[asterisk-users] Send DTMF's before the call is answered

Is there a way to send DTMF's to a channel before the call is answered? For example, send DTMF's to a SIP channel after the 180 Ringing or 183 Session Progress have been received from it, but before the 200 OK, or in the E1 side, after the Q931_ALERTING is received, but before the

[asterisk-users] TE406P not working on Intel D101Ggc motherboard.

I recently moved a TE406P card from an Intel D865GBF motherboard (where it worked fine), to an Intel D101Ggc card, and now I can't get the spans to got up correctly. All I get is an endless burst of: == Primary D-Channel on span 4 up == Primary D-Channel on span 2 up !! Got a UA, but i'm in

[asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's

It appears that CPU1 in not taking any interrupts - What steps do I need to do bring up CPU1 and share IRQ requests for a Linux noob? Run the IRQ balance daemon (/usr/sbin/irqbalance). It's part of the kernel-utils RPM. -- Atte. Álvaro Palma

[asterisk-users] Re: transform bridged call into a conference

Hi! I wonder if it is possible to transform a bridged call into a conference. E.g. phone 1 calls phone 2 (normal bridged call with Dial()). Further phone 3 wants to join? Is this possible? Can you please refer me the proper applications? What I am doing: First, redirect to bridged calls to

[asterisk-users] Getting strange behavior on SIP channels after upgrade to 1.2.11

I upgraded to 1.2.11 and now I see two behaviors different than the existent in 1.2.10: 1.- I get 183 Session Progress instead of 180 Ringing. 2.- If I have three extensions, A, B and C. A using codec X, B using also codec X and C using codec Y. If C dials to B and A tries to pick up the call

[asterisk-users] Which card do you recommend for heavy load application?

I'm thinking to implement an application that may need 120 channels (4 E1 spans) being recorded in WAV49 format simultaneously, with echo cancellation, etc. What card would you recommend for this kind of load? (independently of the underlying hardware, assume the best possible). I've tested a

[asterisk-users] Problem handling agents and queues vía RealTime

Hi. I'm developing an application that dynamically adds agents to a queue. First, the queue is created via RealTime, and then the agents are added vía Manager, using QueueMemberAdd action. However, for some reason, most of the time (NOT ALWAYS, and this is the strange behavior), Asterisk

[Asterisk-Users] Limiting the size of a Queue

Is there a way to limit the size of a Queue? I want to create a queue with for example, 5 agents, and only allow at most 10 persons waiting so this way, they don't saturate my entire PSTN span, which can be also simultaneously used for another Queues or for my outgoing calls. Thanks a lot for

[Asterisk-Users] IAX softphone with RSA support?

Which (preferible free :-) softphone that supports IAX and RSA encryption do you recommend? It seems that IDEFisk doesn't yet. Thanks a lot for your help. -- Atly. Alvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com --

[Asterisk-Users] Generate two calls from Asterisk and bridge them

Is there a way in Asterisk (I guess there's, it's only I can't figure out how :-)) to: 1.- Generate a call to channel 1 (example, to PSTN vía an E1 card, using Zap/g1) 2.- Generate a call to channel 2 (example, an internal SIP extension). 3.- Once both channel have answered, connect the

[Asterisk-Users] Transfer extensions processing control to Manager

I'm developing an application that monitors the state of the incoming calls using Manager events. So, as a part of it, I need to override the control of the extensions by the dialplan itself. The problem is that, if I don't declare the incoming extension, Asterisk hangs up the call by default.

[Asterisk-Users] Problem with options to Dial application

I'm trying to set a dialing rule in my dialplan. As a part of it, from my point of view, this works wrong priorityjumping=no [test_context] exten = 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With j flag exten = 1234,2,Playback(digits/2) exten = 1234,3,Playback(digits/3) exten =

[Asterisk-Users] Re: VoiceMail application: j option not working as I supposed

: Try: exten = _XX,2,VoiceMail([EMAIL PROTECTED],j) and exten = _XX,110,VoiceMail([EMAIL PROTECTED],j) Álvaro Palma wrote: I've the following dialplan. exten = _XX,hint,SIP/${EXTEN} exten = _XX,1,Dial(SIP/${EXTEN},10,j) exten = _XX,2,VoiceMail([EMAIL PROTECTED],u|j) exten

[Asterisk-Users] Re: VoiceMail application: j option not working as I supposed

Ok, I finally fix it. It seems to be a bug in the app_voicemail.c file: http://bugs.digium.com/view.php?id=7164 -- Atly. Alvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

[Asterisk-Users] Re: Problem setting locale for voicemail

step by step procedure to implement voice mail in asterisk. Thanks Regards, Chandra. Álvaro Palma [EMAIL PROTECTED] wrote: I've set voicemail almost successfully, only a minor detail remains :-) I can't get the dates in my local language (spanish). In sip.conf, zapata.conf and voicemail.conf

