysql but still no luck.Anyone has any idea?Anyone has successfully install freePBX with Asterisk together?Can someone give me a hand? Thanksken--___Search for businesses by name, location, or phone number.-Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.asp?SRC="">___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] flash button on asterisk + legacy pbx system
Hi C F, I read the comments but the problem remains...after some tests, I changed some parameters inside zapata.h and recompiled to make flash button work so now my asterisk knows when the user presses the flash button /during a call./ My problem now is how to transfer the flash signal to the old PBX, infact seems like asterisk accept it (even if I cannot use it inside extensions.conf for example with a _FLASH,1,...) but then doesn't re-send it to the line. TIA Giorgio Incantalupo C F wrote: Use features.conf, look here at the comments: http://www.voip-info.org/wiki-Asterisk+cmd+flash On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi C F, you say Flash asterisk command send a flash signal to old pbx so that it sees that command as coming from an analog phone. But since Flash is not a digit, how can I catch it from within asterisk? How can I tell asterisk (es inside extensions.conf) to do something whene receive it from a phone? TIA Giorgio Incantalupo C F wrote: The flash command will do just that. However flash only works on FXO ports and not on SIP FXO ATAs, if you use the later then you will have to find out how your ATA supports it. The easiest way to set this up is to use the features.conf On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I have to connect an old PBX to a new Asterisk box. but I must keep the same flash button functionality of the old system. Is it possible to tell asterisk to send a Flash signal to old pbx when receiving it from a phone? I know there is a flash command inside asteriskis there anybody who tried and deployed such a double-pbx system with success? TIA Giorgio Incantalupo ___ --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 ___ --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] Asterisk-Addons compile problem (cdr_addon_mysql.c)
On Wed, Jun 28, 2006 at 03:42:45PM -0500, Ken Chan wrote: Hello, I am trying to install Asterisk-Addon and got the following problem: - cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/local/mysql/include -I/usr/local/include/mysql-c -o cdr_addon_mysql.o cdr_addon_mysql.c cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz-L/usr/local/mysql/lib -L/usr/local/lib/mysql -L/usr/local/mysql/lib/mysql /usr/lib/gcc-lib/powerpc-linux/3.2/../../../../powerpc-linux/bin/ld: cdr_addon_mysql.so: undefined versioned symbol name _restfpr_22_x@@libmysqlclient_15 /usr/lib/gcc-lib/powerpc-linux/3.2/../../../../powerpc-linux/bin/ld: failed to set dynamic section sizes: Bad value collect2: ld returned 1 exit status make: *** [cdr_addon_mysql.so] Error 1 rm app_saycountpl.o - I am using Addons-1.2.3. I spent almost 2 days and tried to reinstall mysql but still no luck. Anyone has any idea? Anyone has successfully install freePBX with Asterisk together? Can someone give me a hand? And your distro is? And the version of Asterisk is? And the version of mysql is? And the rest of the compilation trace is? (at least we know that the platform is linux/ppc) -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Nokia E61
Thanks guys. How about the quality of the call etc? Are you happy with the phone, do you recommend them? On 7/4/06, Antonio Rabena [EMAIL PROTECTED] wrote: Hi, configuration for E61 is the same as E60. As for the codec, G729 works between E60/61 phones (G729 passthru). At 03:44 PM 7/4/2006, you wrote: Devraj Mukherjee wrote: Hello world, Any success stories of getting a Nokia E61 to work with Asterisk server? Interested to hear before we buy them for work :) I don't know about e61, but I connected an e60 up yesterday that wasn't any hassle. Even the stories about poor quality with WPA + G.729 seemed to be false. ___ --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
[Asterisk-Users] Calling Extensions generates congestion when call answered
Hi,I just installed freePBX n Asterisk (Fedora 5, ast*1.2.9.1)and they are working well except when i created 2 extensions i.e n 1235, when i try to call either from my SIP Phones, when i pick the call from one of the extension, the call fails and i hear a ¨busy tone¨. Another problem arrises when if the call dials for more than 10s, the call fails and generates a busy tone. Ive attached my log file Thanx,Kim ___ --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] Calling Extensions generates congestion when call answered
On Tue, Jul 04, 2006 at 01:49:31PM +0300, Levis Kimotho wrote: Hi, I just installed freePBX n Asterisk (Fedora 5, ast*1.2.9.1)and they are working well except when i created 2 extensions i.e n 1235, when i try to call either from my SIP Phones, when i pick the call from one of the extension, the call fails and i hear a ¨busy tone¨. Another problem arrises when if the call dials for more than 10s, the call fails and generates a busy tone. Ive attached my log file No, you haven't. Or maybe it was cut away by the list server. In that case, add a small call trace inline. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Does asterisk support outbound fax?
Hi all, I am running asterisk 1.2.9 + digium te110p Does my setup about support outbound fax? Regards, rootlinux __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Queues and annoucements
On 7/3/06, Tristan [EMAIL PROTECTED] wrote: Hi everybody ! I need to play a file to customers when an agent answered the line to tell them it's their turn but I don't want to do periodic annoucements, Is there a way or something I misunderstood in the voip.org docs because I can't do this for the moment ! The /trunk version of app_queue has the ability to fire off an AGI against the channel once it's about to be bridged with a queue member. You could probably use this hook to do what you're looking to do. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Queues and annoucements
Good news !!! Do you think I'll be able to use /trunk version of app_queue against 1.2.9.1 ? Or what (stable) version should I'll be looking to use this ? Thanks, Tristan BJ Weschke a écrit : On 7/3/06, Tristan [EMAIL PROTECTED] wrote: Hi everybody ! I need to play a file to customers when an agent answered the line to tell them it's their turn but I don't want to do periodic annoucements, Is there a way or something I misunderstood in the voip.org docs because I can't do this for the moment ! The /trunk version of app_queue has the ability to fire off an AGI against the channel once it's about to be bridged with a queue member. You could probably use this hook to do what you're looking to do. ___ --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] IVR menus on different DIDs
So ...where can I get some help on my problem?thxchristian ___ --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] Need help with config-files
Hello list, I'm a asterisk-beginner and could use some assistance with my configfiles(sip.conf extensions.conf). I'll attach them to this mail, and I hope some of you prof's can give me some advice or point me in the right direction. At the moment by using this configuration, and call somebody internal i get instant voicemail, and sometimes a call isn't even possible. Please help! :) Best Regards, Thomas Jacobsen extensions.conf Description: Binary data sip.conf Description: Binary data ___ --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] IVR menus on different DIDs
Looking (not so deep) at Your logs... mostly cdr INSERT INTO... for me, it's looks that both calls was handled in [iax] context so... You specified [ivr-menu-number-context] - how do You make a jump to such context ? You have 3 dids 10,20,0 and 3 context,don't you ? where dids patterns are matched and proper jumps are executed ? in my opionion there should be somethnig like: calls incoming on did line go to [iax] context [iax] first_did,1,goto(first_did_context|s|1) second_did,1,goto(second_did_context|s|1) third_did,1,goto(third_did_context|s|1) did You try something like this ? F. So ... where can I get some help on my problem? thx christian ___ --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
