Re: [asterisk-users] sending sms from Asterisk server

On 08/19/2010 08:21 AM, Tiago Geada wrote: > I would rather use .call files. So easy to produce a text file... > > On 18 August 2010 21:02, Steve Edwards @sedwards.com > wrote: > > Un-top-posting... > > On 08/17/2010 09:00 AM, Tino wrote: > >

Re: [asterisk-users] channel variables in AGI

On Wednesday, August 11, 2010 11:08:37 am Tino wrote: > #!/bin/bash -x > T="$agi_uniqueid" > > I want to save value of 'agi_uniqueid' channel variable into a variable > called 'T' in my script When executing and AGI from the dialplan, it will dump out it's variables immediately, so you need to t

Re: [asterisk-users] sending sms from Asterisk server

I would rather use .call files. So easy to produce a text file... On 18 August 2010 21:02, Steve Edwards wrote: > Un-top-posting... > > On 08/17/2010 09:00 AM, Tino wrote: >> >> I would like to send sms to some external phone numbers from my asterisk >> server. Is it possible to send sms via so

Re: [asterisk-users] realtime sip peers : musiconhold class

Hi to convert wav file use following sox 'orgFile' -w -r 8000 -c 1 -s 'fixedFile' while replace orgFile and fixedFile with actual filenames If still now luck try with mp3 Regards -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] IXJ issues on 1.4.35

On Wed, 18 Aug 2010, Infra wrote: > On Wed, 18 Aug 2010, Infra wrote: > > > > > My thanks for previous help on fixing IXJ issues in 1.2.40; I now > > have problems with a just-built 1.4.35 on the same host: > > > > I applied the patch for dialtone as per my issue on 1.2.40. > the fxs line is

Re: [asterisk-users] IXJ issues on 1.4.35

On Wed, 18 Aug 2010, Infra wrote: > > My thanks for previous help on fixing IXJ issues in 1.2.40; I now > have problems with a just-built 1.4.35 on the same host: > > [Aug 18 17:26:48] WARNING[27209]: app_dial.c:1298 dial_exec_full: Unable > to create channel of type 'Phone' (cause 0 - Unknown) >

[asterisk-users] IXJ issues on 1.4.35

My thanks for previous help on fixing IXJ issues in 1.2.40; I now have problems with a just-built 1.4.35 on the same host: [Aug 18 17:26:48] WARNING[27209]: app_dial.c:1298 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) == Everyone is busy/congested at this time (1

Re: [asterisk-users] Realtime Context

Hi Matt, That's somewhat closer to what I do in my dialplan as well. But Dan apparantly wants to add new contexts in his `extensions` table. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-18 6:02 PM, "Matt Riddell" wrote: We don't use a context for that. We set up dialplan code in a non

Re: [asterisk-users] Realtime Context

We don't use a context for that. We set up dialplan code in a non asterisk part of MySQL called "routing types". When a customer selects a DDI number they can choose a routing type to use with it. These routing types allow for variable substitution - i.e. if someone adds the routing type "Dir

Re: [asterisk-users] WaitExten() always times out

I must not be receiving them properly, since I can't make it work. I just can't see why :P. My asterisk version is 1.6.2.6. Like I said before, for outgoing .call files WaitExten works fine, it's on incoming calls that I cannot receive the number I need. I had not checked my dtmf mode, this is ne

Re: [asterisk-users] CDR variables

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Subject: [asterisk-users] CDR variables >Hello list! >I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables in h >It seems that these variables always re

Re: [asterisk-users] WaitExten() always times out

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: Re: [asterisk-users] WaitExten() always times out >My .call file goes out to a pstn number. >That work around would be perfect :D, but I need the number given by the

Re: [asterisk-users] MySQL Connect problem...

This is what I ended up doing, working fine now. Cheers On 18 August 2010 08:52, Nasir Iqbal wrote: > Avoid to use MySQL dialplan application, instead write an AGI script for > this purpose > > On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee wrote: > >> Right, I'm baffled. >> >> I have: >> exten

Re: [asterisk-users] WaitExten() always times out

Hi, Are you sure asterisk is receiving and processing DMTF correctly? Are you using rfc2833, SIP INFO or inband DMTF? What is your asterisk version? I use WaitExten(5) all the time, no matter if they are single-digit or multiple-digit extensions. Regards, -- Ing. Miguel Molina Grupo de Tecn

Re: [asterisk-users] WaitExten() always times out

My .call file goes out to a pstn number. That work around would be perfect :D, but I need the number given by the caller. On Wed, Aug 18, 2010 at 2:49 PM, Danny Nicholas wrote: > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of

[asterisk-users] CDR variables

Hello list! I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables in h It seems that these variables always return 0. I am using Asterisk version 1.6.2.11. Can't I get these values other than using CDR reccords ?? -- __

Re: [asterisk-users] WaitExten() always times out

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: Re: [asterisk-users] WaitExten() always times out >Thanks for you reply :). >I thought of that and tried replacing _X with a numbers it should match (9), and it di

Re: [asterisk-users] WaitExten() always times out

Thanks for you reply :). I thought of that and tried replacing _X with a numbers it should match (9), and it didn't work. It still times out as if no number was entered. On Wed, Aug 18, 2010 at 2:11 PM, Danny Nicholas wrote: > *From:* asterisk-users-boun...@lists.digium.com [mailto: > aste

Re: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks

sg01*CLI> core show version Asterisk 1.6.2.11 built by root @ localhost.localdomain on a i686 running Linux on 2010-08-16 15:17:26 UTC On Wed, Aug 18, 2010 at 10:19 AM, Danny Nicholas wrote: > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Beha

Re: [asterisk-users] WaitExten() always times out

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones Subject: [asterisk-users] WaitExten() always times out >Hi, >My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receiv

[asterisk-users] WaitExten() always times out

Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten => xxx,1,Background(hello-world) exten => xxx,2,WaitExten(7) exten => _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, bu

Re: [asterisk-users] sending sms from Asterisk server

Un-top-posting... On 08/17/2010 09:00 AM, Tino wrote: I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn wrote: T

Re: [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?