[Asterisk-Users] VoiceMail application: j option not working as I supposed

I've the following dialplan. exten = _XX,hint,SIP/${EXTEN} exten = _XX,1,Dial(SIP/${EXTEN},10,j) exten = _XX,2,VoiceMail([EMAIL PROTECTED],u|j) exten = _XX,3,Hangup() exten = _XX,102,Goto(110) exten = _XX,103,Playback(pbx-invalid) exten = _XX,104,Hangup() exten = _XX,110,VoiceMail([EMAIL

[Asterisk-Users] Problem setting locale for voicemail

I've set voicemail almost successfully, only a minor detail remains :-) I can't get the dates in my local language (spanish). In sip.conf, zapata.conf and voicemail.conf, I've set: language=es and my locale is es also. However, the days and months names still appear in english in the

[Asterisk-Users] Is there a way to not propagate a context included inside other context?

the propagation of preincluded context beyond the scope of the context that explicitaly include it. Thanks a lot for your answer. -- Atly. Álvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] How to distinguish between UNEXISTENT channels v/s UNAVAILABLE channels

extension doesn't receive a 404 Not Found in that case, so he doesn't know he dialed a wrong number. Thanks a lot for your attention, patience and help. -- Atly. Álvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

[Asterisk-Users] Problem with Asterisk and Grandstream HT286

I've dealing with this issue for a while, and I'd really like to know if anybody has experienced the same pain before :-) I've a lot of Grandstream HandyTone 286, loaded with the latest firmware (1.0.8.16) from the GS website. In my sip.conf, this ATA's are configured as: [05] type=friend

[Asterisk-Users] Can't get Pickup app working

I'm trying to set the Pickup feature. I'm setting my extensions.conf as: exten = _*.,1,Pickup(SIP/${EXTEN:1}) but if, for example, extension 03 is ringing by a call made from extension 01, and I try to pick it up from extension 02 (by dialing *03 from extension 02), I can see in the Asterisk

[Asterisk-Users] Strange second REINVITE being sent

I'm using Asterisk a SIP Server for a lot of GrandStream HandyTone ATA's. Each one of them is configured in sip.conf as: [1234567] type=friend username=1234567 secret=1234567 callerid=ATA 1234567 host=dynamic nat=yes qualify=yes disallow=all allow=g729 canreinvite is set globally to YES. When

[Asterisk-Users] Problem with INVITE's being sent

I've being testing a couple of GrandStream ATA 286 which with no reason start responding 486 Busy to all new incoming INVITES. They are connected to an Asterisk installation as SIP client. Running ethereal between them, I could notice that, for some reason unknown for me at this time, Asterisk

[Asterisk-Users] Speeding up the dial of DTMF's in SIP channel

was looking the options for the Dial command, and nothing like that appears on it. Thanks a lot for your help. -- Atly. Álvaro Palma ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

[Asterisk-Users] Extracting info from the $EXTEN variable

PROTECTED]) How can I do that? I was looking the documentation, but it only speaks of stripping digits from the extension, not to selecting a range. Thanks a lot for your help. -- Atly. Álvaro Palma ___ --Bandwidth and Colocation provided

[Asterisk-Users] Upgrading Asterisk witk G729 license installed

I've an Asterisk 1.2.4 installation, where I've also installed the G729 codec license. I'd like to upgrade that installation to 1.2.5, but I'm not sure if I'll lost the license in the process (and if I'll be able to recover it later!!!). Is there any special consideration I've to keep in mind

[Asterisk-Users] Changing REINVITE status of the channel dynamically

I've an Asterisk server running in my office, which forwards all long distance calls to a third party SIP service using an extension rule: exten = _1XX0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED]) (1XX0 is the international calls rule for Chile) Also, in my sip.conf, I've defined canreinvite=yes to

[Asterisk-Users] Information to program a new driver for Asterisk

I'm interested in developing a new channel driver for a thrid party telephony card for Asterisk. Is there any official document that explains how to do this? We've been looking the doc/channel.txt and doc/modules.txt in the source, but that's not a very complete source of info :) Thanks a

[Asterisk-Users] Codec order sent wrong from Asterisk

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following

[Asterisk-Users] How can I force Asterisk t not override my codec order?

I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to

[Asterisk-Users] Is SIP canreinvite working ok?

I've the following situation: Phone A: Codec GSM supported Phone B: Codec iLBC supported in sip.conf: [general] ... disallow=all allow=gsm allow=ilbc allow=alaw allow=ulaw canreinvite=yes ... (There's a lot of other SIP users, that's why I made the default codec list bigger than just GSM