[Asterisk-Users] Quintum A400 Call Establishment Prob
Hi, I have a little problem related to quintum a400 gateway. I have installed asterisk-1.2.8. Have configured it with SIP and H323 channels to recieve and make calls over lan using softphone (shphone for both SIP and H323). H323 driver version is openh323-v1.17.1 and pwlib-v1.9.0 . pc to pc calls thru asterisk are established without any problem. Recently i connected a quintum a400 gateway to the lan. quintum is programmed like whenever it recieves a call request, it should forward the request to asterisk server, and it does with no error. after call being forwarded to asterisk, asterisk uses 's' extensions to handle the call. initially i am using the following extension: exten=s,1,Dial(H323/192.168.0.23,20) ;23 is the ip address of pc using softphone. a digital phone (simple one which we use for direct pstn comm) is connected with the 1st pbx port of quintum. we dial quintum extension, quintum(using h323) forwards the call to asterisk, asterisk dials the ip .23, softphone rings, as we answere the phone the call gets disconnected atuomatically. SIP account ends up with the same result. here is the log info: H323 LOG == Starting H323/ip$192.168.0.22:24602/21 at default,15,1 failed so falling back to exten 's' -- Executing Dial(H323/ip$192.168.0.22:24602/21, H323/192.168.0.23/20) in new stack -- Called 192.168.0.23/20 Jul 4 16:17:23 WARNING[2955]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/192.168.0.23-2(-2033656) to H323/ip$192.168.0.22:24602/21(-2033656) -- H323/192.168.0.23-2 is ringing -- H323/192.168.0.23-2 is ringing -- H323/192.168.0.23-2 answered H323/ip$192.168.0.22:24602/21 == Spawn extension (default, s, 1) exited non-zero on 'H323/ip$192.168.0.22:24602/21' I ALSO DONT KNOW THE REASON WHAT THIS WARNING IS ABOUT SIP LOG the same thing happens for sip account but without the warning. H323 Channel Configuration [laptopAsus] type=friend host=192.168.0.23 context=default SIP Channel Configuration [Ammad] type=friend secret=tu qualify=4000 nat=yes host=dynamic canreinvite=no context=default I have no idea how to solve this problem. already tried to use different codecs but no progress..plz help ___ --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] Need help with config-files
Hi Thomas, Thomas Jacobsen wrote: Hello list, I'm a asterisk-beginner and could use some assistance with my configfiles(sip.conf extensions.conf). I'll attach them to this mail, and I hope some of you prof's can give me some advice or point me in the right direction. At the moment by using this configuration, and call somebody internal i get instant voicemail, and sometimes a call isn't even possible. Please help! :) I'm also a beginner, so I have just hints. exten = s,1,Background(/home/thomas/banestroeget) exten = s,2,DigitTimeout(3) exten = s,3,WaitExten(3) exten = s,4,Dial(SIP/1001SIP/1002SIP/1003) exten = s,5,Hangup exten = i,1,Dial(SIP/1001SIP/1002SIP/1003) exten = i,2,Hangup include = internal include = office-main include = utilities include = internal include = outbound-46932400 I'll take a deeper look into your files a bit later. Kai ___ --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] Need help with config-files
Your config files are not properly attached. i cant open them. send them again.On 7/4/06, Thomas Jacobsen [EMAIL PROTECTED] wrote:Hello list,I'm a asterisk-beginner and could use some assistance with my configfiles(sip.conf extensions.conf). I'll attach them to this mail,and I hope some of you prof's can give me some advice or point me in theright direction. At the moment by using this configuration, and call somebody internal i get instant voicemail, and sometimes a call isn'teven possible. Please help! :)Best Regards,Thomas Jacobsen___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with config-files
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten = _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten = _ZX[0-8]X,2,Dial(SIP/${temp})
exten = _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten = _ZX[0-8]X,102,Goto(${EXTEN},3)
Shouldn't this be 103?
exten = _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten = _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten = _ZX[0-8]X,5,Hangup
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Re: [Asterisk-Users] Need help with config-files
I can read them, so they are properly attached. Check You mail software. Your config files are not properly attached. i cant open them. send them again. ___ --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] Need help with config-files
Sorry I changed my text and deleted too much :-S Kai Fürstenberg wrote: Hi Thomas, Thomas Jacobsen wrote: Hello list, I'm a asterisk-beginner and could use some assistance with my configfiles(sip.conf extensions.conf). I'll attach them to this mail, and I hope some of you prof's can give me some advice or point me in the right direction. At the moment by using this configuration, and call somebody internal i get instant voicemail, and sometimes a call isn't even possible. Please help! :) I'm also a beginner, so I have just hints. exten = s,1,Background(/home/thomas/banestroeget) exten = s,2,DigitTimeout(3) exten = s,3,WaitExten(3) exten = s,4,Dial(SIP/1001SIP/1002SIP/1003) exten = s,5,Hangup exten = i,1,Dial(SIP/1001SIP/1002SIP/1003) exten = i,2,Hangup include = internal include = office-main include = utilities include = internal include = outbound-46932400 Why do you include internal twice? Kai ___ --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] Now that Nufone is dead...
Try Termilink. www.termilink.net -- Original message -- From: "Carlos Chavez" [EMAIL PROTECTED] Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] Best VoIP provider for Asterisk
Termilink, at www.termilink.net -- Original message -- From: "C F" [EMAIL PROTECTED] Define best. On 5/23/06, Crazy Boy <[EMAIL PROTECTED]>wrote: Hi Friends, Can you please tell me who is the best VoIP Service Provider using Asterisk (With trail version for sometime) . Waiting for your quick response. Thank you. Regards, Chandra. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing l ist 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
[Asterisk-Users] Help getting International Dialing setup in extensions.conf
I am having trouble setting up international dialing. I have an asterisk
server connected to a PRI at our collocation. I have this setup in my
extensions.conf file, yet I still cannot get connected to international
calls.
[OUTBOUND]
exten = _9011.,1,SetCIDNum(XXX-XXX-|a)
exten = _9011.,2,SetCIDName(Some Company|a)
exten = _9011.,3,Dial(Zap/g1/${EXTEN:1},60)
exten = _9011.,4,Playback(invalid)
When I called the support line at my collocation, they mentioned that I
am not setting the call type correctly in the D channel, but I am not
real familiar with PRI lines. Where does this need to be setup.
Also, I am not sure what this portion ${EXTEN:1} means from my above
extensions.conf file, is this the root cause?
Any pointers or real world examples on how to get international dialing
working in Asterisk would be much appreciated.
Thanks
Von L.
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Re: [Asterisk-Users] Need help with config-files
Does phones are registered in Asterisk ? (CLIsip show peers)
CLI log showing such connections will be usefull (no debug for now).
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten = _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten = _ZX[0-8]X,2,Dial(SIP/${temp})
exten = _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten = _ZX[0-8]X,102,Goto(${EXTEN},3)
Shouldn't this be 103?
if 103 is there so failing on 2nd priority will go to 3rd priority...
failing on DBget will go to 3rd and this looks ok for me.
