Re: [asterisk-users] DTMF problem with 1.4.1
it shows empty string On 4/3/07, Rizwan Hisham [EMAIL PROTECTED] wrote: You can use the following to display what you receive from user (dtmf): exten= 1,1,Read(test) exten= 1,2,NoOp(DTMF Received: $test) exten= 1,3,Hangup On 4/3/07, Nitin Gupta [EMAIL PROTECTED] wrote: I upgraded to 1.4.1 and my DTMF has stopped working, I tried rfc2833compensate=yes and relaxdtmf=yes etc but none working. Everything seems to work fine with 1.2.10 Is there any way I dump the dtmf data packets received by asterisk on console? Any idea or pointers to debug the issue will be much appreciated. thanks, Nitin ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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] DTMF problem with 1.4.1
I upgraded to 1.4.1 and my DTMF has stopped working, I tried rfc2833compensate=yes and relaxdtmf=yes etc but none working. Everything seems to work fine with 1.2.10 Is there any way I dump the dtmf data packets received by asterisk on console? Any idea or pointers to debug the issue will be much appreciated. thanks, Nitin ___ --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] detecting the receivers voicemail
Hi, Is there any way asterisk can detect if the outgoing call is being received by a user or it has been forwarded to his voicemail. Actuallymy requirement is to proceed only if user picks up the phone otherwiseto hangup as soon as the call goes to voicemail. Is there anyway to do this? Thanks in advance. Nitin ___ --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] Prompts recording for Asterisk
thanks Dovid, infact I just got things recorded from her. Nitin On 8/27/06, Dovid Bender [EMAIL PROTECTED] wrote: snip 2) What are the best sources (cost effective) to get prompts recorded. /snip I would go with allison. She is the one that did all the voice files that you currently have on asterisk. So if you use her for your prompts you will have the same voice thru out ur PBX. A client of mine just used her for his entire pbx (total of 12 clips i believe ranging in sizes). The price was $75.00 ___ --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] Prompts recording for Asterisk
Hi, this question may sound a little dumb, but I need opinion of who are already using asterisk. questions are : 1) which format is best suited for asterisk (.gsm, .wav etc, also what sampling rate and bit size) 2) What are the best sources (cost effective) to get prompts recorded. thanks in advance. Nitin ___ --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] loading the prompt files in memory on Asteriskstartup
thanks a lot David, its really useful after googling I found one more link on ramfs http://www.linuxfocus.org/English/July2001/article210.shtml thought this can be useful for others. Nitin On 8/18/06, David Gagnon [EMAIL PROTECTED] wrote: Hi, Take a look at ramfs (http://plume.bxlug.be/articles/7 ). All youneed then is to create a link (ln -s) in /var/lib/asterisk/sound to thenramdrive you created using ramfs.David-Message d'origine-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] De la part de Nitin GuptaEnvoyé: 18 août 2006 22:08À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] loading the prompt files in memory onAsteriskstartupHi,Is there any option in asterisk to load all the prompt files intomemory on startup, so that it doesn;t have to hit the disk to read prompts for any call.Or any plugin / suggestion to avoid hitting the disk for prompt files?Thanks in advance.Nitin___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] loading the prompt files in memory on Asterisk startup
Hi, Is there any option in asterisk to load all the prompt files into memory on startup, so that it doesn;t have to hit the disk to read prompts for any call. Or any plugin / suggestion to avoid hitting the disk for prompt files? Thanks in advance. Nitin ___ --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] Asterisk load testing
Hi, did anyone try do load-testing on asterisk,for sip channel calls? I want to have a rough estimate about - how many calls, an asterisk server, running on say dual 240 opteron with 1 GB memory, can handle? Also how much internet bandwidth does a typical call requires? I heard around 20Kbps with typical codecs, is that right? Thanks in advance, Nitin ___ --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] Automation of call initiation
Hi, I need to monitoran asterisk server, so planning to use some tools which can initiate call to a number (for asterisk server)periodically and can interpret the response, is anything as such already available?? or any pointer?? thanks in advance Nitin ___ --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 detect call forwarding to voicemail
Hi, Is there anyway in Asterisk to know that outgoing call has been forwarded to voicemail by the callee system? Someof my users don't want to connect the call if its forwarded to callee voicemail, so I am wondering if theres anyway to identify this in Asterisk and drop the call. Thanks Nitin ___ --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 provider
Thanks for the information, I will surely look into it! Nitin On 5/10/06, Kerry Garrison [EMAIL PROTECTED] wrote: Have you looked at CBeyond? I like their T1 SIPConnect product. From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Nitin GuptaSent: Wednesday, May 10, 2006 7:04 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] VOIP provider Hi, I am looking for voip providers in bay area, any suggestions? My requirements are: handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP protocol support from VOIP provider. Also a dedicated t1 line in case provider can provide this too. Thanks, Nitin ___--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] VOIP provider
Hi, I am looking for voip providers in bay area, any suggestions? My requirements are: handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP protocol support from VOIP provider. Also a dedicated t1 line in case provider can provide this too. Thanks, Nitin ___ --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] Dial command to connect two channels and bypass asterisk server
thanks for the information Peter, its really helpful. Also Ihave one more question - do you have any idea how many such simultaneous calls can an asterisk server handle (say running od 2.6Ghz, 1GB Ram, fedora machine)? Thanks, Nitin On 2/13/06, Peter Fern [EMAIL PROTECTED] wrote: You can enable this on a per-peer basis with:sip peers:canreinvite=yesiax peers:notransfer=no Check the iax.conf.sample and sip.conf.sample files for usage.Nitin Gupta wrote: Hi I was wondering if its possible to make Dial command bridge two channels and after bridging bypass asterisk, so that the voice doesn't need to pass through my asterisk server. For e.g., I have a user dialed in and he verifies himself and then dials an international extension, after the call connects I don't want the call to pass through asterisk server anymore. Is there any command already there for any particular channel type? Thanks, Nitin___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Dial command to connect two channels and bypass asterisk server
