Re: [asterisk-users] Odd error in libpri

On Tue, May 03, 2011 at 04:30:46PM -0400, Richard Kenner wrote: > > Please create a mantis issue describing this problem. > > Pardon my ignorance, but what does "mantis" refer to? Useless trivia: A mantis eats bugs. But sadly the bug tracker is now called "issue tracker". http://issues.asterisk.

Re: [asterisk-users] default context overrides context of peer

It should work when host is dynamic. Is this a bug ? On Tue, May 3, 2011 at 9:03 AM, Deepesh D wrote: > This works when I change the host to non-dynamic and > insecure=port,invite for the peer, but does not work when > host=dynamic. > > Also my sip peers are realtime. If I remove the realtime pee

Re: [asterisk-users] Fading voice problem

On 3/05/11 10:16 PM, Eduardo Leones wrote: Guys, I'm having problems in the fading voice calls, receptive and active, that in SIP accounts. While few people using the system, calls are perfect, but it beats the normal use of connections (average 30 concurrent), the voice begins to fade from peop

[asterisk-users] asterisk 1.4.35 to 1.4.41

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame t

Re: [asterisk-users] How to debug MixMonitor misbehaviour

Thanks for the input. Yes, I did call out many times, but the recording doesn't happen even after the call is bridged and there is two way audio. I also took out the "b" option and so it should recording the ringing right (even before call is bridged) but it doesn't do that or any recording at all

Re: [asterisk-users] asterisk call forwarding

On Tue, 3 May 2011 18:10:55 -0400 Satish Patel wrote: > Thank you so much that solved my database issue. Now how asterisk > will forward call ? > > Or I need to specify gotoif statment in my stdexten to check > database key and take action? Yes, you need to write the dialplan to act on the key.

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

C F wrote: model name : AMD-K6(tm) 3D processor *shudder* Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ --

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

On Mon, May 2, 2011 at 11:46 PM, || dave cantera Mobile wrote: > I've been away from asterisk for a while since 1.4.16 and only installed 1.6 > once to run a test... can someone recommend what the best version to install > is and the recommended CPU/motherboard for an * box these days? I'm just >

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

On Tue, May 3, 2011 at 1:09 AM, A E [Gmail] wrote: > On Mon, May 2, 2011 at 9:45 PM, C F wrote: >> >> Just from my experience with different DBs, stay away from BLOB data >> types as much as possible. >> > Hi CF, > any particular reason why? I've had a good experience with it, in fact > that's re

Re: [asterisk-users] asterisk call forwarding

Thank you so much that solved my database issue. Now how asterisk will forward call ? Or I need to specify gotoif statment in my stdexten to check database key and take action? -- Sent from my iPhone On May 3, 2011, at 5:41 PM, Chad Wallace wrote: On Tue, 3 May 2011 18:45:32 + sa

Re: [asterisk-users] asterisk call forwarding

On Tue, 3 May 2011 18:45:32 + satish patel wrote: > > I found following dialplan on net but somehow its not going to set > CFIM in asterisk database (asterisk 1.8.3.3). Any idea ? > > exten => *72,1,Answer > exten => *72,2,Wait(1) > exten => *72,3,BackGround(please-enter-your) > exten =>

Re: [asterisk-users] Having redundancy, so if first IP failed then send for the other

On Tue, May 3, 2011 at 5:31 PM, bilal ghayyad wrote: > Hi All; > > I need to configure the SIP account so if first IP address failed then to > send for the second IP address. How to do this? > > While configuring the sip account, at the host parameter, can I give two IP > addresses separated by

[asterisk-users] Having redundancy, so if first IP failed then send for the other

Hi All; I need to configure the SIP account so if first IP address failed then to send for the second IP address. How to do this? While configuring the sip account, at the host parameter, can I give two IP addresses separated by comma? Or what should I do to have such redundancy? Regards Bila

Re: [asterisk-users] receive faxes

Enable debug and verbose on CLI ? Did you enable and also at logger.conf full => notice,warning,error,debug,verbose,dtmf,fax Date: Tue, 3 May 2011 16:12:06 -0400 From: vipki...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] receive faxes i have full log.. only thi

