RE: [Asterisk-Users] SER Config For Asterisk
what i want is be able to authenticate user before they connected to my asterisk box. users can be registered with asterisk, but i want that each time a user want to place outgoing call, he is first authenticate, and then authorize to place the call through the asterisk box. this is for billing meaning. Thanks. From: [EMAIL PROTECTED] on behalf of Arnd Vehling Sent: Thu 26/05/2005 14:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SER Config For Asterisk Daniel Eboa wrote: > This is the scenario i want to setup: > > Cisco ATA 186 ---> SER -> Asterisk > > I want the Cisco ATA to register to Asterisk through SER. when the Cisco > ATA place a call, SER querry a data base (MySQL or else), and if there > is a valid Account for the ATA, the call go to Asterisk. > Did someone know how to set SER to work like this with Asterisk? > which version of SER should I use? > I've try both ser-0.8.11 and ser-0.9.0 but seems like something is > missing. I can't find some modules in ser.cfg file like: auth_radius.so > and others. > Can somebody help me with this issue?? Can u explain this in more detail please. Please make clear what u want 2 achieve. What u mean by "and if there is a valid Account for the ATA"? The ATA should be required to register and proxy-auth. If it can do this it must be either in the database or configured on a radius server. If it passes the auth u can just forward all calls to a asterisk box. cheers, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER Config For Asterisk
Hello, This is the scenario i want to setup: Cisco ATA 186 ---> SER -> Asterisk I want the Cisco ATA to register to Asterisk through SER. when the Cisco ATA place a call, SER querry a data base (MySQL or else), and if there is a valid Account for the ATA, the call go to Asterisk. Did someone know how to set SER to work like this with Asterisk? which version of SER should I use? I've try both ser-0.8.11 and ser-0.9.0 but seems like something is missing. I can't find some modules in ser.cfg file like: auth_radius.so and others. Can somebody help me with this issue?? Thanks Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER Config for Asterisk
Hello, This is the scenario i want to setup: Cisco ATA 186 ---> SER -> Asterisk I want the Cisco ATA to register to Asterisk through SER. when the Cisco ATA place a call, SER querry a data base (MySQL or else), and if there is a valid Account for the ATA, the call go to Asterisk. Did someone know how to set SER to work like this with Asterisk? which version of SER should I use? I've try both ser-0.8.11 and ser-0.9.0 but seems like something is missing. I can't find some modules in ser.cfg file like: auth_radius.so and others. Can somebody help me with this issue?? Thanks Daniel. <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Project Consultant/Parner Wanted
Dear sir, I'm interested in your project. Can you tell more about it?? Regards. Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, May 22, 2005 4:23 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Asterisk Project Consultant/Parner Wanted Hello All, How are you all doing today? Good I hope. I am sure that I have asked this question before, but recently lost my emails server and thus any replies that you may have sent me. We are working to get a small online VoIP service established and I am looking for someone who might like to partner on this project or possibly offer reasonable consulting services. We need someone to take the lead on the development of the Asterisk PBX server and site configuration to get the service set up and operating. Please send an email if interested. Have a good day, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MGCP issue
What exactly should I need to change in indications.conf?? Thanks. Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of wells zheng Sent: Wednesday, May 04, 2005 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MGCP issue you need make change in indications.conf wells On 3/25/05, Daniel Eboa <[EMAIL PROTECTED]> wrote: > > Hello List, > > I'm trying to setup MGCP channel with a Centile Media Hub box. My > Centile box has 4 ports and I got no dial tone. Can somebody help with > this isuue? > This is my mgcp.conf and extensions.conf > > Thanks > Daniel. > > ; MGCP Configuration for Asterisk > ; > [general] > port = 2427 > bindaddr = 192.168.11.20 > disallow=all > allow=g729 > allow=alaw > allow=ulaw > > [192.168.11.200] > context=MGCP > host=192.168.11.200 > wcardep=aaln/* > callerid = "test" <8000100> > callwaiting=no > transfer=no > cancallforward=no > dtmfmode=rfc2833 > canreinvite=no > singlepath=no > slowsequence=yes > line => aaln/1 > callerid= "test" <8000101> > callwaiting=no > transfer=no > cancallforward=no > canreinvite=yes > dtmfmode=rfc2833 > line => aaln/2 > callerid= "test" <8000102> > callwaiting=no > transfer=no > cancallforward=no > canreinvite=yes > dtmfmode=rfc2833 > line => aaln/3 > callerid= "test" <8000104> > callwaiting=no > transfer=no > cancallforward=no > canreinvite=yes > dtmfmode=rfc2833 > line => aaln/4 > > extensions.conf > > [MGCP] > include => Toll Free > include => CreoLink > exten => 8000100,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) > exten => 8000100,2,Hangup > exten => 8000101,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) > exten => 8000101,2,Hangup > exten => 8000102,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) > exten => 8000102,2,Hangup > exten => 8000103,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) > exten => 8000103,2,Hangup > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users]Unable to register license for G729 codec
