[asterisk-users] Voicemails do not email through asterisk
Hello All, I am running Asterisk 1.4.23.1 on debian lenny and having issues with it sending out voicemail emails. Let me preface with the following: 1. I have tested with sendmail and ssmtp (with valid smtp server) 2. Googled quite a bit to only find the above 3. The mail.log/err/info shows no requests even trying to mail out I have asterisk running as asterisk and have tested sendmail and ssmtp as the asterisk user. I am out of my ts levels ... has anyone else seen this? Thanks!!! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
On Wed, 6 May 2009, Vincent wrote: On Mon, 4 May 2009 10:07:06 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: Do you want to build your own? If so, you can put togther a 1GHz fanless VIA miniITX board, case (that will take a drive or flash IDE), memory and psu for well under £200. Same system has one PCI slot for a card that might take analogue, ISDN2 or ISDN30. Thanks for the tip. I'll see how much a complete VIA-based system costs. One little tip: You need to compile Asterisk for an i586 processor as the VIA processor is missing a few (mmx, etc.) instructions that a full blown i686 has. Gordon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemails do not email through asterisk
On 6 May 2009, at 08:16, Damon Brown wrote: Hello All, I am running Asterisk 1.4.23.1 on debian lenny and having issues with it sending out voicemail emails. Let me preface with the following: 1. I have tested with sendmail and ssmtp (with valid smtp server) 2. Googled quite a bit to only find the above 3. The mail.log/err/info shows no requests even trying to mail out I have asterisk running as asterisk and have tested sendmail and ssmtp as the asterisk user. I am out of my ts levels ... has anyone else seen this? Could you perhaps show us your config so we can actually look for problems? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
On 06/05/09 08:28, Gordon Henderson wrote: snip / One little tip: You need to compile Asterisk for an i586 processor as the VIA processor is missing a few (mmx, etc.) instructions that a full blown i686 has. Hi Gordon, I'm using a VIA C7 on a Jetway board (http://linitx.com/viewproduct.php?prodid=11212). This is my cpuinfo. Isn't that an i686 class? cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1200MHz stepping: 9 cpu MHz : 1099.969 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm nx up pni est tm2 rng rng_en ace ace_en ace2 ace2_en phe phe_en pmm pmm_en bogomips: 2199.93 clflush size: 64 power management: Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] After transfer context
Hello, in our dialplan we have some variables containing datas from our customers calls like for instance the called number. Now, when caller make a transfer, we would like to catch the new called number. How to get this? Is there a return context _after_ transfer and *before* the call is placed in the user context? A solution would be to test the BLINDTRANSFER variable but this should be done in *each* user context, that's why we are looking for a mmore global solution, if any. Thanks for any hint. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
On Wed, 6 May 2009, Alan Lord (News) wrote: On 06/05/09 08:28, Gordon Henderson wrote: snip / One little tip: You need to compile Asterisk for an i586 processor as the VIA processor is missing a few (mmx, etc.) instructions that a full blown i686 has. Hi Gordon, I'm using a VIA C7 on a Jetway board (http://linitx.com/viewproduct.php?prodid=11212). This is my cpuinfo. Isn't that an i686 class? Hm. You know what - it's possible my information is a little out of date now.. (quite possibly the wiki too!) I use these processors too, but my test boxes (~6 years old) have the VIA C3 chips in them: processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 7 model name : VIA Samuel 2 stepping: 3 cpu MHz : 533.376 cache size : 64 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr cx8 mtrr pge mmx 3dnow bogomips: 1067.55 and I know they definately segfault if I don't compile asterisk for an i586... cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family: 6 model : 10 model name: VIA Esther processor 1200MHz stepping : 9 cpu MHz : 1099.969 cache size: 128 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp: yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm nx up pni est tm2 rng rng_en ace ace_en ace2 ace2_en phe phe_en pmm pmm_en bogomips : 2199.93 clflush size : 64 power management: All my production ones have the newer processors, maybe it's time to abandon my old test boxes... (Although one old box is actually a live system with a customer!) Nice to know they last though - one thing I always wory about is longevity - You put a panasonic on a wall and it's still there and working 15 years later... Are these going to last 5 years, 10 or more? Time will tell, I guess! Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
Those reading the thread amy be interested in Askozia pbx http://www.askozia.com/pbx/ Michael was at AMOOCON (great success by the way, thanks to all who participated) and I was impressed. He will be a guest on VUC very soon, possibly even this Friday. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
On Wed, May 06, 2009 at 08:28:54AM +0100, Gordon Henderson wrote: On Wed, 6 May 2009, Vincent wrote: On Mon, 4 May 2009 10:07:06 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: Do you want to build your own? If so, you can put togther a 1GHz fanless VIA miniITX board, case (that will take a drive or flash IDE), memory and psu for well under £200. Same system has one PCI slot for a card that might take analogue, ISDN2 or ISDN30. Thanks for the tip. I'll see how much a complete VIA-based system costs. One little tip: You need to compile Asterisk for an i586 processor as the VIA processor is missing a few (mmx, etc.) instructions that a full blown i686 has. Which VIA processor? I don't think this applies to the C7 . (And I'm not aware of any x86-compatible processor that is in real-life usage now that has no MMX. Well, maybe some 486-s are still used somewhere) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] precision of wait dialplan application
Hello !
In order to chase after a problem I implemented the following dialplan to have
an
answertime of exactly one minute:
exten = xxx,1,NoOp(Test wait)
exten = xxx,n,Answer
exten = xxx,n,NoOp(Current timestamp:
${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)})
exten = xxx,n,Wait(60)
exten = xxx,n,NoOp(Current timestamp:
${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)})
exten = xxx,n,Hangup
But it seems the Wait(60) lasts longer than 60 seconds:
-- Executing [...@from_meridian:1] NoOp(DAHDI/29-1, Test wait)
in new stack
-- Executing [...@from_meridian:2] Answer(DAHDI/29-1, ) in new
stack
-- Executing [...@from_meridian:3] NoOp(DAHDI/29-1, Current
timestamp: 20090506135813) in new stack
-- Executing [...@from_meridian:4] Wait(DAHDI/29-1, 60) in new
stack
-- Executing [...@from_meridian:5] NoOp(DAHDI/29-1, Current
timestamp: 20090506135915) in new stack
-- Executing [...@from_meridian:6] Hangup(DAHDI/29-1, ) in new
stack
What is wrong in this example ?