Hi I was wondering if its possible to make Dial commandbridge two channels and after bridgingbypass asterisk, so that the voice doesn't need to pass through my asterisk server. For e.g., I have a user dialed in and he verifies himself and then dials an international extension, after the call connects I don't want the call to pass through asterisk server anymore. Is there any command already there for any particular channel type? Thanks, Nitin ___ --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] Problem with Playback sound in 64 bit machine
Sorry for re-posting this message - Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentence which should finish in 4 secs finishes in much lesser time. Where can be the problem? any configuration issue? Thanks in advance. ___ --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] Problem with Playback sound in 64 bit machine
sorry can you elaborate a little, what exactly is timing issue? Thanks On 2/12/06, Martin Joseph [EMAIL PROTECTED] wrote: On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote: Sorry for re-posting this message - Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentence which should finish in 4 secs finishes in much lesser time. Where can be the problem? any configuration issue? A timing issue with your linux distro?Whatever that is___--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] Problem with Playback sound in 64 bit machine
well if I pass the parameter aswhennext/4 instead of whennext/8 in file.c = ast_readaudio_callback()= ast_sched_add(.,whennext/4,...) things start working fine. I debugged the sched.c and time.c didn't find why this should happen. Since the scheduler calculates time-interval and keeps schedule queue item wrt timeval struct, sochanging the machineclock frequencyas long asgettimeofday() returns the same value should not matter. Icompared the whennext valuein my new machine with the one in old machine where asterisk as it is works fine - the values in both machine are same. Anyone willing to debug things further?? Nitin On 2/12/06, Nitin Gupta [EMAIL PROTECTED] wrote: sorry can you elaborate a little, what exactly is timing issue? Thanks On 2/12/06, Martin Joseph [EMAIL PROTECTED] wrote: On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote: Sorry for re-posting this message - Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentence which should finish in 4 secs finishes in much lesser time. Where can be the problem? any configuration issue? A timing issue with your linux distro?Whatever that is ___--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] Problem with Playback sound in 64 bit machine
got the answer the gettimeofday() is twice as fast as the one in older box, problem with system clock. Nitin On 2/12/06, Nitin Gupta [EMAIL PROTECTED] wrote: well if I pass the parameter aswhennext/4 instead of whennext/8 in file.c = ast_readaudio_callback()= ast_sched_add(.,whennext/4,...) things start working fine. I debugged the sched.c and time.c didn't find why this should happen. Since the scheduler calculates time-interval and keeps schedule queue item wrt timeval struct, sochanging the machineclock frequencyas long asgettimeofday() returns the same value should not matter. Icompared the whennext valuein my new machine with the one in old machine where asterisk as it is works fine - the values in both machine are same. Anyone willing to debug things further?? Nitin On 2/12/06, Nitin Gupta [EMAIL PROTECTED] wrote: sorry can you elaborate a little, what exactly is timing issue? Thanks On 2/12/06, Martin Joseph [EMAIL PROTECTED] wrote: On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote: Sorry for re-posting this message - Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentence which should finish in 4 secs finishes in much lesser time. Where can be the problem? any configuration issue? A timing issue with your linux distro?Whatever that is ___--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] Problem with Playback sound in 64 bit machine
The problem is with my new machine's hardware clock, which was running twice as fast as normal(bios bug). After searching through google the fix for it I could find is to disable apic in bios. This is my machine configuration: HP a1230nATI XPress chipsetAMD Athlon 64 X2 3700+ATI X800 XLLinux kernel 2.6.11-1.1369_FC4 basically added noapic acpi=noirq in grub kernel option i.e in /etc/grub.etc append noapic acpi=noirq for kernel kernel /boot/vmlinuz-2.6.15.1 ro root=LABAL=/ rhgb quiet noapic acpi=noirq After reboot of machine clock was normal. Nitin On 2/12/06, Tamas [EMAIL PROTECTED] wrote: Hi,how did you come to that? :)How did you fix it?Regards, TamasNitin Gupta wrote: got the answer the gettimeofday() is twice as fast as the one in older box, problem with system clock. Nitin On 2/12/06, *Nitin Gupta* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: well if I pass the parameter as whennext/4 instead of whennext/8 in file.c = ast_readaudio_callback() = ast_sched_add( .,whennext/4,...) things start working fine. I debugged the sched.c and time.c didn't find why this should happen. Since the scheduler calculates time-interval and keeps schedule queue item wrt timeval struct, so changing the machine clock frequency as long as gettimeofday() returns the same value should not matter. I compared the whennext value in my new machine with the one in old machine where asterisk as it is works fine - the values in both machine are same. Anyone willing to debug things further?? Nitin On 2/12/06, *Nitin Gupta* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: sorry can you elaborate a little, what exactly is timing issue? Thanks On 2/12/06, *Martin Joseph* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Feb 12, 2006, at 1:05 AM, Nitin Gupta wrote: Sorry for re-posting this message - Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentence which should finish in 4 secs finishes in much lesser time. Where can be the problem? any configuration issue? A timing issue with your linux distro?Whatever that is ___ --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] bad sound frequency
Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentece which shoudl finish in 4 secs finishes in much lesser time. Where can be the problem? and configuration issue? Thanks, Nitin ___ --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