Contact Digium For this issue -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohammed Firdosh Nasim Sent: mercredi 13 avril 2005 16:12 To: asterisk-users@lists.digium.com Cc: Mohammed Firdosh Nasim Subject: [Asterisk-Users]Unable to register license for G729 codec Hi, I bought the license for codec g.729a from digium and am now facing some problem registering the codec with them. i got the following message. -- ./register G729-<**key**> Digium Product Registration Copyright (C) 2004, Digium, Inc. Analyzing key 'G729-<**key**>' Connecting to Digium License Server (216.207.245.3:5646)...FAILED(2)! - Kindly give ur valuable suggestion. Thanks, Firdosh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 Question
Hello list, I have a question about Asterisk and H323. Wich H323 channel driver is the best for Asterisk? Asterisk-oh323 or OH323. I’m asking this question because I have big problem running my asterisk with asterisk-oh323. all is well installed but when there are some calls, my asterisk stop running. Right nowm I’m using asterisk-v1.0.2 LSE RPM distro with all the modules ( asterisk-addons, asterisk-oh323, asterisk-zaptel, asterisk-libpri). All these modules are RPMs but I still have the same problem. I’ve first used version 1.0.RC2 of asterisk and corresponding modules. Can some body help me with this issue. Regards. Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP issue
Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP host=192.168.11.200 wcardep=aaln/* callerid = "test" <8000100> callwaiting=no transfer=no cancallforward=no dtmfmode=rfc2833 canreinvite=no singlepath=no slowsequence=yes line => aaln/1 callerid= "test" <8000101> callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line => aaln/2 callerid= "test" <8000102> callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line => aaln/3 callerid= "test" <8000104> callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line => aaln/4 extensions.conf [MGCP] include => Toll Free include => CreoLink exten => 8000100,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten => 8000100,2,Hangup exten => 8000101,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten => 8000101,2,Hangup exten => 8000102,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten => 8000102,2,Hangup exten => 8000103,1,Dial(MGCP/aaln/[EMAIL PROTECTED],30,rt) exten => 8000103,2,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pattern matching in extensions.conf
What is 00 and other numbers? Are different destinations prefix ?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mickey Binder Sent: vendredi 18 mars 2005 12:39 To: Asterisk maillist (asterisk-users@lists.digium.com) Subject: [Asterisk-Users] Pattern matching in extensions.conf Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ## 00 ## 20 ## 30 ## 40 ## 15 ## 35 ## 12 ## 44 Right now I've solved it by doing this: exten => _##[0234]0,1,HangUp exten => _##[13]5,1,HangUp exten => ##12,1,HangUp exten => ##44,1,HangUp The ## symbolises a fixed number, it's just censored away (and not important anyway) I was just wondering if there was a more intelligent approach, so this could be combined into one extension. Best regards, Mickey Binder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home 0.5
Hello list, Just wonder if [EMAIL PROTECTED] can work with asterisk-oh323 0.6. Did any one try it ?? Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.
I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: samedi 12 février 2005 06:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. 1. There is a help file you can run from the Linux command line help-aah. This will tell you how to change the passwords. On a clean install it tells you this in the motd. 2. Not sure about this second one. I made some big changes in asterisk for this release. It now runs as asterisk not as root and it uses amportal to start not the startup files in /etc/init.d I think only a clean install will fix this. 3. A lot of changes in FOP too the config files are in a different place could cause this problem. Sorry about all the changes. As we get closer to a 1.0 release of [EMAIL PROTECTED] a lot of this will stabilize. --- Ariel Batista <[EMAIL PROTECTED]> wrote: > Hello, > > Great job on the [EMAIL PROTECTED] project. Looks great > this new version is really nicer looking. But I > have a few questions. > > 1) For the new web access http://localIP/maint how > and where do I change the password. > 2) Since I don't use the Amp section for setup the > .conf files I use my own. How do I get the asterisk > server running status up. I have it running and > works but shows up as not running on the web page. > 3) I upgraded my system from the older .04 by > downloading the new tar and running your script. > Then I copied my .conf files back and rebooted. I > had already changed my password and logins names > before this. Asterisk is up and running without any > issue's. But the Flash Operator panel comes up > flashing and I can't seem to get it to work. > > I feel you have done a great job and I would like to > thank you for your setup to us. I will be sending > you a donation soon. I am at a small self employed > computer consultant that has limited funds at > present. This is one of the best setups for > Asterisk that I have seen. I feel your name does not > do it right due to it can be used for SOHO's and > other setups. It's great keep up the good work. You > actually make AMP work. > > P.S. one more question do you have an area in the > freenode for chat? If you don't I would love to help > out in it. Something like Asterisk-athome would be > nice. > > Ariel > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 codec for X-lite soft phone