Regards
Hans
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[asterisk-users] Questions on X100P/X101P cards
Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the computer's CPU? 2. Are they all bad, no matter what chipset is used (Intel, Motoral, Ambient)? If not, which offer good enough quality to handle a single POTS line? 3. Why are they often bad quality? Because the driver itself is badly written? Because PC's don't have enough speed to handle the tasks using their own CPU (hard to believe, but I don't know)? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
On 06/05/09 13:43, Vincent wrote: Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the computer's CPU? 2. Are they all bad, no matter what chipset is used (Intel, Motoral, Ambient)? If not, which offer good enough quality to handle a single POTS line? 3. Why are they often bad quality? Because the driver itself is badly written? Because PC's don't have enough speed to handle the tasks using their own CPU (hard to believe, but I don't know)? Hi Vincent, I bought a cheap eBay X100p card over a year ago. When I first tried it was appalling. I couldn't get rid of the echo and noise no matter what. I then came across OSLEC (at the time a new Free Echo Canceller). A bit of hacking to get it to work and hey-presto! No more echo. I have been using the same card ever since with no noticeable issues. I think OSLEC is now the default EC for many distributions so I would have thought you will be fine although, of course, YMMV. For a cheap backup to your VOIP service they do the job. I wouldn't use them for a proper system though. HTH Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
On Wed, 06 May 2009 14:02:20 +0100, Alan Lord (News) alansli...@gmail.com wrote: For a cheap backup to your VOIP service they do the job. I wouldn't use them for a proper system though. Thanks for the feedback. I have two more questions: 1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel Atom not doing much more than this and running Asterisk? 2. Is it good enough to handle a single FXO line for professional use? 3. Can you give me a pointer about which X100p you bought on eBay? AFAIK, there are three chipsets : Intel, Motorola, and Ambient. Using a $15 card over an $80 card is not insignificant because I could then sell a small server for $99. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
Gordon Henderson wrote: On Wed, 6 May 2009, Alan Lord (News) wrote: On 06/05/09 08:28, Gordon Henderson wrote: snip / One little tip: You need to compile Asterisk for an i586 processor as the VIA processor is missing a few (mmx, etc.) instructions that a full blown i686 has. Hi Gordon, I'm using a VIA C7 on a Jetway board (http://linitx.com/viewproduct.php?prodid=11212). This is my cpuinfo. Isn't that an i686 class? Hm. You know what - it's possible my information is a little out of date now.. (quite possibly the wiki too!) I use these processors too, but my test boxes (~6 years old) have the VIA C3 chips in them: Older C3's lacked some of the features, but I believe they all had MMX. Early ones lacked SSE and some other instructions and were best classified as i586. The C7's are definitely i686 compatible. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
On Wed, 6 May 2009, Vincent wrote: On Wed, 06 May 2009 14:02:20 +0100, Alan Lord (News) alansli...@gmail.com wrote: For a cheap backup to your VOIP service they do the job. I wouldn't use them for a proper system though. Thanks for the feedback. I have two more questions: 1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel Atom not doing much more than this and running Asterisk? The OSLEC benchmark tell me it can run 14 concurrent instances on a 550MHz VIA C3 processor. On a 1.6GHz Atom, it tells me it can do 80 concurrent instances, so I think the overhead of OSLEC is the least of your problems there. 2. Is it good enough to handle a single FXO line for professional use? I use OSLEC in my standard PBX products - Not with x100p cards though, but with Digium and OpenVox cards. However I'm in the UK - your lines may be different... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
On Wed, 6 May 2009 12:17:44 +0200, randulo spamsucks2...@gmail.com wrote: Those reading the thread amy be interested in Askozia pbx http://www.askozia.com/pbx/ Thanks for the link. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Preferred language for Asterisk AGIs development ?
On Tue, May 5, 2009 at 9:31 PM, Steve Edwards asterisk@sedwards.com wrote: I doubt any language is going to replace any other language for all future developments. The day one religion replaces all other religions, it may happen because languages = religions = distros = platforms = your subjective belief structures. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI + AGI for outbound click to dial
Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP _call_ to Asterisk box
I believe you can specify a default context: sip.conf [general] context=mysipcontext extensions.conf [mysipcontext] ; be careful allowing calls that could incur toll charges here, especially if this box is exposed to the internet exten = s,1,Answer() ;direct your call exten = s,n,Hangup() The version of zoiper I have running here (2.19, Library revision: 2875) works dialing sip:asterisk IP, or attempts to match an extension in the specified context sip:/extension@asterisk IP It only gets the default context if it isn't already registered to asterisk IP I don't know why zoiper wants one (and only one) forward slash in there. - Original Message - Is it possible to make a call from a SIP/IAX softphone, say Zoiper, on one computer to an Asterisk system without having an extension/account? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI + AGI for outbound click to dial
Send your call to a different extension that will set the header before calling your phone. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 10:51:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI + AGI for outbound click to dial
Ping-ponging the call? That's a good idea.. Now, to try to accomplish that in an AGI script. Thanks Jim! PB On Wed, May 6, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote: Send your call to a different extension that will set the header before calling your phone. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 10:51:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Jason Gehman General Manager North Voice Communications www.NorthVC.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridge() and Goto() and dialplan contexts, oh my!
I may or may not be experiencing the behavior described in:
http://bugs.digium.com/view.php?id=14241
I'm using asterisk-1.6.0.6, Bridge(), and I'm having a hangup context
executed when the caller is still on the line. These channels are all
SIP.
I want a group of expert callers who can dial in to the system, stay
put, and let other learner callers come to them. In between the
learner callers, I want them to hear hold music. :
exten = 111,1,Goto(Expert_Gateway)
exten = 222,1,Goto(Learners)
[Expert_Gateway]
exten = s,1,Answer
exten = s,2,AGI(authenticate_and_register)
exten = s,3,MusicOnHold()
exten = s,4,Goto(Expert_Gateway,s,3)
exten = s,5,HangUp
exten = h,1,AGI(cleanup_registration)
exten = h,2,HangUp
[Learners]
exten = s,1,Answer
exten = s,2,AGI(find_channel_with_expert)
exten = s,3,Bridge(${EXPERT_CHANNEL})
exten = s,4,NoOp(${BRIDGERESULT})
exten = s,5,HangUp
exten = h,1,NoOp(${BRIDGERESULT})
exten = h,2,HangUp
Everything looks good at first. However, I think I'm misunderstanding
something fundamental about Bridge(). Everything works great for the
first Learner call, the learner gets bridged to the expert.
The bad news is that on the Expert side of the dialplan, the Hangup
side of the Expert_Gateway context gets executed during the Bridge()
to the Learners, including the AGI(cleanup_registration).
Is it expected that the Expert side gets hungup context executed if
they haven't really hungup?
When the learner hangs up the expert goes back to hold music, that is,
after the Bridge() exits, the Expert s context continues and properly
follows the Goto.
Should I file a new bug about this, or is this expected behavior? Am I
misunderstanding something fundamental about purpose of Bridge(), and
is there something else I should be using instead?
