Re: [asterisk-users] bare minimum /etc/asterisk for sip based *

On Fri, 14 Aug 2009, Eric Fort wrote: > What files at a bare minimum need to be in /etc/asterisk for an asterisk > server that does sip only and voicemail. I'm setting up an asterisk > server to provide service for a single SIP softphone extension with SIP > origination and termination. The m

Re: [asterisk-users] bare minimum /etc/asterisk for sip based *

On 15/08/09 2:20 PM, Eric Fort wrote: > What files at a bare minimum need to be in /etc/asterisk for an > asterisk server that does sip only and voicemail. I'm setting up an > asterisk server to provide service for a single SIP softphone > extension with SIP origination and termination. The main

[asterisk-users] bare minimum /etc/asterisk for sip based *

What files at a bare minimum need to be in /etc/asterisk for an asterisk server that does sip only and voicemail. I'm setting up an asterisk server to provide service for a single SIP softphone extension with SIP origination and termination. The main purpose of using * is for voicemail and future

Re: [asterisk-users] Time of Day Routing

Tilghman Lesher wrote: > Regardless of how you think it should work, the poster above described > precisely the way it works. If your end boundary is 12:00, it will evaluate > as true all the way up until 12:01:59. If you don't want that, another poster > has suggested using 11:59, which will wor

Re: [asterisk-users] onnecting two asterisk using B410p BRI cards

Use a standard network cable - but you have to activate the 'terminate' jumper on the NT end. - Also, the new BRI stuff in dahdi is much easier to work with than misdn. PaulH voip crazy wrote: > Hello all, > > I'm trying to conect two asterisk servers using two B410p Digium > cards. One card o

Re: [asterisk-users] CURL function with SSL

On Friday 14 August 2009 16:14:32 Wenbin Zhang wrote: > Tilghman Lesher wrote: > > On Friday 14 August 2009 15:48:08 Tilghman Lesher wrote: > >> On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote: > >>> Tilghman Lesher wrote: > On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote: > > Da

Re: [asterisk-users] CURL function with SSL

Tilghman Lesher wrote: > On Friday 14 August 2009 15:48:08 Tilghman Lesher wrote: > >> On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote: >> >>> Tilghman Lesher wrote: >>> On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote: > David Gibbons wrote: >

Re: [asterisk-users] CURL function with SSL

Tilghman Lesher wrote: > On Friday 14 August 2009 15:48:08 Tilghman Lesher wrote: > >> On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote: >> >>> Tilghman Lesher wrote: >>> On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote: > David Gibbons wrote: >

Re: [asterisk-users] CURL function with SSL

On Friday 14 August 2009 15:48:08 Tilghman Lesher wrote: > On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote: > > Tilghman Lesher wrote: > > > On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote: > > >> David Gibbons wrote: > > >>> You probably want to set the option > > >>> > > >>> CURLOPT_SSL

Re: [asterisk-users] CURL function with SSL

On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote: > Tilghman Lesher wrote: > > On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote: > >> David Gibbons wrote: > >>> You probably want to set the option > >>> > >>> CURLOPT_SSL_VERIFYPEER to FALSE. > >>> > >>> Especially with chained certificates

Re: [asterisk-users] CURL function with SSL

This gets into a bit of a "hack", but you could do a quick-and-dirty AGI to do the WGET then set the variable based on what you got back. If you did it in perl, you could actually use LWP instead of WGET. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-us

Re: [asterisk-users] Number of Phone Numbers per Outgoing CALL File

There are at least two solutions to this dilemma: #1. dial using a context instead of a number Or #2 when creating the call files, use a future time (I use 60 seconds plus the expected length of the call). The PERL module Asterisk::Outgoing is supposed to handle it, but I had to modify mine t

Re: [asterisk-users] CURL function with SSL

Danny Nicholas wrote: > You could do a System(wget xx)... > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wenbin Zhang > Sent: Friday, August 14, 2009 3:09 PM > To: Asterisk Users Mailing List - Non-Comme

[asterisk-users] Number of Phone Numbers per Outgoing CALL File

Is it possible to place multiple phone numbers in a single outbound .call file? If I try doing this, only the last phone number in the file is called. However, if I use 1 file per phone number, then Asterisk attempts to process all generated CALL files at once, incrementing the retry count for

Re: [asterisk-users] CURL function with SSL

You could do a System(wget xx)... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wenbin Zhang Sent: Friday, August 14, 2009 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [aste