Sir, I think when somebody asked a question, is because he doesn’t know the answer. Even maybe when for some people like you, the answer is evidence. Thinking that I know the answer of the question I asked, suppose that I’m stupid, while I’m not. I you feel offence by the question I asked, please simply ignore it. Regards. Daniel. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: mercredi 9 février 2005 18:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] G.729 codec for X-lite soft phone Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. /Snip/ Daniel, You know that X-lite does not support G.729 and you also know where to have it, dont you? if you read your questions a couple of times, you will find answers there. Also, if you ever Visit Xten site and look at the information there, you will know what is, and what is not , supported in X-Lite and X-Pro. Sending questions like these to a busy forum like Asterisk only make it ever more difficult for people who are trying to wade through the thousands of emails posted here, for useful information. Please be considerate Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 codec for X-lite soft phone
Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based Asterisk management tool
How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk Compile Problem on Red Hat 9
I have both Stable version (asterisk-1.0-RC2) and the CVS version (asterisk v1-0-5) running on different Red Hat 9 boxes and there is no problem. I have only problem when I installed the oh323 driver (asterisk-oh323). Make sure you install Red Hat with required Package to run Asterisk. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: mercredi 9 février 2005 15:37 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk Compile Problem on Red Hat 9 > I get the following error when trying to compile asterisk 1.05 on red > hat 9. Is this the tarball available for download from the asterisk website? You might try CVS instead - try the CVS HEAD release: # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout zaptel libpri asterisk Or, if that doesn't do it, you can try CVS Stable # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds When you compile, make sure you do a "make clean", first, then "make install" If neither of these works, I might suggest trying a different OS - Red Hat 9 is no longer updated. If you're attached to red hat, you might try Tao Linux or Whitebox Linux - both are essentially Red Hat Enterprise, but they are free and both provide updates (security and otherwise). > -DASTMODDIR=\"/usr/lib/asterisk/modules\" > -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" > >-DBUSYDETECT_MARTIN `ls *.c` > > : invalid option > > Usage: /bin/sh [GNU long option] [option] ... > *** You don't have mpg123 installed. You're going to need *** > > *** it if you want MusicOnHold *** BTW: You can get mpg123 here: http://www-ti.informatik.uni-tuebingen.de/~hippm/mpg123.html Download the 0.59r version. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to Create customized audio file to use withASTCC??
There are good soft that do this very well. You don't have to record anything. You will just have to type what you want and the soft will speak for you. You can find them here: http://www.google.com/search?hl=en&q=Text+to+Voice+Technology&btnG=Google+Search Now what I want is to convert them in gsm format. Thanks. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: lundi 7 février 2005 12:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to Create customized audio file to use withASTCC?? Hi Derek, Yes there is. Take a look at my web pages http://www.g7ltt.com/VoIP/vmfiles.html. You'll see that I started a project to record as many different regional accents (with local lingo) as I could. It started well but as I had to rely on others to create the files (I don't speak with a Welsh accent) it fell by the wayside. If you'd like to voice your files and also know a woman whom would be willing to do the same (female voices are more sought after) then I could do the editing for you. Take a look in your asterisk-sounds directory for the script to the files. I used a Radioshack microphone and recorded the file as a WAV with Audacity then chopped it up into the phrases. If anyone else is interested in getting this project going again please contact me off list. Mark Derek Conniffe wrote: > I did this - I'm in Ireland and needed sounds like "Euro" and "Hash" > rather than "Dollars" and "Pound". I typed up the script of what was > needed, recorded it a number of times on semi-professional equipment and > then I spent the time editing the recordings into the individual wav > files and then, finally, converted the sounds into gsm files. These > sounds are being used in a low cost call shop in Dublin now. I'm not > sure if my ASTCC recordings would suit your (or anyones) needs but if > you would like a copy I have no problem providing them publically for no > charge. > > Derek > > Daniel Eboa wrote: > >> Hello all, >> >> Can anyone help me out with this issue ?? I got ASTCC running, but the >> audios doesn't match my needs (currency, etc.). is there any way to >> create my own audios and replace the current one?? >> >> Thanks. >> >> Daniel. >> >> >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to Create customized audio file to use with ASTCC??