Re: [asterisk-users] Odd error in libpri

> > Please create a mantis issue describing this problem. > > Pardon my ignorance, but what does "mantis" refer to? > Mantis is the issue tracker at: https://issues.asterisk.org Richard -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Odd error in libpri

> Please create a mantis issue describing this problem. Pardon my ignorance, but what does "mantis" refer to? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduc

Re: [asterisk-users] Odd error in libpri

> As I'm reading this, libpri thinks that the SV8300 is complaining that > a "mandatory" IE is missing, in this case time/date. However, the > field is > THERE. But when I go back to a working libpri (r1878), I see that the > time/date is NOT sent on the CONNECT. > > If I'm reading Q.931 correctly

Re: [asterisk-users] receive faxes

i have full log.. only thing that stands out are two warnings: [May 3 16:10:40] WARNING[18176] app_fax.c: Error transmitting fax. result=13: Unexpected message received. [May 3 16:10:40] WARNING[18176] app_fax.c: Transmission failed On Tue, May 3, 2011 at 4:05 PM, satish patel wrote: > I'

Re: [asterisk-users] Asterisk 1.6 Questions

On 05/03/2011 01:16 PM, Gary Graves wrote: Can you answer both? Can Asterisk be configured to _initiate_ such a change at some point, mid-call? I don't know of a way to do that. I suppose it might be possible if a call were asynchronously transferred to a SIP peer that had different codec

Re: [asterisk-users] receive faxes

I'd enable full debug at logger.conf and try to find issue. -S Date: Tue, 3 May 2011 15:55:51 -0400 From: vipki...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] receive faxes I tried with those settings and without... same error: WARNING[18090]: app_fax.c:820 tran

Re: [asterisk-users] receive faxes

I tried with those settings and without... same error: WARNING[18090]: app_fax.c:820 transmit: Transmission failed On Tue, May 3, 2011 at 3:32 PM, satish patel wrote: > did you set faxdetect=both or incoming > > and faxbuffer=? > > -S > > -- > Date: Tue, 3 May 201

Re: [asterisk-users] [IAX] Everyone is busy/congested at this time (1:0/0/1)

add this line at the end of the IAX account definition and try again requirecalltoken=no On Wed, Apr 27, 2011 at 2:40 PM, John Alexis wrote: > Unfortunatelly that doesn't change anything. I got exactly the same error > ("Everyone is busy/congested at this time (1:0/0/1)" ... ). > I did a "dial

Re: [asterisk-users] receive faxes

did you set faxdetect=both or incoming and faxbuffer=? -S Date: Tue, 3 May 2011 15:28:36 -0400 From: vipki...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] receive faxes i have spandsp and app_fax.so is loaded but i get: app_fax.c:820 transmit: Transmission fail

Re: [asterisk-users] receive faxes

i have spandsp and app_fax.so is loaded but i get: app_fax.c:820 transmit: Transmission failed when trying to fax from a POTS line... On Tue, May 3, 2011 at 3:27 PM, satish patel wrote: > You need spandsp i guess following is my dialplan is working example > > [fax] > exten => 9000,1,Set(FAXFI

Re: [asterisk-users] receive faxes

You need spandsp i guess following is my dialplan is working example [fax] exten => 9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif) exten => 9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)}) exten => 9000,n,ReceiveFax(${FAXFILE}) exten => 9000,n,Hangup() Date: Tue, 3 May 2

[asterisk-users] receive faxes

does anybody know a good tutorial on how to setup asterisk to receive faxes (no need to send them) ? i've tried using "app_fax.so" with T38 but i keep getting "Transmission failed" -- _ -- Bandwidth and Colocation Provided by http:

Re: [asterisk-users] Orginate not working well with PSTN lines

On Fri, Apr 29, 2011 at 01:04:42AM +0200, Gilles wrote: > On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali > wrote: >> >> Anybody can explain me why asterisk is unable to detect ringback tone from >> PSTN telco ? . > > I guess it was a lot of work, and nobody bothered adding this to the > Zaptel dr