Re: [asterisk-users] CURL function with SSL

Tilghman Lesher wrote: > On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote: > >> David Gibbons wrote: >> >>> You probably want to set the option >>> >>> CURLOPT_SSL_VERIFYPEER to FALSE. >>> >>> Especially with chained certificates (cheapos from godaddy, etc), I have >>> had lots of troub

Re: [asterisk-users] CURL function with SSL

On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote: > David Gibbons wrote: > > You probably want to set the option > > > > CURLOPT_SSL_VERIFYPEER to FALSE. > > > > Especially with chained certificates (cheapos from godaddy, etc), I have > > had lots of trouble with CURL being able to validate a c

Re: [asterisk-users] Complete neutral Spanish sounds

Try http://www.voipnovatos.es/voces Un saludo Enrique Mora Context M.I.S. SL em...@context.es logo-email De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Alejandro Cabrera Obed Enviad

Re: [asterisk-users] Complete neutral Spanish sounds

I assume that the supplied voice for Spanish is cepstal - marta (www.cepstral.com ). In my (English) installation, there are about 1800 files, so I can see how/why the set might be incomplete. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk

[asterisk-users] Meaning of " requested special control 20, passing it to SIP"

Received this on the console -- IAX2/76.21.238.129:4569-4986 requested special control 20, passing it to SIP/magicjack-08225a58 Did a Google search, but reached a dead end Can anyone explain? Something need to be changed in my configuration? The call completed satisfactorily. Inbound IAX trunk

Re: [asterisk-users] no ring tone

i am not sure what you are talking about. i have extensions and my sip trunk config in that file. see below [200] deny=0.0.0.0/0.0.0.0 type=friend secret=200 qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/200 context=fr

[asterisk-users] Complete neutral Spanish sounds

Dear all, does anybody know about a complete set of neutral Spanish sounds to use in my Asterisk voicemail ??? Because when I get a Spanish sounds package, it always is incomplete. I live in Argentina, so I prefer neutral voices. Special thanks Alejandro

Re: [asterisk-users] i have a error in ivr

No. don’t erase them. You need to move procall3.wav to /var/lib/asterisk/sounds/custom/procall3.wav. The nickel tour of procall [procall] exten => s,1,Set(TIMEOUT(digit)=7) ; Set timeout for digits in background (4) exten => s,2,Set(TIMEOUT(response)=10) Set timeout for response in bg (4) e

Re: [asterisk-users] no ring tone

Hello One question In sip.con or sip_additionals.conf, in freepbx, the context of your client do you put nat = yes externip = You put your public ip. Are you sure that? Regards On Fri, Aug 14, 2009 at 10:43 AM, Ott Rose wrote: > i changed it and still didn't ring. however it did ring

Re: [asterisk-users] chan_dahdi refuses to build

Boehm, Matthew wrote: > Sorry. Fat-finger. 1.6.1.2. Found out that --with-tonezone is required > when ./configure'ing asterisk in order to get chan_dahdi to compile. > > Could that be documented somewhere? Or auto checked in ./configure if > --with-dahdi is specified? Wasted over an hour trying to

Re: [asterisk-users] Vicidial now extension setup

This would have to do with the reason they're being rejected, wouldn't it? What is the reason? Tareq Kibria wrote: > Dear All, > > I am trying to using E1 PRI Connection with vicidialnow > setup..Calls are landed in asterisk .From Asterisk CLI i can see the > caller id from where

[asterisk-users] Same Problem with AEX808E Re: No incoming audio on Dahdi channels (TDM410P)

Hello, i got the same problem with a Digium Card an AEX808E, with dahdi linux 2.2.0 dahdi tools 2.2.0 and asterisk 1.6.1.4 i did update to the latest of everything hoping it will fix the problem, but it still remains. i got: Aug 14 02:29:16 ctg01 kernel: [ 9257.702038] wctdm24xxp0: Missed int

[asterisk-users] Vicidial now extension setup

Dear All,   I am trying to using E1 PRI Connection with vicidialnow setup..Calls are landed in asterisk .From Asterisk CLI  i can see the caller id from where the call came ..but the calls are rejects.      So what should i do now for forwarding this call to a agent who is using sof

Re: [asterisk-users] Cdr error? Help!