Same for me with the french file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edgar de Leon Sent: lundi 7 février 2005 12:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to Create customized audio file to use with ASTCC?? Hello Mark, i tried to get the spanish soun but get You don't have permission to access /VoIP/AsteriskSounds_ES.tar.gz on this server. can you help me??? TIA Edgar > Hi Derek, > > Yes there is. Take a look at my web pages > http://www.g7ltt.com/VoIP/vmfiles.html. You'll see that I started a > project to record as many different regional accents (with local lingo) > as I could. > > It started well but as I had to rely on others to create the files (I > don't speak with a Welsh accent) it fell by the wayside. > > If you'd like to voice your files and also know a woman whom would be > willing to do the same (female voices are more sought after) then I > could do the editing for you. > > Take a look in your asterisk-sounds directory for the script to the files. > > I used a Radioshack microphone and recorded the file as a WAV with > Audacity then chopped it up into the phrases. > > If anyone else is interested in getting this project going again please > contact me off list. > > Mark > > Derek Conniffe wrote: >> I did this - I'm in Ireland and needed sounds like "Euro" and "Hash" >> rather than "Dollars" and "Pound". I typed up the script of what was >> needed, recorded it a number of times on semi-professional equipment and >> then I spent the time editing the recordings into the individual wav >> files and then, finally, converted the sounds into gsm files. These >> sounds are being used in a low cost call shop in Dublin now. I'm not >> sure if my ASTCC recordings would suit your (or anyones) needs but if >> you would like a copy I have no problem providing them publically for no >> charge. >> >> Derek >> >> Daniel Eboa wrote: >> >>> Hello all, >>> >>> Can anyone help me out with this issue ?? I got ASTCC running, but the >>> audios doesn't match my needs (currency, etc.). is there any way to >>> create my own audios and replace the current one?? >>> >>> Thanks. >>> >>> Daniel. >>> >>> >>> >>> ___ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to Create customized audio file to use withASTCC??
Hi Derek, I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ?? Thanks. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Conniffe Sent: lundi 7 février 2005 11:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to Create customized audio file to use withASTCC?? I did this - I'm in Ireland and needed sounds like "Euro" and "Hash" rather than "Dollars" and "Pound". I typed up the script of what was needed, recorded it a number of times on semi-professional equipment and then I spent the time editing the recordings into the individual wav files and then, finally, converted the sounds into gsm files. These sounds are being used in a low cost call shop in Dublin now. I'm not sure if my ASTCC recordings would suit your (or anyones) needs but if you would like a copy I have no problem providing them publically for no charge. Derek Daniel Eboa wrote: > Hello all, > > Can anyone help me out with this issue ?? I got ASTCC running, but the > audios doesn't match my needs (currency, etc.). is there any way to > create my own audios and replace the current one?? > > Thanks. > > Daniel. > > > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which version of asterisk-oh323 should i use with asterisk v1-0-5.
Hi list, I have successfully upgrade my Asterisk V1-0-RC2 to V1-0-5, but I have a problem. The Asterisk box crashes now every time. I’m using asterisk-oh323. is there a stable version of asterisk-oh323 that can work with the v1-0-5 of Asterisk. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC Apllication
Thanks a lot. Now I understand and it's working. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl H. Putz Sent: vendredi 4 février 2005 15:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ASTCC Apllication -Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa >Sent: Friday, February 04, 2005 4:50 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [Asterisk-Users] ASTCC Apllication > > >Hello, >I have some problem using ASTCC application. I've installed the application and everything works >well. I've created card numbers, routes trunk and others. When I dial the desired number (77) in >my case, I'm prompted to enter my card number. All goes well till I'm prompted to enter the >destination number. When I enter a destination number, the system says it's not a recognized >number and the call doesn't go through. Can any one help me out with this issue? Is there a file >where I can define extensions like in extensions.conf? Daniel, It sounds like the problem is the pattern you are trying to use in the "routes" table. The pattern should be a REGEX for matching the dialed number to the appropriate cost for that call. Take a look at http://dev.mysql.com/doc/mysql/en/pattern-matching.html for more specifics on MySQL REGEX matching. In the US, for example, I would use the pattern: '^1312' to match for calls to Chicago or '^01149' for calls to Germany. You can also match for city codes or especially Cellular "exchanges" in specific countries where the termination costs are much higher than land-line termination. The SQL statement in astcc returns all the matched patterns with the longest, most specific match first and uses only that first match in its processing. So you could also use the pattern: '.' to match any dialed number not already matched as a default BUT BE SURE to set that cost high enough to cover yourself. Good luck! Karl Putz > > >Thanks. > >Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to Create customized audio file to use with ASTCC??
Hello all, Can anyone help me out with this issue ?? I got ASTCC running, but the audios doesn’t match my needs (currency, etc.). is there any way to create my own audios and replace the current one?? Thanks. Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Apllication
Hello, I have some problem using ASTCC application. I’ve installed the application and everything works well. I’ve created card numbers, routes trunk and others. When I dial the desired number (77) in my case, I’m prompted to enter my card number. All goes well till I’m prompted to enter the destination number. When I enter a destination number, the system says it’s not a recognized number and the call doesn’t go through. Can any one help me out with this issue? Is there a file where I can define extensions like in extensions.conf? Thanks. Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to use ASTCC with SIP ??