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[asterisk-users] ConfBridge versus MeetMe
Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Can someone please tell me in which file the code for the application to be found? I was not able to find a file named app_bridge.c in the folder apps. app_bridge.c ? app_confbridge.c ? What are you looking for, exactly? The apps folder: http://svn.digium.com/svn/asterisk/trunk/apps/ So I have a question with regard to app_confbridge, which provides an application called ConfBridge. I did find a bug here that discusses bridging and confbridge, but not really what I wanted to know: http://bugs.digium.com/view.php?id=14389 Is there a good bug discussing the story behind the differences of ConfBridge and MeetMe, or where it came from, or if I'm right that ConfBridge is a new approach to MeetMe? What's the best way for me to be able to play with ConfBridge? Do I need to pull down trunk and build that? I just took a look through 1.6.1.0 for app_confbridge.c and it's not there, so perhaps this will be going into 1.6.2.* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Preferred language for Asterisk AGIs development ?
On Tue, 2009-05-05 at 17:00 -0500, Danny Nicholas wrote: Please elaborate; obviously ?? the dialplan is the simplest route to solve any problem. Dialplan is not the simplest route to solve ANY problem. It is the simplest route to solve simple problems. Writing a while loop in AEL or lua is much simpler then writing one in dialplan. Also writing large complex telephony applications is simpler in AEL or lua (or even AGI) because they give you better tools to manage complexity than raw dialplan does. Performance wise, lua should be faster than Dialplan, AEL, or AGI. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Preferred language for Asterisk AGIs development ?
Where are some good lua references? I don't think the O'reilly book even mentions it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Nicholson Sent: Wednesday, May 06, 2009 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Preferred language for Asterisk AGIs development ? On Tue, 2009-05-05 at 17:00 -0500, Danny Nicholas wrote: Please elaborate; obviously ?? the dialplan is the simplest route to solve any problem. Dialplan is not the simplest route to solve ANY problem. It is the simplest route to solve simple problems. Writing a while loop in AEL or lua is much simpler then writing one in dialplan. Also writing large complex telephony applications is simpler in AEL or lua (or even AGI) because they give you better tools to manage complexity than raw dialplan does. Performance wise, lua should be faster than Dialplan, AEL, or AGI. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge versus MeetMe
- David Backeberg dbackeb...@gmail.com wrote: Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Can someone please tell me in which file the code for the application to be found? I was not able to find a file named app_bridge.c in the folder apps. app_bridge.c ? app_confbridge.c ? What are you looking for, exactly? The apps folder: http://svn.digium.com/svn/asterisk/trunk/apps/ So I have a question with regard to app_confbridge, which provides an application called ConfBridge. I did find a bug here that discusses bridging and confbridge, but not really what I wanted to know: http://bugs.digium.com/view.php?id=14389 Is there a good bug discussing the story behind the differences of ConfBridge and MeetMe, or where it came from, or if I'm right that ConfBridge is a new approach to MeetMe? There is no bug because it wasn't really driven out of a bug being present. Development of it was driven by the internal way we bridge channels together and how it isn't flexible for what we really need. Let me explain a bit about each application. MeetMe as you know is a conferencing application that requires DAHDI. It doesn't just require DAHDI because of timing it also requires it because the actual conferencing engine/mixing takes place inside of DAHDI itself. Rewriting MeetMe not to use DAHDI would essentially be writing a new application. ConfBridge is a conferencing application that uses a new internal architecture for developers. The application itself is basically a user of the architecture and provides some additional outside capabilities like an IVR menu, join/leave sounds, etc. The actual mixing is done underneath in the architecture's core by a separate Asterisk module. I originally wrote ConfBridge as a test application of the architecture but in the end it just made sense to continue development on it and make it available as many individuals wanted a conferencing application that did not require DAHDI and it was simple to maintain. What's the best way for me to be able to play with ConfBridge? Do I need to pull down trunk and build that? I just took a look through 1.6.1.0 for app_confbridge.c and it's not there, so perhaps this will be going into 1.6.2.* If you would like to give it a test it is already available in 1.6.2 and the documentation for it available by typing core show application ConfBridge. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile and DTMF
Try the patch on this bug http://bugs.digium.com/view.php?id=15042 I don't get that error with my setup, but others have seen it. I am fairly sure of what is causing it. Still working on a fix. On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz wrote: I get a lot errors from chan_mobile when a call is in progress. More than one line inserted every second. ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9 Regards On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.com wrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI + AGI for outbound click to dial
In your AMI portion, you set the outgoing call first, then the extension you want to be reached at: Action: Originate Channel: Zap/g2/8135551212 Context: default Exten: 101 Priority: 1 Timeout: 3 In the dialplan: [default] exten = 101,1,SIPAddHeader(... exten = 101,n,Dial(... exten = 101,n,... J. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 11:42:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ping-ponging the call? That's a good idea.. Now, to try to accomplish that in an AGI script. Thanks Jim! PB On Wed, May 6, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote: Send your call to a different extension that will set the header before calling your phone. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 10:51:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Jason Gehman General Manager North Voice Communications www.NorthVC.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Understanding Codecs
Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and b A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 15554 Adding codec 0x2000 (amr) to SDP Adding non-codec 0x1 (telephone-event) to SDP Its not building ulaw or alaw into its list. Server B doesn't support AMR, so rejects the call. (I've no idea about the 0x4000 error - but I see it on both the good and bad servers, so I don't think its related). The odd thing is that the sip.conf files for A1 and A2 are exactly the same (save IP info). The build of the Asterisk server is from a 1.4.15 private build to add AMR, but, it's the same source built on both A1 and A2. I'm trying to figure out why A2 isnt offering ulaw and alaw. The codec seems ok, and is listed in the show codecs: 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 8192 (1 13) (0x2000) audioamr (AMR) But I cant see why its not transcoding across to ulaw/alaw. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Codecs