Failing on DBget and user not registered will always go to voicemail
exten = _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten = _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten = _ZX[0-8]X,5,Hangup
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[Asterisk-Users] Libpri + Zaptel + Asterisk polycom_acd_functions error message
I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make install), there was no errors. I used svn to get the polycom_acd_functions asterisk branch release 30432, I have to run make 3 times as it as it comes up with making opts re-run make. It then completes and I run make install, and get the following error message. chan_zap.c:73:2: #error You need newer libpri chan_zap.c:113:2: #error Your zaptel is too old. please update Does anybody know why I'm getting these error message, as I have the newest versions of both? Thanks Dean. ___ --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] Need help with config-files
Filip Drągowski wrote: I can read them, so they are properly attached. Check You mail software. Your config files are not properly attached. i cant open them. send them again. They are attached as: Content-Type: application/octet-stream; name=extensions.conf Content-Transfer-Encoding: 8bit but it is text. So on pricipal they are not attached correctly. But with a proper mail client you are able to open the attachments by choosing a software to open with. Kai ___ --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] Need help with config-files
Hello, It was a mistake(because i edited them before i send them to the list, they are not original like that). Best Regards, Thomas On Tue, 2006-07-04 at 15:19 +0200, Kai Fürstenberg wrote: Sorry I changed my text and deleted too much :-S Kai Fürstenberg wrote: Hi Thomas, Thomas Jacobsen wrote: Hello list, I'm a asterisk-beginner and could use some assistance with my configfiles(sip.conf extensions.conf). I'll attach them to this mail, and I hope some of you prof's can give me some advice or point me in the right direction. At the moment by using this configuration, and call somebody internal i get instant voicemail, and sometimes a call isn't even possible. Please help! :) I'm also a beginner, so I have just hints. exten = s,1,Background(/home/thomas/banestroeget) exten = s,2,DigitTimeout(3) exten = s,3,WaitExten(3) exten = s,4,Dial(SIP/1001SIP/1002SIP/1003) exten = s,5,Hangup exten = i,1,Dial(SIP/1001SIP/1002SIP/1003) exten = i,2,Hangup include = internal include = office-main include = utilities include = internal include = outbound-46932400 Why do you include internal twice? Kai ___ --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] Need help with config-files
On Tue, 2006-07-04 at 15:16 +0200, Kai Fürstenberg wrote:
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten = _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten = _ZX[0-8]X,2,Dial(SIP/${temp})
exten = _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten = _ZX[0-8]X,102,Goto(${EXTEN},3)
Shouldn't this be 103?
I'm not sure, I supposed that if the error occured at step 2, the
errorhandling should be 102?
exten = _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten = _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten = _ZX[0-8]X,5,Hangup
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Re: [Asterisk-Users] Calling Extensions generates congestion when call answered
Hi,Below is part of the log file
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Caller ID name is 'LAN201' number is '1235'
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Methodology of ring is 'none'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Added extension to extension map
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '' in family 'CF'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Extension cf is disabled
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '' in family 'DND'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Extension do not disturb is disabled
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '' in family 'CW'
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '' in family 'CFB'
Jul 4 16:38:06 DEBUG[5871] db.c: Unable to find key '' in family 'CFU'
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command 'login'
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Parsing
'/etc/asterisk/manager.conf': Jul 4 16:38:06 VERBOSE[5876] logger.c: ==
Parsing '/etc/asterisk/manager.conf': Found
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Parsing
'/etc/asterisk/manager_additional.conf': Jul 4 16:38:06 VERBOSE[5876]
logger.c: == Parsing '/etc/asterisk/manager_additional.conf': Found
Jul 4 16:38:06 DEBUG[5876] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer
Jul 4 16:38:06 WARNING[5876] acl.c: 255.255.255.0127.0.0.1/255.255.255.0 is not a valid netmask
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Manager 'admin' logged on from 127.0.0.1
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command 'ExtensionState'
Jul 4 16:38:06 DEBUG[5876] manager.c: Manager received command 'Logoff'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Checking CW and CFB status for extension
Jul 4 16:38:06 VERBOSE[5876] logger.c: == Manager 'admin' logged off from 127.0.0.1
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: DbSet CALLTRACE/ to 1235
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- AGI Script dialparties.agi completed, returning 0
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- Executing Dial(SIP/1235-220e, SIP/|15|tr) in new stack
Jul 4 16:38:06 DEBUG[5871] chan_sip.c: Setting NAT on RTP to 0
Jul 4 16:38:06 DEBUG[5871] chan_sip.c: Outgoing Call for
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- Called
Jul 4 16:38:06 DEBUG[4873] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Jul 4 16:38:06 DEBUG[4873] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- SIP/-bde7 is ringing
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Acked pending invite 102
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Oooh, we need to change our
formats since our peer supports only 0x1 (g723) and not 0x4 (ulaw)
Jul 4 16:38:10 WARNING[4873] channel.c: Unable to find a codec translation path from g723 to ulaw
Jul 4 16:38:10 WARNING[4873] channel.c: Unable to find a codec translation path from g723 to ulaw
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: build_route: Contact hop: Muriuki
Jul 4 16:38:10 VERBOSE[5871] logger.c: -- SIP/-bde7 answered SIP/1235-220e
Jul 4 16:38:10 WARNING[5871] channel.c: No path to translate from SIP/1235-220e(4) to SIP/-bde7(1)
Jul 4 16:38:10 WARNING[5871] app_dial.c: Had to drop call because I couldn't make SIP/1235-220e compatible with SIP/-bde7
Jul 4 16:38:10 DEBUG[5871] chan_sip.c: update_call_counter() - decrement call limit counter
Jul 4 16:38:10 VERBOSE[5871] logger.c: == Spawn extension (macro-dial,
s, 10) exited non-zero on 'SIP/1235-220e' in macro 'dial'
Jul 4 16:38:10 DEBUG[4873] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Match
Found
Jul 4 16:38:10 VERBOSE[5871] logger.c: == Spawn extension (macro-dial,
s, 10) exited non-zero on 'SIP/1235-220e' in macro 'exten-vm'
Jul 4 16:38:10 VERBOSE[5871] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/1235-220e'
Jul 4 16:38:10 DEBUG[5871] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Jul 4 16:38:10 DEBUG[5871] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2006-07-04 16:38:02','\LAN201\
1235','1235','','from-internal',
'SIP/1235-220e','SIP/-bde7','Dial','SIP/|15|tr',8,0,'NO
ANSWER',3,'1235','1152020282.0')
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is 'LAN201 1235'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is '1235'
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function result is ''
Jul 4 16:38:10 DEBUG[5871] pbx.c: Function
Re: [Asterisk-Users] Need help with config-files
On Tue, 2006-07-04 at 15:16 +0200, Kai Fürstenberg wrote:
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten = _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten = _ZX[0-8]X,2,Dial(SIP/${temp})
exten = _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten = _ZX[0-8]X,102,Goto(${EXTEN},3)
Shouldn't this be 103?