I got this error when i try to dial:
-- Executing Answer("SIP/8000104-71a3", "") in new stack
-- Executing DeadAGI("SIP/8000104-71a3", "astcc.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File
astcc-tone does not exist in any format
Jan 29 18:11:37 WARNING[3412]: file.c:475 ast_openstream: File
astcc-accountnum does not exist in any format
Jan 29 18:11:37 WARNING[3412]: file.c:779 ast_streamfile: Unable to open
astcc-accountnum (format alaw): No such file or directory
== Spawn extension (prepaid, 77, 2) exited non-zero on
'SIP/8000104-71a3'
Can somebody tell me why and how to solve it ??
Regards.
Daniel.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wiebe
Sent: samedi 29 janvier 2005 18:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to use ASTCC with SIP ??
I would recommend using the local trunk and then you just need a context
that will dial out in your extensions.conf. Just put the context name
into the "Peer/Trunk" field on the trunks page. Currently there is not
support in astcc for oh-323. It would be trivial to add but
Darren Wiebe
[EMAIL PROTECTED]
Daniel Eboa wrote:
> Hello List,
>
> I've set up asterisk and install astcc application, everything was
> well installed, but i'm having problem using astcc with SIP. I don't
> have any Trunk card or any other analogic VoIP card connected to my
> asterisk box. I'm using SIP and asterisk-oh323 to connect to my VoIP
> provider. Does anyone knows how I can use astcc to work with my config
?
>
> Thanks.
>
>---
-
>___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] How to use ASTCC with SIP ??
Hello List, I’ve set up asterisk and install astcc application, everything was well installed, but i’m having problem using astcc with SIP. I don’t have any Trunk card or any other analogic VoIP card connected to my asterisk box. I’m using SIP and asterisk-oh323 to connect to my VoIP provider. Does anyone knows how I can use astcc to work with my config ? Thanks. <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error while trying to install astcc
Hello list, Here is the error i’m getting when i try to « make install » with astcc. Can somebody know this error and how to fix it? [EMAIL PROTECTED] astcc]# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 .) at ./astcc.agi line 47. BEGIN failed--compilation aborted at ./astcc.agi line 47. make: *** [install] Error 2 Regards. <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk
Hello I got the similar error while trying a call.
-- Executing Answer("SIP/8000104-86ef", "") in new stack
-- Executing Wait("SIP/8000104-86ef", "2") in new stack
-- Executing AGI("SIP/8000104-86ef", "areskicc.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php
areskicc.php: 'agi_request' => 'areskicc.php'
areskicc.php: 'agi_channel' => 'SIP/8000104-86ef'
areskicc.php: 'agi_language' => 'en'
areskicc.php: 'agi_type' => 'SIP'
areskicc.php: 'agi_uniqueid' => '1106824539.3'
areskicc.php: 'agi_callerid' => '"DTA-310" <8000104>'
areskicc.php: 'agi_dnid' => '002379511272'
areskicc.php: 'agi_rdnis' => 'unknown'
areskicc.php: 'agi_context' => 'prepaid'
areskicc.php: 'agi_extension' => '002379511272'
areskicc.php: 'agi_priority' => '3'
areskicc.php: 'agi_enhanced' => '0.0'
areskicc.php: 'agi_accountcode' => ''
areskicc.php:
areskicc.php: >> ANSWER
areskicc.php: string(56) ""DTA-310" <8000104> ; SIP/8000104-86ef ;
1106824539.3 ; "n
-- AGI Script areskicc.php completed, returning 0
-- Executing Wait("SIP/8000104-86ef", "2") in new stack
-- Executing Hangup("SIP/8000104-86ef", "") in new stack
== Spawn extension (prepaid, 002379511272, 5) exited non-zero on
'SIP/8000104-86ef'
Need some help.
Thanks
Daniel.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski
Sent: jeudi 27 janvier 2005 11:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard
ApplicationforAsterisk
Hi Alex,
Concerning the web interface, in this version we need the
register_globals = On
I will try to change it in the next release...
To find out the error on the agi,
can you run the agi script manually.
php areskicc.php
You will get more details about the error!
Regards,
Areski
On Thu, 2005-01-27 at 03:07, Alexander Romanov wrote:
> Hi,
>
> I've tried it and could not get to work any of them (webapp and agi).
>
> On webapp I do not get a full menu, just "logout" and "disconnect"
> With agi nothing happens when I execute the script.
>
> -- Executing Answer("SIP/2204-6221", "") in new stack
> -- Executing Wait("SIP/2204-6221", "2") in new stack
> -- Executing AGI("SIP/2204-6221", "areskicc.php") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php
> -- AGI Script areskicc.php completed, returning 0
> -- Executing Wait("SIP/2204-6221", "2") in new stack
> -- Executing Hangup("SIP/2204-6221", "") in new stack
> == Spawn extension (local, 40, 5) exited non-zero on 'SIP/2204-6221'
>
>
> I have followed instructions to the letter. Am I missing something?