Forgot to add: sip.conf for both A1 and A2 has the following global codec definitions: disallow=all allow=clear allow=amr allow=ulaw allow=alaw The Asterisk build is a private build that adds the clear and AMR codec setups. The two servers are running Fedora, though A1s on 6 and A2s on 10. I cant see why that would make a difference though. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 06 May 2009 17:53 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Understanding Codecs Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and b A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 15554 Adding codec 0x2000 (amr) to SDP Adding non-codec 0x1 (telephone-event) to SDP Its not building ulaw or alaw into its list. Server B doesn't support AMR, so rejects the call. (I've no idea about the 0x4000 error - but I see it on both the good and bad servers, so I don't think its related). The odd thing is that the sip.conf files for A1 and A2 are exactly the same (save IP info). The build of the Asterisk server is from a 1.4.15 private build to add AMR, but, it's the same source built on both A1 and A2. I'm trying to figure out why A2 isnt offering ulaw and alaw. The codec seems ok, and is listed in the show codecs: 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 8192 (1 13) (0x2000) audioamr (AMR) But I cant see why its not transcoding across to ulaw/alaw. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] astcc - outgoing call does not hangup properly
Hi, I am using ASTCC and trying to setup a calling card platform. The problem that I have is that astcc does not hangup calls correctly: 1. If I try to dial a number, call goes through fine. When I hang up the call from my side I get this: -- Called 192.168.1.56/1XX6872 (masked a few digits) -- SIP/192.168.1.56-086c5000 is making progress passing it to SIP/581581-086b3000 -- SIP/192.168.1.56-086c5000 answered SIP/581581-086b3000 == Spawn extension (sippool, 1XX6872, 1) exited non-zero on 'SIP/581581-086b3000' 2. Dialing the same number, but hanging up from the remote side : -- Called 192.168.1.56/1XX6872 -- SIP/192.168.1.56-086c5000 is making progress passing it to SIP/581581-086b3000 -- SIP/192.168.1.56-086c5000 answered SIP/581581-086b3000 -- AGI Script astcc.agi completed, returning 0 -- Executing Hangup(SIP/581581-086b3000, NULL) == Spawn extension (sippool, 1XX6872, 2) exited non-zero on 'SIP/581581-086b3000' Difference between these two cases is: Case 1: call doesn't get billed (there was no hangup returned so astcc won't bill the call - although call was completed) Case 2: call gets billed and everything is fine. So what is wrong in the first case? Why don't I get the hangup correctly? PS: for case 1 I tried two different softphones, a SPA901 and a SPA942 so I guess this is not because of the phone. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7940 phones become unreachable over VPN after a time
Hi. I have a working internal Asterisk setup with 35 phones. Around 5-10 of these phones are physically located in a remote office via a VPN. I am completely happy with Asterisk and would be able to set up external calls but for one serious problem. After a period of time (perhaps a couple of days) the phones over the VPN cease to work. 'sip show peers' returns lines like: 3057/3057 172.16.254.2 D 5060 UNREACHABLE In this case 3057 will be able to make calls but it will not be able to hear the callee at the other end (though the callee can hear 3057). Other phones get 'The person at extension ... is unavailable' and then voicemail. We have tried out a software SIP client and this never goes into an UNREACHABLE state. Phones that don't go over the VPN are also fine. Rebooting the physical asterisk server will not make 3057 work again. Rebooting the phones is similarly useless. The only way I know of to fix the problem is to release the phone's ip and allow it to get a new one. This makes me suspect that either the phones or Asterisk is saving enough state to come back up in a non-working state or that there is a VPN issue. I've included a packet trace of 3057 while it is in a non-working state (below). The trace was done from the ASA. The Asterisk server is behind a Cisco ASA 5520 and the remote sites have Cisco 837s. I see a long list of SIP related issues resolved by http://www.cisco.com/en/US/docs/security/asa/asa80/release/notes/arn804n.html (which I applied), however this still doesn't seem to have solved the problem. Is a Asterisk = Cisco-VPN = 7940 setup known to work? Can anybody provide any suggestions to help debug this? If I'm unable to isolate/resolve the problem then its likely we'll have to drop the Asterisk solution and I've already grown rather attached to it. Thanks for any help, Ian No. TimeSourceDestination Protocol Info 460 2.561806172.16.254.2 10.4.4.102SIP Request: REGISTER sip:10.4.4.102 462 2.56250810.4.4.102172.16.254.2 SIP Status: 100 Trying(1 bindings) 467 2.58754610.4.4.102172.16.254.2 SIP Status: 200 OK(1 bindings) 489 2.713287172.16.254.2 10.4.4.102SIP Request: REGISTER sip:10.4.4.102 492 2.71395910.4.4.102172.16.254.2 SIP Status: 100 Trying(1 bindings) 501 2.72482210.4.4.102172.16.254.2 SIP Status: 200 OK(1 bindings) 685 3.711883172.16.254.2 10.4.4.102SIP Request: REGISTER sip:10.4.4.102 687 3.71258510.4.4.102172.16.254.2 SIP Status: 100 Trying(1 bindings) 694 3.74462710.4.4.102172.16.254.2 SIP Status: 200 OK(1 bindings) 838 4.86591310.4.4.102172.16.254.2 SIP Request: OPTIONS sip:3...@172.16.254.2:5060;transport=udp 853 4.95787310.4.4.102172.16.254.2 SIP Request: OPTIONS sip:3...@172.16.254.2:5060;transport=udp 981 5.87056710.4.4.102172.16.254.2 SIP Request: OPTIONS sip:3...@172.16.254.2:5060;transport=udp 1004 5.96257210.4.4.102172.16.254.2 SIP Request: OPTIONS sip:3...@172.16.254.2:5060;transport=udp 1135 6.674684172.16.254.2 10.4.4.102SIP Request: REGISTER sip:10.4.4.102 1137 6.67608810.4.4.102172.16.254.2 SIP Status: 100 Trying(1 bindings) 1143 6.70683310.4.4.102172.16.254.2 SIP Status: 200 OK(1 bindings) 1155 6.769650172.16.254.2 10.4.4.102SIP Request: REGISTER sip:10.4.4.102 1157 6.77015410.4.4.102172.16.254.2 SIP Status: 100 Trying(1 bindings) 1159 6.77605910.4.4.102172.16.254.2 SIP Status: 200 OK(1 bindings) 1179 6.87064310.4.4.102172.16.254.2 SIP Request: OPTIONS sip:3...@172.16.254.2:5060;transport=udp 1195 6.96197710.4.4.102172.16.254.2 SIP Request: OPTIONS sip:3...@172.16.254.2:5060;transport=udp 1844 7.771771172.16.254.2 10.4.4.102SIP Request: REGISTER sip:10.4.4.102 1847 7.77256510.4.4.102172.16.254.2 SIP Status: 100 Trying(1 bindings) 1849 7.80468310.4.4.102172.16.254.2 SIP Status: 200 OK(1 bindings) 1857 7.87071910.4.4.102172.16.254.2 SIP Request: OPTIONS sip:3...@172.16.254.2:5060;transport=udp 1871 7.96272510.4.4.102172.16.254.2 SIP Request: OPTIONS sip:3...@172.16.254.2:5060;transport=udp
[asterisk-users] shut down a single PRI on a running Asterisk system?
Excuse me if this is in the archives or on the net somewhere - I really did search. ;-) We are wondering if there is any way to shut down a single PRI without having to down Asterisk and/or interrupt other running PRI circuits. We use Asterisk servers with 4 port Digium PRI cards. In the last few days we ran into a situation where the 3 of the 4 PRIs were operating fine and had live calls on them while the 4th was going up and down. The server is remote and so physically pulling the cable is not a quick option. I did find one posting asking this question over a year ago and being told there was no way - I would like to confirm before we possibly change over our design - having a router or other device terminate the PRIs may be required if we cannot take a single circuit out service. Currently running Asterisk 1.4.18.1 but we are also interested if this is implemented in other versions. Also interested if any has other solutions to this problem. Thanks everyone! -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shut down a single PRI on a running Asterisk system?