That is set and here is what I get: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 01217 DCall: 0 [44.55.66.77:4569] USERNAME: 9988 REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Sub

[asterisk-users] WaitExten() always times out

Hi, My WaitExten() is not working as I expect it to. This is the relevant part of my context. It is meant to receive incoming calls. [incoming] exten => xxx,1,Background(hello-world) exten => xxx,2,WaitExten(7) exten => _X,1,AGI(myAGI.php) When I send the call from a .call, it works perfect, bu

Re: [asterisk-users] sending sms from Asterisk server

Le 18/08/2010 16:03, Tino a écrit : > Hello Johann, > > Thanks for your advice in this matter. But i am not sure how to pass > the numbers to be sent sms in the dialplan. agi(script,param1,param2,...,paramX) from your dialplan where script lies in /var/lib/asterisk/agi-bin > > On Wed, Aug 18, 20

Re: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Wood Subject: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for allinbound trunks Since you can see the CLI log, please post your asterisk version (core show version) so

[asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for all inbound trunks

Sending this to asterisk-users, in case anyone has AsteriskNOW experience they can share. Joe -- Forwarded message -- From: Joe Wood Date: Wed, Aug 18, 2010 at 9:22 AM Subject: AsteriskNow REGISTER'ing s@ extension for all inbound trunks To: asterisk...@lists.digium.com Hello.

[asterisk-users] Playing with sipvicious ..

... using it as a tool and understanding what it does... So one part of it's toolset identifys valid SIP accounts - and I was under the impression that alwaysauthreject=yes was supposed to stop this... However, it sends a request for a highly probably non-existent account, then sends requests

Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX

On Tue, 2010-08-17 at 09:42 -1000, Ben Schorr wrote: > Sorry, I should clarify - we have had a similar setup (IPSEC VPN, Polycom > 331) working at a different location with a different handset for this same > firm. We've never gotten the phone/VPN to work at this particular site. I > was just

Re: [asterisk-users] sending sms from Asterisk server

Hello Johann, Thanks for your advice in this matter. But i am not sure how to pass the numbers to be sent sms in the dialplan. On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn wrote: > On 08/17/2010 09:00 AM, Tino wrote: > >> Hello, >> >> I would like to send sms to some external phone numbers fro

Re: [asterisk-users] Monitor asterisk

Look into astassisstant. I don't remember the website, but google will take you their. It doesn't need any installations on the server, just a manager user in manager.conf. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-18 8:39 AM, "Shazaum" wrote: man, see monast http://monast.sourceforg

Re: [asterisk-users] Realtime Context

Thanks for the details. I Agreed you know what you are doing. As for doing it more elegantly, it is not possible to do a fruitful discussion without knowing all the details of how your dialplan works, neither is this the goal here. But I offer similar hosted PBX service, just use one context for al

Re: [asterisk-users] Monitor asterisk

man, see monast http://monast.sourceforge.net/ -- Renato dos Santos shazaum.wordpress.com 2010/8/17 Matt Riddell > On 17/08/10 6:34 PM, Hans Witvliet wrote: > > On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote: > >> Might be worth your time to check out: http://www.humbuglabs.org

Re: [asterisk-users] Realtime Context

Hi I'll give you an example process flow Our customer logs onto our VoIP Portal and orders a SIP Geographic number Geo number is successfully ordered which inserts an entry into our extensions table in MySQL DB in the default context which has a goto command to a newly created specific context

Re: [asterisk-users] Realtime Context

So why you need a reload when you are using the real-time architecture? Are you adding contexts in extensions.conf? If yes then where are you using the real-time? I asked because to me it seems you didn't understand what Dan is asking, and also seems you don't know how real-time works in regards t

Re: [asterisk-users] Realtime Context

Unless I've got some massive misunderstanding, yes we do as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime Why do you ask? On 17/08/10 17:27, Zeeshan Zakaria wrote: Ishfaq, do you use the asterisk real-time architecture? Zeeshan A Zakaria -- www.ilovetovoip.com

[asterisk-users] Enhancing snmp mib

Hi, I'm quite pleased with the asterisk/res_snmp integration (at least a "right" one :) not some hackish scripted thingy) but i felt it's missing quite a few datas. What i would need is: * Per channels, number of inbound call received since asterisk startup (like a network interface) *

Re: [asterisk-users] MySQL Connect problem...

Avoid to use MySQL dialplan application, instead write an AGI script for this purpose On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee wrote: > Right, I'm baffled. > > I have: > exten => s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2) > exten => s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recor

Re: [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?

Hi, Use requirecalltoken=no in your peer configuration Regards On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce wrote: > Hello Everyone, > > I am trying to diagnose issue with my IAX2 extension not working. > > When I have iax2 set debug on all I see is this: > > *Rx-Frame Retry[ No] -- OSeqno: 00

Re: [asterisk-users] sending sms from Asterisk server

Johann Hoehn wrote: > On 08/17/2010 09:00 AM, Tino wrote: >> Hello, >> >> I would like to send sms to some external phone numbers from my >> asterisk server. Is it possible to send sms via softphones like >> X-Lite ? . Any tips regarding this will be helpful >> >> thanks >> >> > This is easy to do