>
> Alex.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Areski
> Sent: Thursday, 27 January 2005 4:05 AM
> To: Asterisk-Users Mailing-list
> Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application
> forAsterisk
>
>
> Hello everyone,
>
>
> If you want to know why I am so tired today :D
> Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just
> finish it yesterday night!
>
>
> Briefly, AreskiCC is an AGI script and PHP-Web application which greatly
> handle the complete CallingCard System.
>
>
> FEATURES - AGI :
> * Authenticate with the use of a Cardnumber
> the Cardnumber can also be defined as accountcode into sip.conf,
> iax.conf, etc..
> * take care of multiple calls using the same Cardnumber
> * Caller gets informed about his credit
> Announce the remaining credit
> * Caller is requested to enter a destination number
> * Announce the maximal call time for the given destination number
> It calculates the remaining duration of the actual call (based
> on tariffrate tables), informs the caller about this and sets a
> timeout
> * Interupt the call if the card balance gets zero
> Warn the caller about the call interupt 60 & 30 seconds before
> the call gets interupted
> * It connects the Caller to the destination through the configured
> trunk
> note : different trunks can be configured and associated by
> prefix
> * After disconnecting the call AGI updates the credit and stores
> the concerning Call-Detail-Records with CallingPartyNumber,
> CalledPartyNumber, CallSetupTime, Duration, Charge and the
> remaining credit
>
>
> FEATURES - WEB INTERFACE:
> * CARD/CUSTOMERS
> * List customers
> * Refill customer
> * CARD/CUSTOMERS
> * List customers/cards
> * Refill customer/card
> * Create customer/card
> * Generate customers/cards
> * BILLING
> * View money situation
> * View Payment
> * Add new Payment
> * RATECARD
> * Li
[Asterisk-Users] Packet8 DTA310 SIP Image
Hello list, Did someone know where to find the SIP image for Packet8 DTA 310 box to work with Asterisk ?? Thanks. <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Restarting *
Title: Message Connect to the asterisk console and type : stop now Then run asterisk by typing: asterisk -vvvc From: Ferguson, Michael [mailto:[EMAIL PROTECTED] Sent: jeudi 2 décembre 2004 15:51 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Restarting * G'Day All What do I type at the command line to stop and start * on a RedHat ES3 box? Thanks <>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Process Stop After few hours
Done. Now that i have the asterisk-oh323-0.6.3b on my system, how to install the asterisk-oh323-0.6.4? Is any update possible, or I have to remove the old version first ? -Original Message- From: Michael Manousos [mailto:[EMAIL PROTECTED] Sent: mercredi 1 décembre 2004 18:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours Daniel Eboa wrote: > How to get it? Download it from here: http://www.inaccessnetworks.com/projects/asterisk-oh323/download > > > > > > -Original Message- > From: Michael Manousos [mailto:[EMAIL PROTECTED] > Sent: mercredi 1 décembre 2004 17:35 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours > > Daniel Eboa wrote: > >>I use asterisk-oh323-0.6.3b, pwlib-v1_6_6 and openh323-v1_13_5. >> >>This is the complete error: H245:818c6c0 PWLIB Assertion Fail: file >>transports.cxx, line 1637 > > > Go up to v0.6.4 version of asterisk-oh323 (I guess that you use > Asterisk CVS stable). > > >>Thanks. >> >>Daniel > > > > Michael > > > >> >> >>-Original Message- >>From: Michael Manousos [mailto:[EMAIL PROTECTED] >>Sent: mercredi 1 décembre 2004 16:47 >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours >> >>Daniel Eboa wrote: >> >> >>>Hello to all, >>> >>>I have a strange behavior of my asterisk box. I'm running asterisk with >>>asterisk-oh323 channel driver and everything works very well. >>>But after few hours, my asterisk stop running and I have to restart it >>>by typing "asterisk -vvvc". Most of the time I connect to my asterisk >>>with a remote host so I don't know exactly which error causes my box to >>>stop, but I found on the console this message: "Segmentation Fault". Did >>>any one has experience this problem?? what is the solution? >> >> >>What versions of Asterisk/asterisk-oh323 do you run? >>Please provide a backtrace of the core file dumped. >> >> >> >>>I use Cisco ATA 186 Boxes with my asterisk. >>> >>>Thank In advance. >>> >>>Daniel. >>> >> >> >> >>Michael. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Process Stop After few hours