On Wed, May 6, 2009 at 1:04 PM, James Van Vleet james.vanvl...@verety.com wrote: We are wondering if there is any way to shut down a single PRI without having to down Asterisk and/or interrupt other running PRI circuits. Somebody who knows Zaptel better could tell you whether this is a bad idea, but my first thought was: take out the config for that span from zaptel.conf, then run ztcfg. Although my memory is that will kill any calls on any zaptel device? A solution that has worked for us to physically switch circuits into Digium cards is: http://www.wti.com/AFS-Series/AFS-16-Automatic-AB-Fallback-Switch.html The AFS-16 has a serial device that you can then control with your favorite serial controller software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940 phones become unreachable over VPN after a time
On Wed, May 6, 2009 at 12:48 PM, i...@comtek.co.uk i...@comtek.co.uk wrote: Can anybody provide any suggestions to help debug this? If I'm unable to isolate/resolve the problem then its likely we'll have to drop the Asterisk solution and I've already grown rather attached to it. I have a number of ideas of what could be happening, and most involve routing issues over your VPN, or your VPN dropping packets. Here's a suggestion: * put another asterisk server on the remote side, and have the two asterisk servers do SIP or IAX trunks back and forth. If you don't want to invest in a server, at least pull an old computer off the curb and do some tests using that computer. If your phones come unregistered but your SIP trunk is fine, change your branch office phones to register to their local asterisk instead, and set your remote server accordingly. You might need to do some prefixing, or redirects, or other tricks to make the trunking transparent to the users if you don't want to reassign extensions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940 phones become unreachable over VPN after a time
On Wed, May 6, 2009 at 12:48 PM, i...@comtek.co.uk i...@comtek.co.uk wrote: Hi. I have a working internal Asterisk setup with 35 phones. Around 5-10 of these phones are physically located in a remote office via a VPN. I am There are a number of other reasons you want a remote phone server at that other location: *911 or name-the-emergency-service of your country *how do you call the other employees at the branch office when the WAN link is down? *this multiplies your options for future growth / expansion / flexibility / when you decide that you would rather have a dedicated telco link joining the offices. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get SIP resposnse codes
Hi all, I need to know the SIP response code from within the dial plan, someone could point me on how to? Gabriel Ortiz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile and DTMF
Thank you very much! I'll try with the patch and post the results. On Wed, May 6, 2009 at 12:37 PM, Matthew Nicholson mnichol...@digium.comwrote: Try the patch on this bug http://bugs.digium.com/view.php?id=15042 I don't get that error with my setup, but others have seen it. I am fairly sure of what is causing it. Still working on a fix. On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz wrote: I get a lot errors from chan_mobile when a call is in progress. More than one line inserted every second. ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9 Regards On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.com wrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile and DTMF
I think I misunderstood your mail. There is no patch available yet, right? I went to the page you linked but I did not found a patch file. On Wed, May 6, 2009 at 2:45 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.comwrote: Thank you very much! I'll try with the patch and post the results. On Wed, May 6, 2009 at 12:37 PM, Matthew Nicholson mnichol...@digium.comwrote: Try the patch on this bug http://bugs.digium.com/view.php?id=15042 I don't get that error with my setup, but others have seen it. I am fairly sure of what is causing it. Still working on a fix. On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz wrote: I get a lot errors from chan_mobile when a call is in progress. More than one line inserted every second. ERROR[6312]: chan_mobile.c:1050 mbl_read: read error 9 Regards On Tue, May 5, 2009 at 9:00 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you for your reply! I downloaded the latest revision of asterisk trunk and asterisk-addons trunk but it's not working at all, no key pulsation was detected. The last stable release at least detects first key pulsation. I checked out using: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons and svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk I also tried: svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 and svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 but compiling chan_mobile.c produce errors. What can I do? Thanks! On Tue, May 5, 2009 at 5:46 PM, Matthew Nicholson mnichol...@digium.com wrote: This is a known bug. It is fixed in the trunk version of chan_mobile. On Tue, 2009-05-05 at 11:38 -0400, Carlos Ruiz Diaz wrote: Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is being playing so I always have to background a sound to be able to receive DTMF tones. When I press a button (looking for IVR interaction) asterisk catches the correct key but the background song is inmediately muted and the next pulsation is not detected because of the previous problem. Is there a patch or a method to solve this problem? Thanks in advance. Carlos. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940 phones become unreachable over VPN after a time
I have a number of ideas of what could be happening, and most involve routing issues over your VPN, or your VPN dropping packets. Here's a suggestion: * put another asterisk server on the remote side, and have the two asterisk servers do SIP or IAX trunks back and forth. Thanks for the suggestion. An additional server in our 'main' remote location is likely to happen anyway at some stage, but there are still a couple of users who have a permanent VPN at home with a single phone and it just isn't practical to run Asterisk at every such location. 999 calls and 'WAN down' aren't really a big deal since every location has at least a couple of alternate means of communication (old fashioned landline or mobile). Ian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where are 2 letter language values defined?
I've googled for way too long, where are the 2 letter language values defined? I know: en = English es = Spanish fr = French but what about Croatian, Russian, Serbian, Vulcan, etc? Is there a list documented for Asterisk or is it just use the 2 letter country code Internet TLD? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are 2 letter language values defined?
They are 2-letter ISO country codes. http://www.iso.org/iso/english_country_names_and_code_elements On Wed, May 6, 2009 at 1:05 PM, Steve Edwards asterisk@sedwards.com wrote: I've googled for way too long, where are the 2 letter language values defined? I know: en = English es = Spanish fr = French but what about Croatian, Russian, Serbian, Vulcan, etc? Is there a list documented for Asterisk or is it just use the 2 letter country code Internet TLD? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are 2 letter language values defined?