I'm not sure, I supposed that if the error occured at step 2, the
errorhandling should be 102?
exten = _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten = _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten = _ZX[0-8]X,5,Hangup
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[Asterisk-Users] Quintum A400 Configuration
Hi, can anybody tell me where can i find help for configuring quintum gateway with asterisk? ___ --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] Need help with config-files
Hello, It was a mistake(because i edited them before i send them to the list, they are not original like that). Best Regards, Thomas On Tue, 2006-07-04 at 15:19 +0200, Kai Fürstenberg wrote: Sorry I changed my text and deleted too much :-S Kai Fürstenberg wrote: Hi Thomas, Thomas Jacobsen wrote: Hello list, I'm a asterisk-beginner and could use some assistance with my configfiles(sip.conf extensions.conf). I'll attach them to this mail, and I hope some of you prof's can give me some advice or point me in the right direction. At the moment by using this configuration, and call somebody internal i get instant voicemail, and sometimes a call isn't even possible. Please help! :) I'm also a beginner, so I have just hints. exten = s,1,Background(/home/thomas/banestroeget) exten = s,2,DigitTimeout(3) exten = s,3,WaitExten(3) exten = s,4,Dial(SIP/1001SIP/1002SIP/1003) exten = s,5,Hangup exten = i,1,Dial(SIP/1001SIP/1002SIP/1003) exten = i,2,Hangup include = internal include = office-main include = utilities include = internal include = outbound-46932400 Why do you include internal twice? Kai ___ --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: SV: [Asterisk-Users] Running 40 active calls (too much f�r CPU?)
Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized? -- Original message -- From: [EMAIL PROTECTED] I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Running 40 active calls (too much för CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 < BR> ; of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: h ttp:// 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
Re: [Asterisk-Users] flash button on asterisk + legacy pbx system
Sorry I didn't realize this is how you wanted it to work - that the user is on a FXS and you want when the user flashes that it flashes the host pbx. I disagree with you on this setup the user should be requried to press some DTMF and not just flash the phone. The main reason being that otherwise you will lose 3way and callwaiting features on asterisk. I'm assuming your answer to this is that you don't care since you just want to make the phone an extended extension on the host PBX, and want it to be as much an extension of the old PBX as posible. I still disagree because as much as you are going to try, your users will still not see this as a direct extension, and sooner or later you/they will have to learn how to deal with it anyhow. On 7/4/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi C F, I read the comments but the problem remains...after some tests, I changed some parameters inside zapata.h and recompiled to make flash button work so now my asterisk knows when the user presses the flash button /during a call./ My problem now is how to transfer the flash signal to the old PBX, infact seems like asterisk accept it (even if I cannot use it inside extensions.conf for example with a _FLASH,1,...) but then doesn't re-send it to the line. TIA Giorgio Incantalupo C F wrote: Use features.conf, look here at the comments: http://www.voip-info.org/wiki-Asterisk+cmd+flash On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi C F, you say Flash asterisk command send a flash signal to old pbx so that it sees that command as coming from an analog phone. But since Flash is not a digit, how can I catch it from within asterisk? How can I tell asterisk (es inside extensions.conf) to do something whene receive it from a phone? TIA Giorgio Incantalupo C F wrote: The flash command will do just that. However flash only works on FXO ports and not on SIP FXO ATAs, if you use the later then you will have to find out how your ATA supports it. The easiest way to set this up is to use the features.conf On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I have to connect an old PBX to a new Asterisk box. but I must keep the same flash button functionality of the old system. Is it possible to tell asterisk to send a Flash signal to old pbx when receiving it from a phone? I know there is a flash command inside asteriskis there anybody who tried and deployed such a double-pbx system with success? TIA Giorgio Incantalupo ___ --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 ___ --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] Need help with config-files
Hello,
Yes all phones and trunks are registered.
/Thomas
On Tue, 2006-07-04 at 15:30 +0200, Filip Drągowski wrote:
Does phones are registered in Asterisk ? (CLIsip show peers)
CLI log showing such connections will be usefull (no debug for now).
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten = _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten = _ZX[0-8]X,2,Dial(SIP/${temp})
exten = _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten = _ZX[0-8]X,102,Goto(${EXTEN},3)
Shouldn't this be 103?
if 103 is there so failing on 2nd priority will go to 3rd priority...
failing on DBget will go to 3rd and this looks ok for me.
Failing on DBget and user not registered will always go to voicemail
exten = _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten = _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten = _ZX[0-8]X,5,Hangup
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Re: [Asterisk-Users] Help getting International Dialing setup in extensions.conf
Hi,
Von L. wrote:
I am having trouble setting up international dialing. I have an asterisk
server connected to a PRI at our collocation. I have this setup in my
extensions.conf file, yet I still cannot get connected to international
calls.
[OUTBOUND]
exten = _9011.,1,SetCIDNum(XXX-XXX-|a)
exten = _9011.,2,SetCIDName(Some Company|a)
exten = _9011.,3,Dial(Zap/g1/${EXTEN:1},60)
exten = _9011.,4,Playback(invalid)
When I called the support line at my collocation, they mentioned that I
am not setting the call type correctly in the D channel, but I am not
real familiar with PRI lines. Where does this need to be setup.
Also, I am not sure what this portion ${EXTEN:1} means from my above
extensions.conf file, is this the root cause?
It means that the first digit of your dialed extension will be cut off.
In other words:
A call to 9011something will be routed to 011something.
Kai
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Re: [Asterisk-Users] IVR menus on different DIDs
i didn't thought of that, and i tried it - it works when i use the Goto commandbefore i had one incoming context like [iax] which includes the different sub-contexts ofthe three ivr-menus - and the menupoints of the first listet included context were played. Interesting.thanks to you Filip !On 7/4/06, Filip Drągowski [EMAIL PROTECTED] wrote:Looking (not so deep) at Your logs... mostly cdr INSERT INTO...for me, it's looks that both calls was handled in [iax] context so...You specified [ivr-menu-number-context] - how do You make a jump to suchcontext ?You have 3 dids 10,20,0 and 3 context,don't you ?where dids patterns are matched and proper jumps are executed ? in my opionion there should be somethnig like:calls incoming on did line go to [iax] context[iax]first_did,1,goto(first_did_context|s|1)second_did,1,goto(second_did_context|s|1)third_did,1,goto(third_did_context|s|1) did You try something like this ?F. So ... where can I get some help on my problem? thx christian ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel 1.2.6 / Upgrade Problem
I've encountered a strange problem in what I thought would be a straightforward upgrade to Asterisk 1.2 and was hoping someone out here may have run into something similar. The system is Linux FC3 with a 2.6.9 kernel. The problem is that the new wctdm module will not load during modprobe. Everything compiles and builds just fine. I have search the 'net and forums for typical solutions and even tried to build from the latest source code, but the results are still the same. I ran "make linux26; make install; make install-udev; make config" and verified that everything was okay, including the udev configuration. I rebooted, audited my configuration files (the server has a 4-port FXS TDM400P card), but still no go. modprobe zaptel this works just fine but when I try to load the module modprobe wctdm I get the following returned on the command line: FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667smp/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) and the following in /var/log/messages: kernel: wctdm: disagrees about version of symbol zt_receive kernel: wctdm: Unknown symbol zt_receive kernel: wctdm: disagrees about version of symbol zt_qevent_lock kernel: wctdm: Unknown symbol zt_qevent_lock kernel: wctdm: disagrees about version of symbol zt_ec_chunk kernel: wctdm: Unknown symbol zt_ec_chunk kernel: wctdm: disagrees about version of symbol zt_transmit kernel: wctdm: Unknown symbol zt_transmit kernel: wctdm: disagrees about version of symbol zt_unregister kernel: wctdm: Unknown symbol zt_unregister kernel: wctdm: disagrees about version of symbol zt_hooksig kernel: wctdm: Unknown symbol zt_hooksig kernel: wctdm: disagrees about version of symbol zt_register kernel: wctdm: Unknown symbol zt_register Is it possible there is something left over from the previous zaptel installation that is causing this mismatch? I'm not sure where to look and any help would be appreciated. I've already verified the CCITT module is present and tried everything else I could dig up to resolve this. If I can't I'll need to rollback to a previous version of the zaptel drivers. Thanks in advance for your time and help. ___ --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] Libpri + Zaptel + Asterisk polycom_acd_functions error message
On 7/4/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make install), there was no errors. I used svn to get the polycom_acd_functions asterisk branch release 30432, I have to run make 3 times as it as it comes up with making opts re-run make. It then completes and I run make install, and get the following error message. chan_zap.c:73:2: #error You need newer libpri chan_zap.c:113:2: #error Your zaptel is too old. please update Does anybody know why I'm getting these error message, as I have the newest versions of both? You need the /trunk versions of libpri and zaptel instead of the branches/1.2 releases. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Need help with config-files
Filip Drągowski wrote:
Does phones are registered in Asterisk ? (CLIsip show peers)
CLI log showing such connections will be usefull (no debug for now).