How to get it? -Original Message- From: Michael Manousos [mailto:[EMAIL PROTECTED] Sent: mercredi 1 décembre 2004 17:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours Daniel Eboa wrote: > I use asterisk-oh323-0.6.3b, pwlib-v1_6_6 and openh323-v1_13_5. > > This is the complete error: H245:818c6c0 PWLIB Assertion Fail: file > transports.cxx, line 1637 Go up to v0.6.4 version of asterisk-oh323 (I guess that you use Asterisk CVS stable). > > Thanks. > > Daniel Michael > > > > -Original Message- > From: Michael Manousos [mailto:[EMAIL PROTECTED] > Sent: mercredi 1 décembre 2004 16:47 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours > > Daniel Eboa wrote: > >>Hello to all, >> >>I have a strange behavior of my asterisk box. I'm running asterisk with >>asterisk-oh323 channel driver and everything works very well. >>But after few hours, my asterisk stop running and I have to restart it >>by typing "asterisk -vvvc". Most of the time I connect to my asterisk >>with a remote host so I don't know exactly which error causes my box to >>stop, but I found on the console this message: "Segmentation Fault". Did >>any one has experience this problem?? what is the solution? > > > What versions of Asterisk/asterisk-oh323 do you run? > Please provide a backtrace of the core file dumped. > > >>I use Cisco ATA 186 Boxes with my asterisk. >> >>Thank In advance. >> >>Daniel. >> > > > > Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Process Stop After few hours
I use asterisk-oh323-0.6.3b, pwlib-v1_6_6 and openh323-v1_13_5. This is the complete error: H245:818c6c0 PWLIB Assertion Fail: file transports.cxx, line 1637 Thanks. Daniel -Original Message- From: Michael Manousos [mailto:[EMAIL PROTECTED] Sent: mercredi 1 décembre 2004 16:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours Daniel Eboa wrote: > Hello to all, > > I have a strange behavior of my asterisk box. I'm running asterisk with > asterisk-oh323 channel driver and everything works very well. > But after few hours, my asterisk stop running and I have to restart it > by typing "asterisk -vvvc". Most of the time I connect to my asterisk > with a remote host so I don't know exactly which error causes my box to > stop, but I found on the console this message: "Segmentation Fault". Did > any one has experience this problem?? what is the solution? What versions of Asterisk/asterisk-oh323 do you run? Please provide a backtrace of the core file dumped. > > I use Cisco ATA 186 Boxes with my asterisk. > > Thank In advance. > > Daniel. > Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Process Stop After few hours
Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing "asterisk -vvvc". Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on the console this message: "Segmentation Fault". Did any one has experience this problem?? what is the solution? I use Cisco ATA 186 Boxes with my asterisk. Thank In advance. Daniel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rtp codec error
Hello all, I just register my asterisk with Digium g729 codec. But now when I place a call with my SIP phone through my Cisco ATA 186 box, I have this error: rtp.c:319 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256. Can some body tell me why?? Part of my SIP.conf: Disallow=all Allow=g729 Allow=alaw [5000500] type=friend callgroup=1 host=dynamic defaultip=xxx.xxx.xxx.xxx dtmfmode=rfc2833 context=sip-provider allow=g729 allow=alaw canreinvite=no callerid="John" <5000500> mailbox=5000500 pickupgroup=1 Thanks. <>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk-prepaid
Hello all, Did some body have this application working? If yes, please help with some examples of how to compile and prepaid.conf file. Thanks. <>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Strange error
This is another error for the same config below : Nov 12 19:02:44 WARNING[1170207680]: chan_sip.c:1838 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 4/8) From: Daniel Eboa Sent: vendredi 12 novembre 2004 18:16 To: 'Asterisk Mailing List ([EMAIL PROTECTED])' Subject: Strange error Hello all, I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b channel driver for H323. All installation and packages compilation was successful. I have a SIP account to a SIP provider and I use it for outgoing calls. I’m using Cisco ATA boxes both SIP and H323, and all the boxes connect to my Asterisk Server. I can call a SIP box from H323 and vis-versa. But when I want to dial out through my SIP account from my H323 box, the call goes through but I got this error: chan_oh323.c:3180 setup_h323_connection: Channel's format changed from 8 to 4??? Can some body help me out to find where is the problem ?? Thanks. <><>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange error
This is the version with whom i compile asterisk-oh323 channel successfully. I try the latest version of both asterisk and asterisk-oh323, but impossible to compile -Original Message- From: Paradise Dove [mailto:[EMAIL PROTECTED] Sent: vendredi 12 novembre 2004 18:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Strange error you're using out of date and buggy versions of * and oh323. try to update them and check if the error is occurring again. On Fri, 12 Nov 2004 18:16:23 +0100, Daniel Eboa <[EMAIL PROTECTED]> wrote: > > > > Hello all, > > > > I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b > channel driver for H323. All installation and packages compilation was > successful. I have a SIP account to a SIP provider and I use it for outgoing > calls. I'm using Cisco ATA boxes both SIP and H323, and all the boxes > connect to my Asterisk Server. I can call a SIP box from H323 and vis-versa. > But when I want to dial out through my SIP account from my H323 box, the > call goes through but I got this error: chan_oh323.c:3180 > setup_h323_connection: Channel's format changed from 8 to 4??? > > Can some body help me out to find where is the problem ?? > > > > Thanks. > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange error