Also check out: http://www.w3.org/International/questions/qa-lang-2or3.en.php On Wed, May 6, 2009 at 1:22 PM, Steve Johnson stevej...@gmail.com wrote: They are 2-letter ISO country codes. http://www.iso.org/iso/english_country_names_and_code_elements On Wed, May 6, 2009 at 1:05 PM, Steve Edwards asterisk@sedwards.com wrote: I've googled for way too long, where are the 2 letter language values defined? I know: en = English es = Spanish fr = French but what about Croatian, Russian, Serbian, Vulcan, etc? Is there a list documented for Asterisk or is it just use the 2 letter country code Internet TLD? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Dialplan Digitmaps
I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from the old configs though, and I need some help figuring out why these expressions lock up the new phone. Old Configs digitmap dialplan.digitmap=[2-9]11|[*]xx|891xxx|[1-7]xxx|8[0-46-8]xx|8500|851xxx|9,1[2-9]x|9,xxxT dialplan.digitmap.timeOut=3/ Template from Polycom digitmap dialplan.digitmap=[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT dialplan.digitmap.timeOut=3|3|3|3|3|3/ Anyone have any insight or suggestions on this issue, and on upgrading Polycom configs in general? -- Justin Phelps www.onitato.com 850.866.6864 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI + AGI for outbound click to dial
Would this return during the ring or only after the remote party has picked up? On Wed, May 6, 2009 at 12:51 PM, Jimmy Godbout s...@inbox.com wrote: In your AMI portion, you set the outgoing call first, then the extension you want to be reached at: Action: Originate Channel: Zap/g2/8135551212 Context: default Exten: 101 Priority: 1 Timeout: 3 In the dialplan: [default] exten = 101,1,SIPAddHeader(... exten = 101,n,Dial(... exten = 101,n,... J. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 11:42:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ping-ponging the call? That's a good idea.. Now, to try to accomplish that in an AGI script. Thanks Jim! PB On Wed, May 6, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote: Send your call to a different extension that will set the header before calling your phone. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 10:51:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Jason Gehman General Manager North Voice Communications www.NorthVC.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Jason Gehman General Manager North Voice Communications www.NorthVC.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI + AGI for outbound click to dial
In this case, Placing a call from an outgoing channel to a local extension, this will cause the local extension not to ring until the Zap channel has picked up - http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 16:06:36 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Would this return during the ring or only after the remote party has picked up? On Wed, May 6, 2009 at 12:51 PM, Jimmy Godbout s...@inbox.com wrote: In your AMI portion, you set the outgoing call first, then the extension you want to be reached at: Action: Originate Channel: Zap/g2/8135551212 Context: default Exten: 101 Priority: 1 Timeout: 3 In the dialplan: [default] exten = 101,1,SIPAddHeader(... exten = 101,n,Dial(... exten = 101,n,... J. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 11:42:43 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ping-ponging the call? That's a good idea.. Now, to try to accomplish that in an AGI script. Thanks Jim! PB On Wed, May 6, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote: Send your call to a different extension that will set the header before calling your phone. -Original Message- From: pallet...@gmail.com Sent: Wed, 6 May 2009 10:51:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial Ok - after a lot of playing I'm still a bit stuck. I'd like to accomplish the following - can't get it to work as it should (at least in my head! LOL) I've got an app that initiates an AMI call for Originate. I want to click a number onscreen, send the Originate, then (this is the part I can't figure out) have the box call my extension, adding in the SIPAddHeader info for answer-after:0 (so my phone auto-picks up) as well as spawning a MixMonitor to record the call in a specified format. (I have the AGI working for the mixmonitor) Make sense? maybe? ... chuckle Thanks for the help! PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Jason Gehman General Manager North Voice Communications www.NorthVC.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Jason Gehman General Manager North Voice Communications www.NorthVC.com GET FREE 5GB EMAIL - Check out spam free email with many cool features! Visit http://www.inbox.com/email to find out more! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shut down a single PRI on a running Asterisk system?
On Wednesday 06 May 2009 12:04:08 James Van Vleet wrote: We are wondering if there is any way to shut down a single PRI without having to down Asterisk and/or interrupt other running PRI circuits. We use Asterisk servers with 4 port Digium PRI cards. In the last few days we ran into a situation where the 3 of the 4 PRIs were operating fine and had live calls on them while the 4th was going up and down. The server is remote and so physically pulling the cable is not a quick option. I did find one posting asking this question over a year ago and being told there was no way - I would like to confirm before we possibly change over our design - having a router or other device terminate the PRIs may be required if we cannot take a single circuit out service. Currently running Asterisk 1.4.18.1 but we are also interested if this is implemented in other versions. Also interested if any has other solutions to this problem. In trunk (1.6.3 and later), we have a command pri service disable channel for doing this exact procedure. Support for this has been a long time coming, as the patch came from issue 3450 (http://bugs.digium.com/view.php?id=3450). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are 2 letter language values defined?
On Wednesday 06 May 2009 14:05:58 Steve Edwards wrote: I've googled for way too long, where are the 2 letter language values defined? I know: en = English es = Spanish fr = French but what about Croatian, Russian, Serbian, Vulcan, etc? Is there a list documented for Asterisk or is it just use the 2 letter country code Internet TLD? ISO 3166 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are 2 letter language values defined?
On Wed, May 6, 2009 at 1:05 PM, Steve Edwards asterisk@sedwards.com wrote: Is there a list documented for Asterisk or is it just use the 2 letter country code Internet TLD? On Wed, May 6, 2009 at 1:22 PM, Steve Johnson stevej...@gmail.com wrote: They are 2-letter ISO country codes. http://www.iso.org/iso/english_country_names_and_code_elements Which references ISO 3166-1-alpha-2 code elements. On Wed, 6 May 2009, Steve Johnson wrote: Also check out: http://www.w3.org/International/questions/qa-lang-2or3.en.php Which references ISO 639-1 two-letter language code. Which is why I love standards -- because you can have so many of them :) Without investing the time to note [any] differences, which one does Asterisk officially use? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shut down a single PRI on a running Asterisk system?
On Wed, 6 May 2009, Tilghman Lesher wrote: In trunk (1.6.3 and later), we have a command pri service disable channel for doing this exact procedure. Support for this has been a long time coming, as the patch came from issue 3450 (http://bugs.digium.com/view.php?id=3450). Will this work for RBS T1s as well? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with Sphinx
Hi, Did anyone tried speech recognition using Sphinx ? I used sphinx using this website (http://scribblej.com/svn/) but when i run astsphinx i am getting the following error. Any clue what might have caused this problem ? Thanks -Azher INFO: s2_semi_mgau.c(1080): 1 mixture Gaussians, 256 components, 4 feature streams, veclen 51 INFO: s2_semi_mgau.c(748): Loading senones from dump file /opt/sphinx/Communicator_semi_40.cd_semi_6000/sendump INFO: s2_semi_mgau.c(764): BEGIN FILE FORMAT DESCRIPTION INFO: s2_semi_mgau.c(793): Rows: 256, Columns: 6256 INFO: s2_semi_mgau.c(801): Using memory-mapped I/O for senones INFO: fe_interface.c(287): You are using the internal mechanism to generate the seed. INFO: feat.c(849): Initializing feature stream to type: 's2_4x', ceplen=13, CMN='current', VARNORM='no', AGC='none' INFO: cmn.c(142): mean[0]= 12.00, mean[1..12]= 0.0 INFO: dict.c(232): Allocating 20 placeholders for new OOVs ERROR: dict.c, line 556: 'hello': Unknown phone 'AH0' ERROR: dict.c, line 440: Failed to add hello to dictionary ERROR: dict.c, line 556: 'hello(1)': Unknown phone 'EH0' ERROR: dict.c, line 440: Failed to add hello(1) to dictionary INFO: dict.c(494): 0 = words in file [mydict] Restart checking timeout (1241647848 - 1241647845 2), 28 DIFF: 1.50, LIMIT: 7.50, RESTARTS: 18.66 Spawning too quickly Killed ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960G with static config
Hi, I am using Cisco 7960G with asterisk and it works perfect but it needs a dhcp/tftp server for ip address and configuration files. Is there any way i can config the phone with static configuration i.e. without dhcp/tftp ? Thanks -Azher ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shut down a single PRI on a running Asterisk system?