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten = _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten = _ZX[0-8]X,2,Dial(SIP/${temp})
exten = _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten = _ZX[0-8]X,102,Goto(${EXTEN},3)
Shouldn't this be 103?
if 103 is there so failing on 2nd priority will go to 3rd priority...
failing on DBget will go to 3rd and this looks ok for me.
Failing on DBget and user not registered will always go to voicemail
So in this config, it will be the same if he leaves away priority 102.
exten = _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten = _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten = _ZX[0-8]X,5,Hangup
And if then VoiceMail fails for some reason, it will just hangup, right?
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[Asterisk-Users] H323 Asterisk best practices
I recently have been required to terminate traffic via H323. We have beensuccessfully handling this traffic as SIP. We often have 30 + concurrent calls on this server and I am not quite sure the best way to handle this new H322 traffic. Which of the h323 channels for * can handle this traffic reliably? Any suggestions would be greatly appreciated. Thanks,JC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help getting International Dialing setup in extensions.conf
I am having trouble setting up international dialing. I have an asterisk
server connected to a PRI at our collocation. I have this setup in my
extensions.conf file, yet I still cannot get connected to international
calls.
[OUTBOUND]
exten = _9011.,1,SetCIDNum(XXX-XXX-|a)
exten = _9011.,2,SetCIDName(Some Company|a)
exten = _9011.,3,Dial(Zap/g1/${EXTEN:1},60)
exten = _9011.,4,Playback(invalid)
When I called the support line at my collocation, they mentioned that I
am not setting the call type correctly in the D channel, but I am not
real familiar with PRI lines. Where does this need to be setup.
Also, I am not sure what this portion ${EXTEN:1} means from my above
extensions.conf file, is this the root cause?
${EXTEN:1} = Dial(Zap/g1/011[dialednumber],60)
${EXTEN:1} cut 1 leading digit from EXTEN
Any pointers or real world examples on how to get international dialing
working in Asterisk would be much appreciated.
Did You try not to set CIDNum and CIDName ?
only Dial.
What number is dialed for international call ?
For poland is 0 + coutry code + area code + number
in You dialplan i can't dial international.
i would dial 90110+countr+area+number dialplan will cut 9
and the number send to telco would be 0110+coutry+area+number
Compare it with Your international dialing rules.
(i got no other ideas for now)
Thanks
Von L.
F.
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[Asterisk-Users] Mediatrix 1204 and Asterisk
Hi Everyone, I am new to Asterisk but I have found that quite a few people have implemented it with the Mediatrix 1204. Does anyone know of a wiki or place where there is good documentation regarding this configuration? Thanks Julian___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help with config-files
I'm sorry for the double posts.
On Tue, 2006-07-04 at 15:52 +0200, Thomas Jacobsen wrote:
On Tue, 2006-07-04 at 15:16 +0200, Kai Fürstenberg wrote:
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten = _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten = _ZX[0-8]X,2,Dial(SIP/${temp})
exten = _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten = _ZX[0-8]X,102,Goto(${EXTEN},3)
Shouldn't this be 103?
I'm not sure, I supposed that if the error occured at step 2, the
errorhandling should be 102?
exten = _ZX[0-8]X,4,VoiceMail(u${EXTEN})
exten = _ZX[0-8]X,104,VoiceMail(b${EXTEN})
exten = _ZX[0-8]X,5,Hangup
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Re: [Asterisk-Users] Now that Nufone is dead...
Who says nufone is dead? I use them, but my did is through sellvoip.net ___ --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] SIP -- H323 RTP Questions (1 WAY Audio only)
Hello, I have been trying to get the SIP -- H323 working in the last few weeks. I tried different H323 channel drivers. I need help badly. I got SIP -- SIP (with canreinvite=yes) and it was working fine. So, I believe the problem is not in SIP side. Here are my problems: a) I am currently using Asterisk-Addon ooh323 channel driver. I dial from SIP to OpenPhone. I have 1 way audio only (the voice from OpenPhone to SIP is fine. But there is no voice from SIP to OpenPhone). The signalling part looks good. At least I could make a call and hang up the call. Anyone has any idea why I had 1 way audio? All the phones and Asterisk are on the same LAN. Here is part of my ooh323.conf file: [general] port=1720 bindaddr=10.3.3.239 [ken_op] type=peer context=default ip=10.1.1.155 port=1720 allow=ulaw dtmfmode=rfc2833 Here is part of my extensions.conf file: exten = 6111,1,Dial(SIP/voip6111,20) exten = 7401,1,Dial(OOH323/ken_op,20) b)After I established a call, I typed rtp debug to enable the debug for RTP. Seems to me that the RTP packets ARE GOING through Asterisk. It is normal? Can someone that had success on setting up SIP -- h323 (Asterisk-Addon 1.2.3) please provide me some more information (such as conf files and hints how to solve my problems). Ken -- ___ Search for businesses by name, location, or phone number. -Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.asp?SRC=lycos10 ___ --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] how to send flash command from asterisk to old pbx when pressing button on phone
Hi, I connected an Asterisk box to an old pbx using a TDM400P (one fxs and one fxo). Then I connected an analog phone to Asterisk FXS port. Is it possible to send a flash command to old pbx via asterisk box when pressing flash button on the analog phone? When I press the flash button the console shows asterisk putting the call in hold: -- Starting simple switch on 'Zap/3-1' -- Executing Dial(Zap/3-1, SIP/linux) in new stack -- Called linux -- SIP/linux-739c is ringing -- SIP/linux-739c answered Zap/3-1 When pressing flash button: -- Starting simple switch on 'Zap/3-2' -- Started three way call on channel 3 -- Started music on hold, class 'default', on channel 'SIP/linux-739c' -- Stopped music on hold on SIP/linux-739c This point I hear a strange sound. Then pressing the flash button again: -- Dumping incomplete call on on Zap/3-1 -- Hungup 'Zap/3-2' I'd like asterisk send the flash command to the old pbx. The flash command does not work. Is there anybody who could interface asterisk and a normal pbx so that the users could use the same phones in the same way? TIA Giorgio Incantalupo ___ --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
SV: SV: [Asterisk-Users] Running 40 active calls (to o much för CPU?)