Re: [asterisk-users] asterisk call forwarding

I found following dialplan on net but somehow its not going to set CFIM in asterisk database (asterisk 1.8.3.3). Any idea ? exten => *72,1,Answer exten => *72,2,Wait(1) exten => *72,3,BackGround(please-enter-your) exten => *72,4,Playback(extension) exten => *72,5,Read(fromext,then-press-pound

Re: [asterisk-users] dial from voicemail

Kelly Opal wrote: I would like my customers to have a choice of dialing 1 to get my cell phone while in voicemail I use a combination of the follow me feature, along with the 'a' extension: [s-NOANSWER] exten => s,1,Gosub(mailbox_exist,s,1) exten => s,n,Set(VoiceMailCount=${VMCOUNT(${ARG1}@si

Re: [asterisk-users] dial from voicemail

From: Kelly Opal Sent: Tue 5/3/2011 1:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dial from voicemail Hi Is it possible to dial from within voicemail to reach another extension. I would like my customers to have a choice of dialing 1 to get my cell phone while in voi

[asterisk-users] asterisk call forwarding

Hey Guys! Anybody have basic and simple call forwarding dialplan ? I search on google and i found many but those are pretty complicated and most are for trixbox and GUI. -S -- _ -- B

Re: [asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend

Hmm... it's just that I've seen implementation of this done already, in Google Voice, BlindSide, BigBlueButton and others, however none provide a simple interface for voice-only broadcast from the browser. I'm sure there's a way to do it using Asterisk, I just don't know of it! Please suggest way

[asterisk-users] dial from voicemail

Hi Is it possible to dial from within voicemail to reach another extension. I would like my customers to have a choice of dialing 1 to get my cell phone while in voicemail or to just leave a message at the tone. Thanks Kelly -- _

Re: [asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend

On 11-04-30 03:10 PM, Alec Taylor wrote: > Good Evening, > > I'm setting up an Internet Radio website with call-in functionality, > and need to know the kinds of FOSS tools I should install to get the > job done. > > Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png >

Re: [asterisk-users] Asterisk 1.6 Questions

Can you answer both? Can Asterisk be configured to _initiate_ such a change at some point, mid-call? and Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by the dialog counterparty? On Tue, May 3, 2011 at 12:56 PM, Alex Balashov wrote: > On 05/03/2011 12:43

Re: [asterisk-users] Asterisk 1.6 Questions

On 05/03/2011 12:43 PM, Gary Graves wrote: Can you change codecs mid-call upon re-invite? Do you mean: can Asterisk be configured to _initiate_ such a change at some point, mid-call? Or do you mean: Will Asterisk properly react to such a re-INVITE and change codecs if asked to do so by th

[asterisk-users] Asterisk 1.6 Questions

I have a couple of questions about asterisk 1.6: Can you change codecs mid-call upon re-invite? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Thanks in advance for any insight. Gary -- _

Re: [asterisk-users] Asterisk, bicolor BLF and DEVSTATE

2011/5/3 Steven Howes > > On 3 May 2011, at 13:34, Olivier wrote: > > It seems that Asterisk DEVICE_STATE function can't be mapped with this > bicolor feature. > > So how could this 4-states BLF be implemented ? > > Any suggestion ? > > The handset is what maps the colours to the states. You can'

[asterisk-users] record call transfered from IAX softphone to SIP hardphone

hello List i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversat

Re: [asterisk-users] Join and listen to conference call through web-interface

Thanks, looks really helpful for managing connected users (half my problem). On the web-interface question, how do I create a website with a [call-in] button? I'm using Drupal, so will make it a members only page. Basically they click the [call-in] button, and straight away they're in the convers

Re: [asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing

On Tue, May 03, 2011 at 09:32:06AM -0500, Dean Hoover wrote: > I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an > HP ML110 G6 using Ubuntu Linux 10.04 LTS. > > I have two Digium TE121 single T1 port cards and a Digium AEX800 > 8-port FXS card. All PCI Express cards. > > Co-