Dpto. de Sistemas escribió: > The error is here, > > Q-aereos","1147739512","9536","outbound","1147739512","DAHDI/ > 30-1","SIP/9536-137a4cc0","Dial","SIP/9536|60|t","2009-08-13 > 13:41:03","2009-08-13 13:41:06","2009-08-13 13:44:01", > 178,175,"ANSWERED","DOCUMENTATION","1250170815.1512","" > >

Re: [asterisk-users] i have a error in ivr

yes procall3 is in /var/lib/asterisk/sounds/procall3/wav erase these: exten => i,1,Playback(invalid) exten => i,2,Playback(goodbye) exten => i,3,hangup exten => t,1,goto(procall,s,1) exten => h,1,Hangup ? On Fri, Aug 14, 2009 at 9:56 AM, Danny Nicholas wrote: > Procall3 is /var/lib/asterisk/

Re: [asterisk-users] chan_dahdi refuses to build

Sorry. Fat-finger. 1.6.1.2. Found out that --with-tonezone is required when ./configure'ing asterisk in order to get chan_dahdi to compile. Could that be documented somewhere? Or auto checked in ./configure if --with-dahdi is specified? Wasted over an hour trying to figure this out. Thanks, -Matt

Re: [asterisk-users] chan_dahdi refuses to build

Boehm, Matthew wrote: > Using asterisk 1.6.2.1 and dahdi 2.2.0.2. dadhi-linux installed just > fine. Using dahdi_dummy as there is no card in system. Did not install > libpri, again, no card. There isn't any 1.6.2.1 release of Asterisk; which version did you try to build? -- Kevin P. Fleming Di

Re: [asterisk-users] i have a error in ivr

Procall3 is /var/lib/asterisk/sounds/procall3.wav? IMO, procall should look like this: [procall] exten => s,1,Set(TIMEOUT(digit)=7) ; exten => s,2,Set(TIMEOUT(response)=10) exten => s,3,Set(CHANNEL(language)=en) ; define como idioma predefinido el ingles y usas las voces en este idioma exten =

[asterisk-users] chan_dahdi refuses to build

Using asterisk 1.6.2.1 and dahdi 2.2.0.2. dadhi-linux installed just fine. Using dahdi_dummy as there is no card in system. Did not install libpri, again, no card. When compiling asterisk, I include -with-dahdi and everything ./configure's fine but when I do 'make', everything goes fine but chan_

Re: [asterisk-users] Cdr error? Help!

The error is here, Q-aereos","1147739512","9536","outbound","1147739512","DAHDI/ 30-1","SIP/9536-137a4cc0","Dial","SIP/9536|60|t","2009-08-13 13:41:03","2009-08-13 13:41:06","2009-08-13 13:44:01", 178,175,"ANSWERED","DOCUMENTATION","1250170815.1512","" 9536 is extension and 1147739512 is dst,

Re: [asterisk-users] CURL function with SSL

David Gibbons wrote: > You probably want to set the option > > CURLOPT_SSL_VERIFYPEER to FALSE. > > Especially with chained certificates (cheapos from godaddy, etc), I have had > lots of trouble with CURL being able to validate a cert. That's probably > because I didn't tell it where the root cer

Re: [asterisk-users] no ring tone

i changed it and still didn't ring. however it did ring on one call to a cell phone but it hasn't done it again. Date: Fri, 14 Aug 2009 09:39:33 -0500 From: crt.ro...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] no ring tone Hello, I never use externhost y use \

Re: [asterisk-users] no ring tone

yes i just copied that form the freepbx site. sorry about that > From: st...@geekinter.net > To: asterisk-users@lists.digium.com > Date: Fri, 14 Aug 2009 15:43:00 +0100 > Subject: Re: [asterisk-users] no ring tone > > > On 14 Aug 2009, at 15:18, Ott Rose wrote: > > > how do i troubleshoot no r

Re: [asterisk-users] CURL function with SSL

You probably want to set the option CURLOPT_SSL_VERIFYPEER to FALSE. Especially with chained certificates (cheapos from godaddy, etc), I have had lots of trouble with CURL being able to validate a cert. That's probably because I didn't tell it where the root certs were... but either way. -Dave

Re: [asterisk-users] CURL function with SSL

Tilghman Lesher wrote: > On Friday 14 August 2009 09:04:12 Wenbin Zhang wrote: > >> Hi all, >> I hope you guys can help me out. I got a problem with using function >> CURL. I >> did Set(CURL=${CURL(URL)}); but the URL I was using is https, so when I >> generated the call, the CURL function c