Hello all, I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b channel driver for H323. All installation and packages compilation was successful. I have a SIP account to a SIP provider and I use it for outgoing calls. I’m using Cisco ATA boxes both SIP and H323, and all the boxes connect to my Asterisk Server. I can call a SIP box from H323 and vis-versa. But when I want to dial out through my SIP account from my H323 box, the call goes through but I got this error: chan_oh323.c:3180 setup_h323_connection: Channel's format changed from 8 to 4??? Can some body help me out to find where is the problem ?? Thanks. <>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with asterisk-oh323
Hello to all, I’m trying to get h323 working with Asterisk, I’ve downloaded all require modules (most are .tar.gz files, but if some body knows where to find working rpm file, it will help me), I’ve installed Pwlib, Openh323, and Asterisk. When I want to compile the asterisk-oh323 module, I got an error. The error has been reporting to the list, but I did not find the answer. I found another openh323 version (V. 1.13.5) and a patch to this version, but nothing work. Can somebody know how to make Asterisk Work with H323 ?? Which files are needed ?? Where to find Working files ?? I have both Fedora Core 1 and RedHat 9. Thanks. <>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compatibility of Asterisk With Cisco 5350/5400Gateway
Hello Emilio, I can't send the Cisco Gateway config since is handled by my SIP provider; I can send my sip.conf file. Can't send me the cisco Gateway config that match with my asterisk config ?? -Original Message- From: Emilio Panighetti [mailto:[EMAIL PROTECTED] Sent: samedi 16 octobre 2004 08:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Compatibility of Asterisk With Cisco 5350/5400Gateway Yes, it's compatible. If you have "signaling forward unconditional" on the dial-peer on on sip-ua, won't work (look at previous threads from this list). If that didn't work, maybe you can post your config for review. On Oct 16, 2004, at 3:03 AM, Daniel Eboa wrote: > Hi to all, > > I just wanna know if Asterisk works properly with Cisco 5350/5400 > Gateway. I register my Asterisk box on a Cisco 5350/5400 Gateway for > outgoing calls, and Asterisk is sending errors messages to the Cisco > Gateway, and shut it down after a short period. Is there any specific > configuration to make this work well ? I'm thinking at maybe a > protocol or codec issue. I can't tell with type of errors messages > since the gateway is a remote site of my SIP provider. If any one has > deal with this problem or know about it, I will appreciate his help. > > > > Thanks. > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compatibility of Asterisk With Cisco 5350/5400 Gateway
Hi to all, I just wanna know if Asterisk works properly with Cisco 5350/5400 Gateway. I register my Asterisk box on a Cisco 5350/5400 Gateway for outgoing calls, and Asterisk is sending errors messages to the Cisco Gateway, and shut it down after a short period. Is there any specific configuration to make this work well ? I’m thinking at maybe a protocol or codec issue. I can’t tell with type of errors messages since the gateway is a remote site of my SIP provider. If any one has deal with this problem or know about it, I will appreciate his help. Thanks. <>___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need Help !!
Hello to all,
I’m new user
of Asterisk. I’m running Asterisk on a RedHat 9 platform. Everything seems
to be ok but I got lot of error messages and I don’t know their meaning. Can
somebody help me ??
These are the
messages:
WARNING[163850]:
chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Non-critical
Request)
The server
displayed this message when the party I called picks up the phone. Immediately
after picking up the phone, the call drop.
-- Executing Dial("SIP/Daniel-7e41",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/195.149.11.98-86d9 is ringing
-- SIP/195.149.11.98-86d9 answered SIP/Daniel-7e41
-- Attempting native bridge of SIP/Daniel-7e41 and
SIP/195.149.11.98-86d9
-- Got SIP response 481 "Invalid CSeq
Number" back from 195.149.11.98
When a call
cannot be complete, this is what the server send as error.
-- Executing Dial("SIP/Daniel-3d52",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/195.149.11.98-a4a6 is making progress
passing it to SIP/Daniel-3d52
-- Got SIP response 500 "Internal Server
Error" back from 195.149.11.98
-- SIP/195.149.11.98-a4a6 is circuit-busy
== Everyone is busy/congested at this time
This error is
reporting also when the call cannot be completed. But the behave is different
with the previous error.
-- Executing Dial("SIP/Daniel-57d4",
"SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/195.149.11.98-fda1 is making progress
passing it to SIP/Daniel-57d4
-- Got SIP response 480 "Temporarily Not
Available" back from 195.149.11.98
-- SIP/195.149.11.98-fda1 is circuit-busy
== Everyone is busy/congested at this time
Thanks for your
help.
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