[asterisk-users] Multiple cards using same IRQ - getting IRQ errors and hissing

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ erro

[asterisk-users] audiohook.c: Failed to get 160 samples from write factory

Hello, I see a lot of these messages in the debug log : /[May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohook.c: Failed to get 160 samples from write factory 0xae17e18 [May 3 15:47:09] DEBUG[19081] audiohoo

[asterisk-users] asterisk 1.8 rpms and additional modules from source

Hi All i have a asterisk install from rpm (digium yum) and i wish to install res spandsp i comipled from source using the same asterisk version and copyied the file to the module directory but when trying to load i get this WARNING[18463] loader.c: Module 'res_fax_spandsp.so' was not compiled w

Re: [asterisk-users] Asterisk, bicolor BLF and DEVSTATE

On 3 May 2011, at 13:34, Olivier wrote: > It seems that Asterisk DEVICE_STATE function can't be mapped with this > bicolor feature. > So how could this 4-states BLF be implemented ? > Any suggestion ? The handset is what maps the colours to the states. You can't send a colour in SIP. You could

[asterisk-users] Asterisk, bicolor BLF and DEVSTATE

Hi, 1. You can now find several SIP harphones with bicolor BLFs (see Polycom, Cisco, ...). Is there a protocol which best describes how to use this bicolor BLD feature ? 2. I would like to map these BLFs to the following user activities : - user is logged off: no light - user is logged in: gr

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

On Tue, May 03, 2011 at 07:26:25AM -0400, A E [Gmail] wrote: > yes, after having noodled with this for a while, it appears that the best > way would be to somehow replicate what the Realtime Voicemail does but just > not bother storing the media files in the DB. Extracting the BLOB data, > storing

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

On Tue, May 3, 2011 at 5:19 AM, Tzafrir Cohen wrote: > On Tue, May 03, 2011 at 01:09:14AM -0400, A E [Gmail] wrote: > > On Mon, May 2, 2011 at 9:45 PM, C F wrote: > > > > > Just from my experience with different DBs, stay away from BLOB data > > > types as much as possible. > > > > > > Hi CF, > >

Re: [asterisk-users] ATA refuses to answer a call?

2011/5/3 Ernie Dunbar > I'm kind of at a loss to diagnose problems like this, yet we get them a > lot. > > - The ATA (Thomson 784 in this particular case) is logged into the > Asterisk server. 'sip show peer' shows their IP address, port, and > useragent. > - The ATA is connected directly to the

[asterisk-users] Fading voice problem

Guys, I'm having problems in the fading voice calls, receptive and active, that in SIP accounts. While few people using the system, calls are perfect, but it beats the normal use of connections (average 30 concurrent), the voice begins to fade from people. Soon I figured some network problem,

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

On Tue, May 03, 2011 at 01:09:14AM -0400, A E [Gmail] wrote: > On Mon, May 2, 2011 at 9:45 PM, C F wrote: > > > Just from my experience with different DBs, stay away from BLOB data > > types as much as possible. > > > > Hi CF, > any particular reason why? I've had a good experience with it, in fa

Re: [asterisk-users] Retrieving/Streaming audio/video files from DB using over AGI

On Tue, May 3, 2011 at 4:41 AM, Thorsten Göllner wrote: > Am 02.05.2011 15:59, schrieb A E [Gmail]: > > On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] wrote: > >> Hello All, >> >> Probably a silly question, but we're wondering if people have had any >> experience and have data to demonstrate if t

Re: [asterisk-users] Retrieving/Streaming audio/video files from DB using over AGI

Am 02.05.2011 15:59, schrieb A E [Gmail]: On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] wrote: Hello All, Probably a silly question, but we're

Re: [asterisk-users] Password to be ecrypted?

> On Tuesday 26 Apr 2011, bilal ghayyad wrote: >> Hi All; >> >> I am using Asterisk 1.8, how I can protect my self from hackers in >> case they was able to see my sip.conf file? I need the password to be >> encrypted, how? > > Short answer: You can't. Asterisk itself needs to be able > to read t