Phones are not behind NAT. Every client is on the sameinternal network as the asterisk pbx (nothing is sent throughthe internet). It's not the network since I tested this by calling asterisk from an outside phone (cell) and let asterisk play a message for me. Same "cutting" and "chopping" when many SIP-clients where active in a call at the same time. Computer RAM is 2 gb. If the E1 is channelized or not I don't actually know. How would I know this and why would it affect the call quality when many people are in a call at the same time (same lines work fine with an Ericsson BusinessPhone Exchange)? Thanks! Regards, Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]Skickat: den 4 juli 2006 15:55Till: Asterisk Users Mailing List - Non-Commercial DiscussionÄmne: Re: SV: [Asterisk-Users] Running 40 active calls (too much för CPU?) Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized? -- Original message -- From: [EMAIL PROTECTED] I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Running 40 active calls (too much för CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 BR ; of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: h ttp:// 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
Re: [Asterisk-Users] Need help with config-files
Thomas Jacobsen wrote:
On Tue, 2006-07-04 at 15:16 +0200, Kai Fürstenberg wrote:
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody internal i get instant voicemail, and sometimes a call isn't
even possible. Please help! :)
BTW:
[internal]
exten = _ZX[0-8]X,1,DBget(temp=CFIM/${EXTEN})
exten = _ZX[0-8]X,2,Dial(SIP/${temp})
exten = _ZX[0-8]X,3,Dial(SIP/${EXTEN},20)
exten = _ZX[0-8]X,102,Goto(${EXTEN},3)
Shouldn't this be 103?
I'm not sure, I supposed that if the error occured at step 2, the
errorhandling should be 102?
Oh, I thought this should be Busy-handling.
Do you have any console outputs, so that we know what aterisk is doing
exactly?
Kai
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Re: [Asterisk-Users] trixbox 1.1 download
On Thu, 2006-08-03 at 11:40 +0300, Khaled Chehab wrote: I have trixbox 1.0 how I can update it to 1.1 or from where I can download trixbox 1.1 Obviously the responses are falling on deaf ears so I'll just rinse and repeat. Hopefully it will register this time: 1) trixbox questions should go to trixbox mailinglist or forum. This is the asterisk mailing list, not the trixbox mailinglist 2) fix the time on your PC. it is not August 3rd no matter where you are 3) stop including that silly disclaimer at the end of your email Someone already answered your question. Do you actually *read* the responses you get? Patrick ___ --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] Zaptel 1.2.6 / Upgrade Problem
I just resolved the problem. The older zaptel kernel modules (IIRC) were installed into the /misc/ subdirectory and the newer modules are installed into /extra/. To further complicate matters, I had entries in my /etc/modprobe.conf that were still loading the previous kernel modules from the /misc/ directory and apparently causing the problem. To resolve, I removed all the previous forced module load commands from /etc/modprobe.conf (removing all the zaptel related lines), cleared out /lib/modules/version/misc, ran depmod -a, verified my /etc/sysconfig/zaptel configuration file and rebooted. VOILA! Nothing wrong with 1.2.6, just make sure you really clean out all of any previous install before upgrading. - Jerry Jerry Brady wrote: I've encountered a strange problem in what I thought would be a straightforward upgrade to Asterisk 1.2 and was hoping someone out here may have run into something similar. The system is Linux FC3 with a 2.6.9 kernel. The problem is that the new wctdm module will not load during modprobe. Everything compiles and builds just fine. I have search the 'net and forums for typical solutions and even tried to build from the latest source code, but the results are still the same. I ran "make linux26; make install; make install-udev; make config" and verified that everything was okay, including the udev configuration. I rebooted, audited my configuration files (the server has a 4-port FXS TDM400P card), but still no go. modprobe zaptel this works just fine but when I try to load the module modprobe wctdm I get the following returned on the command line: FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667smp/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) and the following in /var/log/messages: kernel: wctdm: disagrees about version of symbol zt_receive kernel: wctdm: Unknown symbol zt_receive kernel: wctdm: disagrees about version of symbol zt_qevent_lock kernel: wctdm: Unknown symbol zt_qevent_lock kernel: wctdm: disagrees about version of symbol zt_ec_chunk kernel: wctdm: Unknown symbol zt_ec_chunk kernel: wctdm: disagrees about version of symbol zt_transmit kernel: wctdm: Unknown symbol zt_transmit kernel: wctdm: disagrees about version of symbol zt_unregister kernel: wctdm: Unknown symbol zt_unregister kernel: wctdm: disagrees about version of symbol zt_hooksig kernel: wctdm: Unknown symbol zt_hooksig kernel: wctdm: disagrees about version of symbol zt_register kernel: wctdm: Unknown symbol zt_register Is it possible there is something left over from the previous zaptel installation that is causing this mismatch? I'm not sure where to look and any help would be appreciated. I've already verified the CCITT module is present and tried everything else I could dig up to resolve this. If I can't I'll need to rollback to a previous version of the zaptel drivers. Thanks in advance for your time and help. ___ --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] Need help with config-files
Hello, I decided to resend the files, because i made alot of typos in them. - Please use these files instead. Best Regards, Thomas extensions.conf Description: Binary data sip.conf Description: Binary data ___ --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] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with -c for color CLI?
Salve *! I'm using asterisk for a while and now I want to have a colord CLI. I have apt-get install asterisk/testing, that is asterisk 1.2.7.1 I use Debian stable/testing on a vserver with any /dev/tty*. So, of course, I comment out #TTY=9 inside /usr/sbin/safe_asterisk. /etc/init.d/asterisk start calls /usr/sbin/safe_asterisk root 5757 1 0 1149 1348 0 16:08 pts/30 00:00:00 /bin/sh/usr/sbin/safe_asterisk -g -vvv -U asterisk asterisk 5758 5757 0 4502 7992 0 16:08 pts/30 00:00:00 \_/usr/sbin/asterisk -g -vvv -U asterisk When I start asterisk by hand with asterisk -cgvvv I got a colored CLI ;) but inside /usr/sbin/safe_asterisk (started by /etc/init.d/asterisk - with -U asterisk) or inside /etc/init.d/asterisk itself all my tries to use -c as additional parameter faild. ;( E.g. I changed line 52 inside etc/init.d/asterisk PARAMS=$PARAMS -U $USER PARAMS=$PARAMS -c -U $USER This does run the CLI with color :) also with color for asterisk -rcvvv *but* there is an fast running repeating output after running /etc/init.d/asterisk start Use EXIT or QUIT to exit the asterisk console Use EXIT or QUIT to exit the asterisk console Use EXIT or QUIT to exit the asterisk console [...] The handycap of vserver is, that as [EMAIL PROTECTED] you can't creat devices yourself. I think asterisk is also reading from /dev/tty9, so a ls -s /dev/null /dev/tty9 does not help (tested). Does somebody have an idea how to deal with this? Can I run asterisk with color CLI without a TTY*? Greetings, rob ___ --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] time variable
I want to get a variable, depending on the time. I tried this one, but it does not work: exten = 75,1,Set(guess=SYSTEM(echo $((1 + $(date +%S)*100 % 23))) The idea is that the variable guess will change every 23 times per minute. How would be the right syntax? bye Ronald Wiplinger ___ --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] VoIP Cheap Asterisk
Have you tried without reinviting ?? (canreinvite=no) Is your * box behind a nat ? maxx Scott Pinhorne ha scritto: Hi All I have setup my asteriks to use voipcheap.com for the outgoing trunk on local calls (because they are free), my setup is below: register = username:[EMAIL PROTECTED] [voipcheap] type=peer host=sip.voipcheap.com domain=voipcheap.com dtmfmode=inband context=mycontext allow=all canreinvite=yes qualify=yes username=username password=password When I start asterisk I am able to make calls out via this trunk but only for a certain period (random) and then after this I get a 503 Forbidden error, if i restart asterisk then it connects and it is ok again for a certain period. The logs show: Forbidden - wrong password on authentication for INVITE but how can this be if I am able to make calls for a while before hand, I have tried playing with various settings but cannot get it constant, if anyone has any ideas then I would be very grateful as it is doing my head in now :-) Many thanks Scott Pinhorne VoxIT.co.uk ___ --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] H323 Asterisk best practices
Joshua Laroff wrote: I recently have been required to terminate traffic via H323. We have beensuccessfully handling this traffic as SIP. We often have 30 + concurrent calls on this server and I am not quite sure the best way to handle this new H322 traffic. Which of the h323 channels for * can handle this traffic reliably? Any suggestions would be greatly appreciated. Thanks, JC -- Hi JC, oh323, which uses OpenH323 is pretty solid and reliable from inaccessnetworks. I like it much more than the other two. There is also something called chan_woomera, a new channel for Asterisk which can hook up to OpenH323 or Opal. try it! -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Recommendations for best Voicemail application manager?
Hello List.I am looking to build an Asterisk Voicemail application to serve approx. 100 users.I will be building the Voicemail system using a standard Asterisk install on a stable Debian system.The system will house 100x20mb/each voicemail boxes. On to my question:The Voicemail system will most likely be maintained by a single person at the customer location, most likely an office admin. I wouuld like the office admin to be able to conduct standard moves/add/changes/resets etc.. Any thoughts on the best WWW UI to provide these moves adds and changes?Thanks!!_Chris_ ___ --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
SV: [Asterisk-Users] Running 40 active calls (too much för CPU?)
Phones are not behind NAT. Every client is on the sameinternal network as the asterisk pbx (nothing is sent throughthe internet). It's not the network since I tested this by calling asterisk from an outside phone (cell) and let asterisk play a message for me. Same "cutting" and "chopping" when many SIP-clients where active in a call at the same time. Computer RAM is 2 gb. If the E1 is channelized or not I don't actually know. How would I know this and why would it affect the call quality when many people are in a call at the same time (same lines work fine with an Ericsson BusinessPhone Exchange)? Thanks! Regards, Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]Skickat: den 4 juli 2006 15:55Till: Asterisk Users Mailing List - Non-Commercial DiscussionÄmne: Re: SV: [Asterisk-Users] Running 40 active calls (too much för CPU?) Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized? -- Original message -- From: [EMAIL PROTECTED] I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Running 40 active calls (too much för CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 BR ; of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: h ttp:// 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
Re: [Asterisk-Users] flash button on asterisk + legacy pbx system
Hi C F, ok, I also thought to make the user to press some keys for example * and 3 so I setup a little test made using an Asterisk box with a TDM400P (2 FXS + 2 FXO) connected to an analog phone (fxs port) and an analog line (fxo port). I searched on internet and found some interesting stuff so I made my extensions.conf: My extension.conf is (in brief): [zap] exten = s,1,Set(DYNAMIC_FEATURES=zapflash) exten = s,2,Dial(Zap/3,15,tw) --- Zap/3 is my analog phone exten = s,3,HangUp My zapata (Zap/1 is the line and Zap/3 is the phone): context = zap language = it signalling = fxs_ks threewaycalling=yes transfer = yes channel = 1 language = it signalling = fxo_ks callerid = tel1 100 threewaycalling=yes transfer = yes channel = 3 and my features.conf: [applicationmap] ... zapflash = *3,caller,flash,() When I call the number xxx, Asterisk answers on zap line passing the call to zap/3. I pick up zap/3 phone and then I press *3 but all I get is (on asterisk console): WARNING[3082]: app_flash.c:101 flash_exec: Zap/3-1 is not an FXO Channel Why? It seems Asterisk sends Flash command to the phone but it is not what I want. Is this the right way to follow? Press *3 (or other code) to send command to host pbx while the callee is on the phone? Is this what you meant? If yes, why Asterisk does not send the flash command to the line? Thanks for patience Giorgio Incantalupo C F wrote: Sorry I didn't realize this is how you wanted it to work - that the user is on a FXS and you want when the user flashes that it flashes the host pbx. I disagree with you on this setup the user should be requried to press some DTMF and not just flash the phone. The main reason being that otherwise you will lose 3way and callwaiting features on asterisk. I'm assuming your answer to this is that you don't care since you just want to make the phone an extended extension on the host PBX, and want it to be as much an extension of the old PBX as posible. I still disagree because as much as you are going to try, your users will still not see this as a direct extension, and sooner or later you/they will have to learn how to deal with it anyhow. On 7/4/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi C F, I read the comments but the problem remains...after some tests, I changed some parameters inside zapata.h and recompiled to make flash button work so now my asterisk knows when the user presses the flash button /during a call./ My problem now is how to transfer the flash signal to the old PBX, infact seems like asterisk accept it (even if I cannot use it inside extensions.conf for example with a _FLASH,1,...) but then doesn't re-send it to the line. TIA Giorgio Incantalupo C F wrote: Use features.conf, look here at the comments: http://www.voip-info.org/wiki-Asterisk+cmd+flash On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi C F, you say Flash asterisk command send a flash signal to old pbx so that it sees that command as coming from an analog phone. But since Flash is not a digit, how can I catch it from within asterisk? How can I tell asterisk (es inside extensions.conf) to do something whene receive it from a phone? TIA Giorgio Incantalupo C F wrote: The flash command will do just that. However flash only works on FXO ports and not on SIP FXO ATAs, if you use the later then you will have to find out how your ATA supports it. The easiest way to set this up is to use the features.conf On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I have to connect an old PBX to a new Asterisk box. but I must keep the same flash button functionality of the old system. Is it possible to tell asterisk to send a Flash signal to old pbx when receiving it from a phone? I know there is a flash command inside asteriskis there anybody who tried and deployed such a double-pbx system with success? TIA Giorgio Incantalupo ___ --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 ___ --Bandwidth and Colocation