Re: [asterisk-users] Asterisk + CDRTool

Did you get CDRTool to work with Asterisk or Areski's CDR Stats? On Fri, Aug 14, 2009 at 10:20 AM, harry R wrote: > Hi > > I just solve my problem today. Just a package on redhat that I need > install. > > H. > > ___ > -- Bandwidth and Colocation Pro

Re: [asterisk-users] i have a error in ivr

the audio is in format wav i save in Format PCM attributte 8,000 KHz; 8bit; Mono 7kb/s my extension.conf is the next: exten => 651085,1,Playback(procall3) exten => 651085,n,Playback(procall3) exten => 651085,n,Queue(procall|n|||) exten => 651085,n,Playback(voicemail-invitation) ex

Re: [asterisk-users] i have a error in ivr

Playback is expecting a frequency of 8000. use sox to correct. As for 101/103, that is how the dialplan is written, not an error per se. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo Sanchez Sent: Friday, August

[asterisk-users] i have a error in ivr

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[2593

Re: [asterisk-users] CURL function with SSL

On Friday 14 August 2009 09:04:12 Wenbin Zhang wrote: > Hi all, > I hope you guys can help me out. I got a problem with using function > CURL. I > did Set(CURL=${CURL(URL)}); but the URL I was using is https, so when I > generated the call, the CURL function could not get access to that > https

Re: [asterisk-users] no ring tone

On 14 Aug 2009, at 15:18, Ott Rose wrote: > how do i troubleshoot no ring tone. It was working and all i added > was the lines below now it doesn't ring. > > Edit sip_nat.conf for proper NAT: > localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set > your external hostname name h

Re: [asterisk-users] no ring tone

Hello, I never use externhost y use \ externip=public ip And work fine Regards On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose wrote: > how do i troubleshoot no ring tone. It was working and all i added was the > lines below now it doesn't ring. > > Edit sip_nat.conf for proper NAT: > localne

Re: [asterisk-users] Asterisk + CDRTool

Hi I just solve my problem today. Just a package on redhat that I need install. H. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asteris

Re: [asterisk-users] Cdr error? Help!

On 14 Aug 2009, at 15:01, Dpto. de Sistemas wrote: > hi all > do you guys know why asterisk sometimes, in the cdr put the dst (the > extension) number in the src ?? > I have 4 digit extensions (DID) (95XX) and sometimes, the same > values if found > in the src that usually have the calling user c

[asterisk-users] no ring tone

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your ex

[asterisk-users] Call no reject when receive 'PROGRESS with cause code 27 received' in zap channel

Hi, I have an asterisk connected with PRI (Zap channels). If I try to call a number, and recieve cause code 27 because the line 553192 is out of service, but the call continue...is it ok? Here the console messages -- Executing [...@troncal-pri-76:5] Dial("Zap/1-1", "Zap/g1/553192") in new sta

[asterisk-users] CURL function with SSL

Hi all, I hope you guys can help me out. I got a problem with using function CURL. I did Set(CURL=${CURL(URL)}); but the URL I was using is https, so when I generated the call, the CURL function could not get access to that https://URL server. What should I do with it? Thank you very much __

[asterisk-users] Cdr error? Help!

hi all do you guys know why asterisk sometimes, in the cdr put the dst (the extension) number in the src ?? I have 4 digit extensions (DID) (95XX) and sometimes, the same values if found in the src that usually have the calling user caller id. Example Q-aereos.csv "Q-aereos","9532","4379","

Re: [asterisk-users] Time of Day Routing

On Friday 14 August 2009 08:15:55 Tony Mountifield wrote: > In article <4a855630.5080...@arcdiv.com>, SIP wrote: > > Tony Mountifield wrote: > > > In article <05d03313-994b-4892-b045-f61332ddb...@geekinter.net>, > > > > > > Steve Howes wrote: > > >> On 14 Aug 2009, at 09:17, Neeraj Chand wrote: >

Re: [asterisk-users] Time of Day Routing

In article <4a855630.5080...@arcdiv.com>, SIP wrote: > Tony Mountifield wrote: > > In article <05d03313-994b-4892-b045-f61332ddb...@geekinter.net>, > > Steve Howes wrote: > > > >> On 14 Aug 2009, at 09:17, Neeraj Chand wrote: > >> > >> > >>> Asterisk version 1.4 > >>> From: Neeraj Chand >