On Wednesday 06 May 2009 16:23:08 Jeff LaCoursiere wrote: On Wed, 6 May 2009, Tilghman Lesher wrote: In trunk (1.6.3 and later), we have a command pri service disable channel for doing this exact procedure. Support for this has been a long time coming, as the patch came from issue 3450 (http://bugs.digium.com/view.php?id=3450). Will this work for RBS T1s as well? No, only PRI. Simply put, RBS is too dumb of a protocol to let you do anything more than very rudimentary tasks. Taking a line out of service was not one that the designers thought about, so it's not part of the protocol. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] shut down a single PRI on a running Asterisk system?
On Wed, May 06, 2009 at 11:04:08AM -0600, James Van Vleet wrote: Excuse me if this is in the archives or on the net somewhere - I really did search. ;-) We are wondering if there is any way to shut down a single PRI without having to down Asterisk and/or interrupt other running PRI circuits. Brute force: zap destroy channel N zap destroy channel N+1 ... You'll need to restart Asterisk to get them back (if you'll want to) eventually. On newer (= 1.4.22) version of Asterisk, 'dahdi restart' will also do. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960G with static config
Yes (I assume you have it configured for SIP) Press the Setting button (square with check) Scroll down to the Unlock Config Enter the password of cisco Press the accept softkey Now you can scroll up to Network Configuration / SIP Configuration and manually change settings. Jimmy M. Ezell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Azher Mughal Sent: Wednesday, May 06, 2009 03:21 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7960G with static config Hi, I am using Cisco 7960G with asterisk and it works perfect but it needs a dhcp/tftp server for ip address and configuration files. Is there any way i can config the phone with static configuration i.e. without dhcp/tftp ? Thanks -Azher ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960G with static config
It works, Thanks. Jimmy Ezell wrote: Yes (I assume you have it configured for SIP) Press the Setting button (square with check) Scroll down to the Unlock Config Enter the password of cisco Press the accept softkey Now you can scroll up to Network Configuration / SIP Configuration and manually change settings. Jimmy M. Ezell -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Azher Mughal Sent: Wednesday, May 06, 2009 03:21 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7960G with static config Hi, I am using Cisco 7960G with asterisk and it works perfect but it needs a dhcp/tftp server for ip address and configuration files. Is there any way i can config the phone with static configuration i.e. without dhcp/tftp ? Thanks -Azher ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
I use these cards and they work pretty well. FWIW when Digium sold them they were also just winmodems with a resistor removed to change the PCI device ID. Later on the Zaptel driver included the device ID of the winmodem. I used to be able to get the winmodem itself for under $10, but I think they are discontinued now. Ambient = Intel, FWIW. If you want I'll dig out out and give you the details. If you need a large quantity I would try to find the winmodems that are compatible. On Wed, May 6, 2009 at 08:43, Vincent vincent.delpo...@bigfoot.com wrote: Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the computer's CPU? 2. Are they all bad, no matter what chipset is used (Intel, Motoral, Ambient)? If not, which offer good enough quality to handle a single POTS line? 3. Why are they often bad quality? Because the driver itself is badly written? Because PC's don't have enough speed to handle the tasks using their own CPU (hard to believe, but I don't know)? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice Mail Delete Notification
Do you know if there is a way to have an script run whenever a user has deleted a voicemail message? I want to have multiple users, all with there own passwords, share the same mailbox. When any one of them deletes a message I want it deleted from everyone's mailbox. I believe that I can do this by using symbolic links on the file system for their voice mail folders. However, I would like to keep a record of which users deleted messages when... I do not see a way to hook into the voice mail to do this. Thanks, -Brian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Sphinx
scribb...@scribblej.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Azher Mughal Sent: May-06-09 6:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with Sphinx Hi, Did anyone tried speech recognition using Sphinx ? I used sphinx using this website (http://scribblej.com/svn/) but when i run astsphinx i am getting the following error. Any clue what might have caused this problem ? Thanks -Azher INFO: s2_semi_mgau.c(1080): 1 mixture Gaussians, 256 components, 4 feature streams, veclen 51 INFO: s2_semi_mgau.c(748): Loading senones from dump file /opt/sphinx/Communicator_semi_40.cd_semi_6000/sendump INFO: s2_semi_mgau.c(764): BEGIN FILE FORMAT DESCRIPTION INFO: s2_semi_mgau.c(793): Rows: 256, Columns: 6256 INFO: s2_semi_mgau.c(801): Using memory-mapped I/O for senones INFO: fe_interface.c(287): You are using the internal mechanism to generate the seed. INFO: feat.c(849): Initializing feature stream to type: 's2_4x', ceplen=13, CMN='current', VARNORM='no', AGC='none' INFO: cmn.c(142): mean[0]= 12.00, mean[1..12]= 0.0 INFO: dict.c(232): Allocating 20 placeholders for new OOVs ERROR: dict.c, line 556: 'hello': Unknown phone 'AH0' ERROR: dict.c, line 440: Failed to add hello to dictionary ERROR: dict.c, line 556: 'hello(1)': Unknown phone 'EH0' ERROR: dict.c, line 440: Failed to add hello(1) to dictionary INFO: dict.c(494): 0 = words in file [mydict] Restart checking timeout (1241647848 - 1241647845 2), 28 DIFF: 1.50, LIMIT: 7.50, RESTARTS: 18.66 Spawning too quickly Killed ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
I'd say in life you get what you pay for.. and sometime you even pay for stuff that should be free.. These knockoff cards, can be built in-house for 20$ or less using an old walkie talkie, a rope, some standard matches, and an old MCgyver Tv episode..They do just that, echo the sound back to the other end.. no seriously.. Used 3, all 3 where thrown out either a window, or another opening .. It's great for testing don't get me wrong, once PSTN echo starts to drive you or the other party mad , or your cheap cap's start to go off specs, you'll say I should of bought 1 of the 50$ instead of 5 of the $10 ones.. of course, same strategy goes with china knockoffs, you run 6 pair of shoes for the price of 1 that lasts 6 times the other.. So basically for 10$ try it out.. Then when you want 2-3 ports, or do voice rec (spynx openmrc etc whatever) you'll need quality My 0.02 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Andrew Joakimsen Sent: May-06-09 7:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Questions on X100P/X101P cards I use these cards and they work pretty well. FWIW when Digium sold them they were also just winmodems with a resistor removed to change the PCI device ID. Later on the Zaptel driver included the device ID of the winmodem. I used to be able to get the winmodem itself for under $10, but I think they are discontinued now. Ambient = Intel, FWIW. If you want I'll dig out out and give you the details. If you need a large quantity I would try to find the winmodems that are compatible. On Wed, May 6, 2009 at 08:43, Vincent vincent.delpo...@bigfoot.com wrote: Hello, I'm looking for a dirt cheap solution for SOHO use to handle at most a couple of POTS lines, and I notice that X10?P cards go for $15 on eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma. I have a couple of questions about those cheap FXO cards: 1. Are they all glorified softmodems, ie. none has an on-board CPU or DSP and outsources all processing to the computer's CPU? 2. Are they all bad, no matter what chipset is used (Intel, Motoral, Ambient)? If not, which offer good enough quality to handle a single POTS line? 3. Why are they often bad quality? Because the driver itself is badly written? Because PC's don't have enough speed to handle the tasks using their own CPU (hard to believe, but I don't know)? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Sphinx
I sent email twice, but no reply :( ContactTel Business wrote: scribb...@scribblej.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Azher Mughal Sent: May-06-09 6:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with Sphinx Hi, Did anyone tried speech recognition using Sphinx ? I used sphinx using this website (http://scribblej.com/svn/) but when i run astsphinx i am getting the following error. Any clue what might have caused this problem ? Thanks -Azher INFO: s2_semi_mgau.c(1080): 1 mixture Gaussians, 256 components, 4 feature streams, veclen 51 INFO: s2_semi_mgau.c(748): Loading senones from dump file /opt/sphinx/Communicator_semi_40.cd_semi_6000/sendump INFO: s2_semi_mgau.c(764): BEGIN FILE FORMAT DESCRIPTION INFO: s2_semi_mgau.c(793): Rows: 256, Columns: 6256 INFO: s2_semi_mgau.c(801): Using memory-mapped I/O for senones INFO: fe_interface.c(287): You are using the internal mechanism to generate the seed. INFO: feat.c(849): Initializing feature stream to type: 's2_4x', ceplen=13, CMN='current', VARNORM='no', AGC='none' INFO: cmn.c(142): mean[0]= 12.00, mean[1..12]= 0.0 INFO: dict.c(232): Allocating 20 placeholders for new OOVs ERROR: dict.c, line 556: 'hello': Unknown phone 'AH0' ERROR: dict.c, line 440: Failed to add hello to dictionary ERROR: dict.c, line 556: 'hello(1)': Unknown phone 'EH0' ERROR: dict.c, line 440: Failed to add hello(1) to dictionary INFO: dict.c(494): 0 = words in file [mydict] Restart checking timeout (1241647848 - 1241647845 2), 28 DIFF: 1.50, LIMIT: 7.50, RESTARTS: 18.66 Spawning too quickly Killed ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
On Wed, May 6, 2009 at 8:47 PM, ContactTel Business li...@contacttel.com wrote: I'd say in life you get what you pay for.. and sometime you even pay for stuff that should be free.. I have to agree. I have a few of these cards I started out with. They were great for the wow, I finally got asterisk to do something but worthless for actually running in a system for any kind of real work. That being said, I have every intention of leaving one in the system as a quick way to get a cordless phone in our work area (we have an old cordless telephone laying around... hook it up, lets me call up front with no problems... that I can't live with). I would not suggest using these cheap cards in production systems where a little bit of bad service really matters. After all, it's for the phone system the one thing most people assume just always works... as always... ymmv.. -- jon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Messaging System
Hi to All, I need to implement an automatic telephone messaging system that works like this: -the system generates the call based on mysql records or any database -when the client answer the phone, the Asterisk PBX playback a recorded message -when finish, hang up the channel. Only for voice messages not SMS. Exists some application based on Asterisk that makes this, or any code to implement in dialplan Thanks in advance. Ricardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
Jonathan Moore wrote: On Wed, May 6, 2009 at 8:47 PM, ContactTel Business li...@contacttel.com wrote: I'd say in life you get what you pay for.. and sometime you even pay for stuff that should be free.. I have to agree. I have a few of these cards I started out with. They were great for the wow, I finally got asterisk to do something but worthless for actually running in a system for any kind of real work. That being said, I have every intention of leaving one in the system as a quick way to get a cordless phone in our work area (we have an old cordless telephone laying around... hook it up, lets me call up front with no problems... that I can't live with). Not sure how you would do that, as the X100 card is an FXO card, won't provide either battery or dial tone to the cordless. What you will want for that is an FXS card or ATA. The X100 card will connect to a central office line, and with the later software echo cancel works OK. Not nearly as bad as some have made it out to be, though for US/Canada lines. Not suitable for UK and others John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Messaging System
On Wed, 6 May 2009, Ricardo Melendez wrote: Hi to All, I need to implement an automatic telephone messaging system that works like this: -the system generates the call based on mysql records or any database -when the client answer the phone, the Asterisk PBX playback a recorded message -when finish, hang up the channel. Is this how do I do a call tree for my kid's soccer team or how do I bug the hell out of 20,000,000 innocents? Seriously though, a bit more information will go a long way. A script to read records from a database and originate calls using AMI or call files is pretty easy stuff. (The Devil is always in the details.) If there is money involved, your choice of the technology used to place the calls will be important -- stay away from analog PSTN hardware. What do you want to accomplish? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions on X100P/X101P cards
John Novack ha scritto: Not sure how you would do that, as the X100 card is an FXO card, won't provide either battery or dial tone to the cordless. What you will want for that is an FXS card or ATA. The X100 card will connect to a central office line, and with the later software echo cancel works OK. Not nearly as bad as some have made it out to be, though for US/Canada lines. Not suitable for UK and others The problem is: analog line is a delicated environment where impedance, volts, and line quality are some of the critical components. I found my X100 cards failing in production, no software component can solve the line impedance or other physical things. I try but no way out. After i bougt a 'real' analog board, even the worst is much much much better. I now, the cost is a problem, and there is NOT a single cheap analog board from DIGIUM or SANGOMA or others Bye. begin:vcard fn:Massimo Nuvoli n:Nuvoli;Massimo org:Progetto Archivio SRL adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia email;internet:mass...@archivio.it title:Amministratore Delegato tel;work:0121303544 tel;fax:0121040601 x-mozilla-html:FALSE url:www.progettoarchivio.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