Re: [Asterisk-Users] Zaptel 1.2.6 / Upgrade Problem
On Tue, Jul 04, 2006 at 10:06:27AM -0400, Jerry Brady wrote: I've encountered a strange problem in what I thought would be a straightforward upgrade to Asterisk 1.2 and was hoping someone out here may have run into something similar. The system is Linux FC3 with a 2.6.9 kernel. The problem is that the new wctdm module will not load during modprobe. Everything compiles and builds just fine. I have search the 'net and forums for typical solutions and even tried to build from the latest source code, but the results are still the same. I ran make linux26; make install; make install-udev; make config and verified that everything was okay, including the udev configuration. I rebooted, audited my configuration files (the server has a 4-port FXS TDM400P card), but still no go. Assuming that you have the proper kernel-devel package installed, this should work well. modprobe zaptel this works just fine but when I try to load the module modprobe wctdm I get the following returned on the command line: FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667smp/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) and the following in /var/log/messages: kernel: wctdm: disagrees about version of symbol zt_receive kernel: wctdm: Unknown symbol zt_receive kernel: wctdm: disagrees about version of symbol zt_qevent_lock kernel: wctdm: Unknown symbol zt_qevent_lock kernel: wctdm: disagrees about version of symbol zt_ec_chunk kernel: wctdm: Unknown symbol zt_ec_chunk kernel: wctdm: disagrees about version of symbol zt_transmit kernel: wctdm: Unknown symbol zt_transmit kernel: wctdm: disagrees about version of symbol zt_unregister kernel: wctdm: Unknown symbol zt_unregister kernel: wctdm: disagrees about version of symbol zt_hooksig kernel: wctdm: Unknown symbol zt_hooksig kernel: wctdm: disagrees about version of symbol zt_register kernel: wctdm: Unknown symbol zt_register What is the output of 'modinfo zaptel' , 'modinfo wctdm' ls -l on the location of both. Are they both from the recent install (look at the dates) Is it possible there is something left over from the previous zaptel installation that is causing this mismatch? I'm not sure where to look and any help would be appreciated. Maybe you have an old version of zaptel still loded in the memory. rmmod it . I've already verified the CCITT module is present and tried everything else I could dig up to resolve this. If I can't I'll need to rollback to a previous version of the zaptel drivers. This is a problem with zaptel, not with ccitt. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Zaptel 1.2.6 / Upgrade Problem
On Tue, Jul 04, 2006 at 10:53:46AM -0400, Jerry Brady wrote: I just resolved the problem. The older zaptel kernel modules (IIRC) were installed into the /misc/ subdirectory and the newer modules are installed into /extra/. To further complicate matters, I had entries in my /etc/modprobe.conf that were still loading the previous kernel modules from the /misc/ directory and apparently causing the problem. To resolve, I removed all the previous forced module load commands from /etc/modprobe.conf (removing all the zaptel related lines), cleared out /lib/modules/version/misc, ran depmod -a, verified my /etc/sysconfig/zaptel configuration file and rebooted. Reboot? You should need no reboot to install/update zaptel modules. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with -c for color CLI?
On Tue, Jul 04, 2006 at 05:10:35PM +0200, Robert Michel wrote:
Salve *!
I'm using asterisk for a while and now I want to have a colord CLI.
I have apt-get install asterisk/testing, that is asterisk 1.2.7.1
I use Debian stable/testing on a vserver with any /dev/tty*.
So, of course, I comment out #TTY=9 inside /usr/sbin/safe_asterisk.
safe_asterisk has a flawed logic: it assumes that the tty device will
always exist. Thus it is not suited for use with screen.
However wouldn't it be better to tell asterisk to have colors even in a
remote terminal unless you use -n?
See attached patch for a possible route. I don't remember if I tested
it, though.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED] http://www.xorcom.com
#! /bin/sh /usr/share/dpatch/dpatch-run
## remote_color.dpatch by Tzafrir Cohen [EMAIL PROTECTED]
##
## All lines beginning with `## DP:' are a description of the patch.
## DP: Make Asterisk's terminal use colors by default. Doesn't work
@DPATCH@
diff -urNad asterisk-1.2.7.1.dfsg/term.c
/tmp/dpep.2czPCU/asterisk-1.2.7.1.dfsg/term.c
--- asterisk-1.2.7.1.dfsg/term.c2005-11-29 20:24:39.0 +0200
+++ /tmp/dpep.2czPCU/asterisk-1.2.7.1.dfsg/term.c 2006-05-13
19:05:50.209354595 +0300
@@ -78,9 +78,11 @@
char buffer[512] = ;
int termfd = -1, parseokay = 0, i;
+ if (! option_nofork) /* if we daemonize, our terminal is irrelevant */
+ term = xterm;
if (!term)
return 0;
- if (!option_console || option_nocolor || !option_nofork)
+ if (option_nocolor)
return 0;
for (i=0 ;; i++) {
___
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[Asterisk-Users] I am looking for a (graphical) statistic program
I am looking for a graphical statistic program. What I want to see is: a. my bandwidth (MRTG I use now from my upstream, but the time seems to be 20 minutes wrong,...) b. how many phone calls are at the same time (to get the feeling how much bandwidth how many phone calls are using) c. how long phone calls are, separated to different criteria, like prefix number, duration. most of these is in the program from areski, with the exeption that the numbers are wrong, like graphic shows 5 phone call and load shows 4 calls, . What are you using? bye Ronald Wiplinger ___ --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] voip-magazine article Using DUNDi with a Cluster of Asterisk Servers
Hi JR,I also noticed this article and thought why not! let's try this. After I followed the document I still wasn't able to do dundi lookups. This is what I get when I try to do a lookup:Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: DPDISCOVER (Command) Flags: 00 STrans: 31739 DTrans: 0 [192.168.1.13:4520] VERSION : 1 DIRECT EID : 00:0c:29:30:ed:19 CALLED NUMBER : 1601 CALLED CONTEXT : priv TTL : 5Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: DPRESPONSE (Response) Flags: 00 STrans: 04267 DTrans: 31739 [192.168.1.13:4520] (Final) CAUSE : NOAUTH: Unencrypted responses not permittedRx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response) Flags: 00 STrans: 31739 DTrans: 04267 [ 192.168.1.13:4520] (Final)So this doesn't look very good... I'm using trixbox 1.1. Do you have an idea what it could be? I couldn't find anything on the net so that's why I'm mailing you directly. Thanks in advance.Best regardsTijmen van den BrinkA University student from the NetherlandsOn 6/21/06, JR Richardson [EMAIL PROTECTED] wrote:Hi All,Check out the article Using DUNDi with a Cluster of Asterisk Servers at voip-magazine.comThat Mark Spencer sure can crank out the words.Hope you all get asmuch out of it as I did.Good job Mark!JR--JR Richardson Engineering for the Masses___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-magazine article Using DUNDi with aCluster of Asterisk Servers
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --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] voip-magazine article Using DUNDi with aClusterof Asterisk Servers
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --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] voip-magazine article Using DUNDi withaClusterof Asterisk Servers
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --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] voip-magazine article Using DUNDiwithaClusterof Asterisk Servers
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --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] voip-magazine article Using DUNDiwithaClusterofAsterisk Servers
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --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] voip-magazine article UsingDUNDiwithaClusterofAsterisk Servers
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --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] voip-magazine articleUsingDUNDiwithaClusterofAsterisk Servers
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --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