Re: [asterisk-users] Time of Day Routing

Tony Mountifield wrote: > In article <05d03313-994b-4892-b045-f61332ddb...@geekinter.net>, > Steve Howes wrote: > >> On 14 Aug 2009, at 09:17, Neeraj Chand wrote: >> >> >>> Asterisk version 1.4 >>> From: Neeraj Chand >>> Sent: Friday, 14 August 2009 8:17 PM >>> To: 'asterisk-users@lists.d

Re: [asterisk-users] Time of Day Routing

Tony Mountifield wrote: > In article <05d03313-994b-4892-b045-f61332ddb...@geekinter.net>, > Steve Howes wrote: > >> On 14 Aug 2009, at 09:17, Neeraj Chand wrote: >> >> >>> Asterisk version 1.4 >>> From: Neeraj Chand >>> Sent: Friday, 14 August 2009 8:17 PM >>> To: 'asterisk-users@lists.di

Re: [asterisk-users] Time of Day Routing

Tony Mountifield wrote: > Hmm, I would still consider it a bug, whether on 1 or 2 minute resolution. > I haven't seen the 2 minutes issue with the below: GotoIfTime(07:59-16:59|mon-fri|*|*?office-hours,s,1) Our plant closes at 5pm. And, at exactly 5pm, the afterhours context takes over.

Re: [asterisk-users] Time of Day Routing

In article <05d03313-994b-4892-b045-f61332ddb...@geekinter.net>, Steve Howes wrote: > > On 14 Aug 2009, at 09:17, Neeraj Chand wrote: > > > Asterisk version 1.4 > > From: Neeraj Chand > > Sent: Friday, 14 August 2009 8:17 PM > > To: 'asterisk-users@lists.digium.com' > > Subject: [asterisk-users]

Re: [asterisk-users] play prompt after hanup

On Friday, August 14, 2009, Rilawich Ango wrote: >Hi, > > Can I play a prompt after hanging up a call? I have tried below but failed. > > >... >exten => s,n,Dial(SIP/1234) >... > >exten => h,1,Playback(demo-instruct) > > >-- Executing [...@macro-safedial:2] Playback("SIP/3601-09856bf0", >"de

[asterisk-users] onnecting two asterisk using B410p BRI cards

Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first

[asterisk-users] play prompt after hanup

Hi, Can I play a prompt after hanging up a call? I have tried below but failed. ... exten => s,n,Dial(SIP/1234) ... exten => h,1,Playback(demo-instruct) -- Executing [...@macro-safedial:2] Playback("SIP/3601-09856bf0", "demo-instruct") in new stack [Aug 14 17:24:03] WARNING[2496]: file

[asterisk-users] Astricon 2009 - dCAP

Hi folks, Going to astricon this year? Feeling a bit nervous as planning to take the exam this time. Any one else doing the same? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoen

Re: [asterisk-users] Time of Day Routing

On 14 Aug 2009, at 09:17, Neeraj Chand wrote: > Asterisk version 1.4 > From: Neeraj Chand > Sent: Friday, 14 August 2009 8:17 PM > To: 'asterisk-users@lists.digium.com' > Subject: [asterisk-users] Time of Day Routing > > Hi David, > > With this: >ifTime(00:00-12:00|*|*|*) > > Whatever time yo

[asterisk-users] Time of Day Routing

Hi David, With this: ifTime(00:00-12:00|*|*|*) Whatever time you specify at the end, I believe asterisk continues to evaluate this condition as true for 2 more minutes. So in this case, it will be valid for 00:00-12:02, even though you've specified 12:00 Cheers! Neeraj

Re: [asterisk-users] Time of Day Routing

Asterisk version 1.4 From: Neeraj Chand Sent: Friday, 14 August 2009 8:17 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] Time of Day Routing Hi David, With this: ifTime(00:00-12:00|*|*|*) Whatever time you specify at the end,

Re: [asterisk-users] Snom Phones Registration/Failover Feature

On Aug 13, 2009, at 6:24 PM, Usman Tahir wrote: > Hi Raimund, Hello, > > snom uses basically the same concept. As explained under: > http://wiki.snom.com/Settings/user_failover_identity. > > You select the line id that should be used when a registration fails. thank you for the link, as a matter

[asterisk-users] CPU Spikes in asterisk connected via IAX trunk

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients a