Re: [asterisk-users] Time counting while playback
> On Tue, 16 Mar 2010, Pham Quy wrote: > >> How can I count down 60s? MixMonitor app doesnt have any time out >> argument. On Tue, 16 Mar 2010, Jeff LaCoursiere wrote: > > I think you would be more successful and have more control if you wrote > it as an AGI. Then you could set a timer that would interrupt the > process and you could do what you like from there (hangup?). I think > you are asking too much of the dialplan. I would tend to leap into an AGI also, but did you try setting an absolute timeout? Externivr() may also be a good approach. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time counting while playback
On Tue, 16 Mar 2010, Pham Quy wrote:
> Hi all,
>
> This question has been asked for days, I think that would be more
> comprehensible if i post it in a new thread.
>
> What i want to do is something like karaoke. when users call to
> asterisk, a music song is played while caller sings. Their voice
> will be recorded and mixed with the music. To do that i used
> MixMonitor() and Playback() applications.
>
> I also want to enable users to select a part of song to be recorded
> (monitored) for example: Users press '*' to start recording. For
> stopping record, there are two ways: (1) he press '#'to stop recording
> OR it will be stopped (stop MixMonitor) AUTOMATICALLY after 60 seconds.
>
> How can I count down 60s? MixMonitor app doesnt have any time out
> argument.
>
> I detect '#' using Read() app as following
>
>
> [ivr-test]
> exten => test,1,Answer()
> exten => test,n,Wait(2)
> exten => test,n(prompt),Read(digit,hello-world,1,,3,2)
> exten => test,n,NoOp("Input digit - $[${digit}]")
> exten => test,n,GotoIf($[${digit} = 1]?one,1)
> exten => test,n,GotoIf($[${digit} = #]?sharp,1)
> exten => test,n,GotoIf($["${digit}" = ""]?nokey,1)
> exten => test,n,Goto(prompt)
> exten => test,n,Hangup()
>
> exten => one,1,NoOp(1 pressed)
> exten => one,n,Hangup()
>
> exten => sharp,1,NoOp(You press # )
> exten => sharp,n,HangUp()
>
> exten => nokey,1,NoOp(No key pressed)
> exten => nokey,n,Hangup()
> ---
>
> But it couldnt read #, key '#' have recognized as NoKey
>
> ps: sorry for my english
>
> Quyps
>
I think you would be more successful and have more control if you wrote it
as an AGI. Then you could set a timer that would interrupt the process
and you could do what you like from there (hangup?). I think you are
asking too much of the dialplan.
j
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[asterisk-users] Time counting while playback
Hi all,
This question has been asked for days, I think that would be more
comprehensible if i post it in a new thread.
What i want to do is something like karaoke. when users call to
asterisk, a music song is played while caller sings. Their voice
will be recorded and mixed with the music. To do that i used
MixMonitor() and Playback() applications.
I also want to enable users to select a part of song to be recorded
(monitored) for example: Users press '*' to start recording. For
stopping record, there are two ways: (1) he press '#'to stop recording
OR it will be stopped (stop MixMonitor) AUTOMATICALLY after 60 seconds.
How can I count down 60s? MixMonitor app doesnt have any time out
argument.
I detect '#' using Read() app as following
[ivr-test]
exten => test,1,Answer()
exten => test,n,Wait(2)
exten => test,n(prompt),Read(digit,hello-world,1,,3,2)
exten => test,n,NoOp("Input digit - $[${digit}]")
exten => test,n,GotoIf($[${digit} = 1]?one,1)
exten => test,n,GotoIf($[${digit} = #]?sharp,1)
exten => test,n,GotoIf($["${digit}" = ""]?nokey,1)
exten => test,n,Goto(prompt)
exten => test,n,Hangup()
exten => one,1,NoOp(1 pressed)
exten => one,n,Hangup()
exten => sharp,1,NoOp(You press # )
exten => sharp,n,HangUp()
exten => nokey,1,NoOp(No key pressed)
exten => nokey,n,Hangup()
---
But it couldnt read #, key '#' have recognized as NoKey
ps: sorry for my english
Quyps
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[asterisk-users] Asterisk 1.4.24 DUNDi CLI commands not found
Are there DUNDi CLI commands for Asterisk 1.4? I have searched google and I only see the dundi commands in Asterisk 1.6, although I see reference to them in older version's of Asterisk such as Asterisk 1.4 here: http://www.asteriskguru.com/tutorials/cli_cmd_14.html. When I view the CLI commands through help I don't see any of the dundi commands and there are errors when I run a command such as dundi debug asterisknowdev5*CLI> dundi debug No such command 'dundi debug' (type 'help dundi debug' for other possible commands) Here is my version output. asterisknowdev5*CLI> show version Asterisk 1.4.24 built by root @ localhost.localdomain on a i686 running Linux on 2009-03-20 21:27:25 UTC I can reload dundi by doing the following: module reload pbx_dundi.so, but is there a way I can get these commands? Thanks, John Haigh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up RTP to flow between endpoints directlybypassing Asterisk
Vikram- > http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly > > The link above indicates that it is possible to setup RTP streams to > directly flow between endpoints and completely bypass Asterisk. I would > like to know if this configuration would work when, > > a) both endpoints are behind NAT, and/or > b) both endpoints don't support same codecs > > with media flowing through a SIP+rtpproxy server that can do > transcoding ? This would be 'native bridging' mode as I've seen it described a few places on the web, correct? If Asterisk is "out of the RTP loop", then what can it still do? Only billing? It would not detect DTMF, no RTP record or announcement playout, etc. I'm not clear on whether anyone actually uses Asterisk in this mode and if so, for what reason. -Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Article - a method on how to evaluate an Asteriskserver
Ioan- Sounds like this would give a useful measurement regardless of server type, network config, and other variable issues. That should be a great tool. Do you have any plans to test with Asterisk in 'native bridging' mode? I.e. with RTP streams not touched in any way by Asterisk? I assume that would be the absolute max that Asterisk can handle. -Jeff > I would like to share with you an article [1] we have issued last week > (sorry, currently only in Romanian language - we plan to provide an > English version soon). > > This article is describing a method to be used for obtaining the > maximum number of SIP simultaneous calls an Asterisk server could > process safely (meaning no errors/maintain control of the machine and > without RTP frame drops) > > We used SIPP (with modified uas and uac_pcap scenarios) + 2 scripts > for controlling the test (one is running on the tested Asterisk server > - start-test.sh, for data collection and load analysis and the other > is running on the SIPP+Asterisk testing machine, for call quality > control and SIPP instance control - sipp-controller.sh) + customized > Asterisk dialplans and SIP configuration. > > The best part is that this method could be used for testing any type > of Asterisk PBXs (from embedded to bigger servers), having > capabilities to balance the load to several SIPP call > generators/answer engines in case the tested server have more > processing power than the testing machine. We have use this method to > test 4 machines and the results are for the maximum number of G.711 > ulaw - ulaw SIP calls are summarized in [2]. > > Also, this method is describing how to configure SIPP and Asterisk in > order to test different transcoding scenarios (like ulaw to gsm). > > Basically the controller script increase the number of simultaneous > calls (one SIPP call generator is calling an extension on the tested > Asterisk server and the call is answered by anotther SIPP answer > engine) till one of the load or quality tests failed. > > The tests are: > - load evaluation -> how much time a `sleep 1` command take on the > tested server > - SIP RTT evaluation -> what is the average RTT of a SIP INVITE message > - audio quality evaluation -> based on evaluating of the call > "monitor" file size (on the tested Asterisk server we use an echo > application and the file is recorded on the testing machine) > > Even that the translation service provided free by Google is not the > best way to read our article in English (or other languages) I > encourage you to read it (the pictures and the results are very easy > to understand) and send your feedback or comments here. > > Best regards, > -- > Ioan Indreias > www.modulo.ro > > Notes: > [1] - > http://www.modulo.ro/Modulo/ro/Articole/Determinarea_capacitatii_maxime_a_unei_centrale_Asterisk.html > > [2] Maximum number of G.711 ulaw - ulaw SIP calls >38 - Norhtec MicroClient Jr DX > 130 - VIA EPIA EN12000EG > 176 - Asus Pundit R350 > 320 - Gigabyte 945GCM-S2L -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
> I have read 2 solutions
> (a) Changing the Dial plan and capturing DNID and inserting it into
> one of the existing column in CDR table.
> (b) Copy new CDR related .c & .h files which have added the
> functionality of recording DNID into MySQL.
> For this, CDR table structure needs to be changed and a new field has
> be created in CDR table.
> But I am still not very sure on how to go about doing this.
> Since I only have a production server, I do not have the options of
> experimenting.
> Can someone help with a step-by-step?
> Thx
> Sanjay
>> On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer
>> wrote:
>> Isn't the use of DNID separate to the userfield? I'd like to have this
>> working also.
>>
>> Lee
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
>> Balashov
>> Sent: 15 March 2010 08:34
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
>> (DNID) field into MySQL
>>
>> Use the userfield.
>>
>> On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
>>
>>> Hi,
>>>
>>> I would like to see the DNID in my MySQL CDR logs.
>>>
>>> I have read one big thread in the Asterisk Developer List, but I could
>>> not figure out how to implement it ?
>>> Is there a simple step-by-step.
If this is Asterisk 1.6.*, then you can use the adaptive ODBC, which is
configured using /etc/asterisk/cdr_adaptive_odbc.conf. If you compiled
Asterisk with samples, you will find a sample file that has pretty much
everything that you need. From there, simply set the fieldname that you wish
to write to the CDR, like this:
; Using Adaptive ODBC CDR's, sets the caller ID DNID to the CDR's custom field
named "DNID"
Set(CDR(DNID)=${CALLERID(DNID)})
Personally, I like to set the DNID to a variable, just in case, when the
inbound call first hits Asterisk from the trunk. This probably isn't
necessary, but I am always afraid that the CALLERID(DNID) value will change
with a transfer or a channel redirect, which we use. From there I write the
variable to the CDR.
For more information on the adaptive concept, please see
http://www.asterisk.org/node/48492. There is also more detail from Tilghman
Lesher here:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg210573.html
It's very elegant in it's design and it works like a champ- we use it in
production.
If you are using Asterisk 1.4.*, you can use the the CDR's userfield. This is
an optional, user defined field that can store just about whatever data you
wish depending on the data type defined in the database. You will have to
google around to find out more information on how to enable it, although I
believe that it's an option in the /etc/asterisk/cdr.conf configuration file
that you are using.
Again, if you are using Asterisk 1.6.* I would strongly recommend that you take
advantage of the Adaptive CDR system.
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[asterisk-users] Article - a method on how to evaluate an Asterisk server
Hello all, I would like to share with you an article [1] we have issued last week (sorry, currently only in Romanian language - we plan to provide an English version soon). This article is describing a method to be used for obtaining the maximum number of SIP simultaneous calls an Asterisk server could process safely (meaning no errors/maintain control of the machine and without RTP frame drops) We used SIPP (with modified uas and uac_pcap scenarios) + 2 scripts for controlling the test (one is running on the tested Asterisk server - start-test.sh, for data collection and load analysis and the other is running on the SIPP+Asterisk testing machine, for call quality control and SIPP instance control - sipp-controller.sh) + customized Asterisk dialplans and SIP configuration. The best part is that this method could be used for testing any type of Asterisk PBXs (from embedded to bigger servers), having capabilities to balance the load to several SIPP call generators/answer engines in case the tested server have more processing power than the testing machine. We have use this method to test 4 machines and the results are for the maximum number of G.711 ulaw - ulaw SIP calls are summarized in [2]. Also, this method is describing how to configure SIPP and Asterisk in order to test different transcoding scenarios (like ulaw to gsm). Basically the controller script increase the number of simultaneous calls (one SIPP call generator is calling an extension on the tested Asterisk server and the call is answered by anotther SIPP answer engine) till one of the load or quality tests failed. The tests are: - load evaluation -> how much time a `sleep 1` command take on the tested server - SIP RTT evaluation -> what is the average RTT of a SIP INVITE message - audio quality evaluation -> based on evaluating of the call "monitor" file size (on the tested Asterisk server we use an echo application and the file is recorded on the testing machine) Even that the translation service provided free by Google is not the best way to read our article in English (or other languages) I encourage you to read it (the pictures and the results are very easy to understand) and send your feedback or comments here. Best regards, -- Ioan Indreias www.modulo.ro Notes: [1] - http://www.modulo.ro/Modulo/ro/Articole/Determinarea_capacitatii_maxime_a_unei_centrale_Asterisk.html [2] Maximum number of G.711 ulaw - ulaw SIP calls 38 - Norhtec MicroClient Jr DX 130 - VIA EPIA EN12000EG 176 - Asus Pundit R350 320 - Gigabyte 945GCM-S2L -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to find Asterisk compile time options for building app_swift module
Hi, I have Asterisk 1.6.0.20 running on Red Hat Enterprise Linux Server release 5.4 (Tikanga). I am trying build an app_swift module which uses the Cepstral software. I am compiling it with the following command line gcc -I/opt/swift/include -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -c -o app_swift.o app_swift.c gcc -shared -Xlinker -x -o app_swift.so -L/opt/swift/lib -lswift -lm -lswift -lceplang_en -lceplex_us app_swift.o When I try to load it into Asterisk it gives the following error. [Feb 5 12:12:45] WARNING[32425] loader.c: Module 'app_swift.so' was not compiled with the same compile-time options as this version of Asterisk. [Feb 5 12:12:45] WARNING[32425] loader.c: Module 'app_swift.so' will not be initialized as it may cause instability. [Feb 5 12:12:45] WARNING[32425] loader.c: Module 'app_swift.so' could not be loaded. We are using the yum -install/update to directly install the binaries on our system so I don't have the compile logs to see the flags. The question is how do I find the compile-time options that were used to build the asterisk binaries. Or the other question is, if you have come across this problem building other modules how do you generically solve it. Thanks in advance for any help. -Regards, Irfan Lateef -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing cdr_pgsql on asterisk 1.6.0.26
Hi folks, I am struggling to install cdr_pgsql in asterisk 1.6.0.26. When I do the ./configure, it complains about the function PQescapeStringConn not existing in -lpq, so when I do a make menuconfig, I can't select the cdr_pgsql module. I am using CentOS 5.4 with the yum PGDG repository for 8.4 version. Some of the installed packages are postgresql-libs, compat-postgresql-libs, postgres, postgresql-contrib. Any help would be very appreciated. Thanks, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExtenSpy Problem
David Gibbons wrote: > > Bumping a thread without adding anything useful is annoying. If you do > it again, I won't be helping. > > > Although I have gotten quite a chuckle from your posts, it's really going to > hurt when you fall from that high horse. > > I thought that was a little harsh myself especially seeing as I frequently contribute to solving people's issues in this list, i.e. am a contributing member of this community... Ish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExtenSpy Problem
David Backeberg wrote: > On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik wrote: > >> David Backeberg wrote: >> >>> On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote: >>> You didn't mention version. Could be relevant. >>> > > >> Apologies for not adding the version, it's 1.4.17 >> > > Yeah, that's relevant. > > >> I will try ChanSpy to see what happens and post the results but it >> doesn't really do what I need whereas ExtenSpy does (the functionality >> is required for a call centre to listen in on incoming calls and they >> are not the only people using the asterisk server i.e. hosted VoIP for >> multiple customers and using RealTime to boot). >> > > Do it on a test machine first. ChanSpy on 1.4.17 while running > MixMonitor may crash asterisk. You might want to consider upgrading to > latest 1.4 while you're at it to avoid the possible crash. > I always work on a test machine, we have a production environment, and test environment and a development environment. We're using the latest stable ubuntu version so upgrading is not really an option. There are currently over 1500 sip extension attached to the production server and close to 200 numbers coming in to their own dial plans. Any version upgrade would have to be done by itself outside of any other development to make sure that nothing breaks. > I do realize you want ExtenSpy because that's the way you originally > planned it. I will let you know that ChanSpy works if you can come up > with a clever way to demonstrate which Chan to Spy on. And yes, I use > it in a call center environment, while running MixMonitor, on 1.6.0. > series > > I'm sure I can think up some way to use ChanSpy. I was more hoping that the issue I was having with ExtenSpy was familiar to someone as I can't see that I'm doing anything wrong. I really want to be able to tell that customer to dial 5 or any other arbitrary number and then the extension number they wish to listen in on as that would be a very user friendly system. Thanks Ish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL in 1.6 and Gosub
Am 15.03.2010 13:48, schrieb Kevin P. Fleming:
> Klaus Darilion wrote:
>> Hi!
>>
>> I just updated from 1.4 to 1.6.2.6 and Asterisk complains about my AEL
>> dialplan:
>>
>> application call to Gosub affects flow of control, and needs to
>> be re-written using AEL if, while, goto, etc. keywords instead
>>
>> What is the suggested replacement for an explicit Gosub() call? I use it
>> like this:
>>
>> ...
>> Gosub(blacklist,${exten},1);
>> ...
>>
>> context blacklist {
>> _+43900! => Hangup();
>> _+43910! => Hangup();
>> _+X. => return;
>>
>> }
>
> In 1.6, AEL macro() is implemented using Gosub(), so you can use it as a
> direct replacement. This is listed in the CHANGES file.
Hi Kevin!
I know that AEL macro does not use Macro() anymore, but Gosub(). But
does that imply the other way round too?
Using an AEL macro (which is implemented as Gosub) instead of a Gosub
does not work as the target is a context, not a macro which is
implemented as pseudo context with an 's' extension.
I do not see a way to implement the above dialplan using an AEL macro.
Do I miss something?
regards
Klaus
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Re: [asterisk-users] ExtenSpy Problem
On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik wrote: > David Backeberg wrote: >> On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote: >> You didn't mention version. Could be relevant. > Apologies for not adding the version, it's 1.4.17 Yeah, that's relevant. > I will try ChanSpy to see what happens and post the results but it > doesn't really do what I need whereas ExtenSpy does (the functionality > is required for a call centre to listen in on incoming calls and they > are not the only people using the asterisk server i.e. hosted VoIP for > multiple customers and using RealTime to boot). Do it on a test machine first. ChanSpy on 1.4.17 while running MixMonitor may crash asterisk. You might want to consider upgrading to latest 1.4 while you're at it to avoid the possible crash. I do realize you want ExtenSpy because that's the way you originally planned it. I will let you know that ChanSpy works if you can come up with a clever way to demonstrate which Chan to Spy on. And yes, I use it in a call center environment, while running MixMonitor, on 1.6.0. series -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
- "Mohit Saxena" wrote:
> extension.conf
> exten=07028XX,1,Dial(SIP/PCCW-KPN)
Here is your issue. Shouldn't you be sending the number you'd like to dial with
the call? Try this:
exten => 07028XX,1,Dial(SIP/PCCW-KPN/${EXTEN})
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
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Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
You aren't sending an outbound DID with just SIP/PCCW-KPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Monday, March 15, 2010 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Sip.comf
[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709
extension.conf
exten=07028XX,1,Dial(SIP/PCCW-KPN)
Cisco Gateway:
dial-peer voice 110 voip
description Voip peer to test the server
destination-pattern 1234
session protocol sipv2
session target ipv4:196.3.60.24
session transport udp
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1
hs-redundancy 1 fallback pass-through g711ulaw
clid strip
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh...@starcomms.com
www.starcomms.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, March 15, 2010 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for
inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
> also to some softphones extensions but >when making a call from
> softphone from asterisk box to PSTN, it fails. While I debug on Cisco
> gateway I found >that the To field is SIP header is coming as
> sip:41.205.190.15 which is not correct, instead it should be dialed
> >number:41.205.190.15
>
> Then the problem seems to be between your asterisk box and your
> Cisco.
> Perhaps if you told us what you were trying to SIP dial, we would be
> able to tell us what you did wrong.
>
> > Has any one of you tried using Asterisk in this scenario
>
> yes.
>
> > and also to do LCR and Quality based routing of International
> calls?
>
> I don't know what that means.
>
> > Please let me know if there is any documentation /example of this
> kind available
>
> There is.
> cisco.com
> you pay them, then you can use their documentation.
>
> --
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[asterisk-users] dnd
I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1 second flash on the screen then the phone hangs up. the FOP says it is on DND but some ext are still getting calls. once i do a *76 FOP still says I am on dnd. I am running asterisk 1.6.0.21. before i was getting a message like dnd activated and dnd deactivated. i posted this on the freepbx site and here is what i got "DND should not have seen any change in 2.6.1???" _ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/210850553/direct/01/-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Yes, I mean the same Least Cost routing. Br, Mohit C. Saxena I Data/ISP Department Starcomms Plc. 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 email:moh...@starcomms.com www.starcomms.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Monday, March 15, 2010 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways > and also to do LCR and Quality based routing of International calls? I don't know what that means. LCR = "Least Cost Routing" Routing a call based on the quality or cost of a route (PSTN term vs SIP to PSTN term vs SIP to SIP) is actually quite common. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Sip.comf
[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709
extension.conf
exten=07028XX,1,Dial(SIP/PCCW-KPN)
Cisco Gateway:
dial-peer voice 110 voip
description Voip peer to test the server
destination-pattern 1234
session protocol sipv2
session target ipv4:196.3.60.24
session transport udp
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1
hs-redundancy 1 fallback pass-through g711ulaw
clid strip
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh...@starcomms.com
www.starcomms.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, March 15, 2010 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
> also to some softphones extensions but >when making a call from
> softphone from asterisk box to PSTN, it fails. While I debug on Cisco
> gateway I found >that the To field is SIP header is coming as
> sip:41.205.190.15 which is not correct, instead it should be dialed
> >number:41.205.190.15
>
> Then the problem seems to be between your asterisk box and your
> Cisco.
> Perhaps if you told us what you were trying to SIP dial, we would be
> able to tell us what you did wrong.
>
> > Has any one of you tried using Asterisk in this scenario
>
> yes.
>
> > and also to do LCR and Quality based routing of International
> calls?
>
> I don't know what that means.
>
> > Please let me know if there is any documentation /example of this
> kind available
>
> There is.
> cisco.com
> you pay them, then you can use their documentation.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> 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 --
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>http://www.asterisk.org/hello
>
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in this communication nor for any delay in
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
> and also to do LCR and Quality based routing of International calls? I don't know what that means. LCR = "Least Cost Routing" Routing a call based on the quality or cost of a route (PSTN term vs SIP to PSTN term vs SIP to SIP) is actually quite common. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExtenSpy Problem
Bumping a thread without adding anything useful is annoying. If you do it again, I won't be helping. Although I have gotten quite a chuckle from your posts, it's really going to hurt when you fall from that high horse. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExtenSpy Problem
David Backeberg wrote:
> On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote:
>
>> Hi
>>
>> I'm trying to get ExtenSpy to work but it wont, I'm dialling a number
>> from my mobile which comes into our server and answering the number on a
>> particular SIP extension which all works fine. I'm then dialling an
>> exten from my own SIP extension which executes the ExtenSpy for the
>> correct extension but I hear nothing.
>>
>
> You didn't mention version. Could be relevant.
>
> Bumping a thread without adding anything useful is annoying. If you do
> it again, I won't be helping.
>
> Try turning off the recording before using ExtenSpy().
> You tell us what happens.
>
> Try ChanSpy() instead of ExtenSpy()
>
> Type your version number and your results.
>
>
Hi
Apologies for not adding the version, it's 1.4.17
The results were in the previous email but I'll put them in again. I
only bumped because I originally asked the question at close of play on
Friday and a lot of people don't look at this list over the weekend
(myself being one).
I will try ChanSpy to see what happens and post the results but it
doesn't really do what I need whereas ExtenSpy does (the functionality
is required for a call centre to listen in on incoming calls and they
are not the only people using the asterisk server i.e. hosted VoIP for
multiple customers and using RealTime to boot).
I did have MixMonitor running on the dial plan to start with but I took
it out fearing that it might be interfering with what I was trying to do
but it made no different to the CLI output (other then the MixMonitor
step not being there obviously).
Here is the output in the CLI
-- Executing Goto("SIP/hidden-081aba30", "pack-hidden|s|1")
-- Goto (pack-hidden,s,1)
-- Executing NoOp("SIP/hidden-081aba30", "")
-- Executing Wait("SIP/hidden-081aba30", "2")
-- Executing Set("SIP/hidden-081aba30", "CALLERID(num)=hidden")
-- Executing Set("SIP/hidden-081aba30", "CALLERID(name)=Ish Test")
-- Executing Dial("SIP/hidden-081aba30", "SIP/PACK504|30")
-- Called PACK504
-- SIP/PACK504-081a6b18 is ringing
-- SIP/PACK504-081a6b18 is ringing
-- SIP/PACK504-081a6b18 is ringing
-- SIP/PACK504-081a6b18 answered SIP/213.166.5.133-081aba30
-- Packet2Packet bridging SIP/213.166.5.133-081aba30 and
SIP/PACK504-081a6b18
-- Executing ExtenSpy("SIP/PACK501-081a80a8", "pack...@pack-local|bq")
== Spawn extension (pack-local, 5504, 1) exited non-zero on
'SIP/PACK501-081a80a8'
PACK504 does exist under the pack-local context and I get the same thing
if I leave out the context part. I get the same thing whether I put in
the b option or not and if I don't put in the q option I get the
following. Also, you can see ExtenSpy being executed for the same
extension that has answered the call.
-- Executing ExtenSpy("SIP/PACK501-081acfe0", "PACK503")
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
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Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, 15 Mar 2010, Mohit Saxena wrote: > We are a mobile operator so has to work with the PSTN side E1s from the > Mobile switch. This is the reason for using Cisco Media gateways. I know you may be stuck with them, but you could just as easily plug in a Digium/Sangoma/Rhino T1/E1 card (or Xorcom channel bank?) into your asterisk box and you would be able to accomplish the same thing, but in IMO a much more asterisk-friendly way. Can't help you with the Cisco config... you will need to post a lot more details about your asterisk config if you want help on that side. j > > Kindly help > > Br, > Mohit C. Saxena I Data/ISP Department > Starcomms Plc. > 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 > email:moh...@starcomms.com > www.starcomms.com > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere > Sent: Monday, March 15, 2010 6:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways > > > > On Mon, 15 Mar 2010, David Backeberg wrote: > >> >>> and also to do LCR and Quality based routing of International calls? >> >> I don't know what that means. >> > > Least Cost Routing. Asterisk doesn't have anything built in for this. We > do it with an in-house AGI. Others have done similar things that you > might be able to buy. Try on asterisk-biz. > > The question I have is - why the Cisco? Assuming you have SIP or H.323 > capable phones, just dump the Cisco and use the asterisk box for the whole > shebang. > > j > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
> also to some softphones extensions but >when making a call from
> softphone from asterisk box to PSTN, it fails. While I debug on Cisco
> gateway I found >that the To field is SIP header is coming as
> sip:41.205.190.15 which is not correct, instead it should be dialed
> >number:41.205.190.15
>
> Then the problem seems to be between your asterisk box and your
> Cisco.
> Perhaps if you told us what you were trying to SIP dial, we would be
> able to tell us what you did wrong.
>
> > Has any one of you tried using Asterisk in this scenario
>
> yes.
>
> > and also to do LCR and Quality based routing of International
> calls?
>
> I don't know what that means.
>
> > Please let me know if there is any documentation /example of this
> kind available
>
> There is.
> cisco.com
> you pay them, then you can use their documentation.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> 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 --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
We are a mobile operator so has to work with the PSTN side E1s from the Mobile switch. This is the reason for using Cisco Media gateways. Kindly help Br, Mohit C. Saxena I Data/ISP Department Starcomms Plc. 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 email:moh...@starcomms.com www.starcomms.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Monday, March 15, 2010 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways On Mon, 15 Mar 2010, David Backeberg wrote: > >> and also to do LCR and Quality based routing of International calls? > > I don't know what that means. > Least Cost Routing. Asterisk doesn't have anything built in for this. We do it with an in-house AGI. Others have done similar things that you might be able to buy. Try on asterisk-biz. The question I have is - why the Cisco? Assuming you have SIP or H.323 capable phones, just dump the Cisco and use the asterisk box for the whole shebang. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
The problem is not with cisco as the SIP header on debug doesn't contain the called number. It only says To:sip:ip add of cisco gw. It should say number:ip addr of cisco gw. Br, Mohit C. Saxena I Data/ISP Department Starcomms Plc. 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 email:moh...@starcomms.com www.starcomms.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Monday, March 15, 2010 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena wrote: > I have been trying to do this since 2 days but couldn't make itneed your > help.. Well, you could certainly ask Cisco for help. You did pay Cisco money, right? > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers > I am able to place call from cisco gateway to the asterisk box and also to > some softphones extensions but >when making a call from softphone from > asterisk box to PSTN, it fails. While I debug on Cisco gateway I found >that > the To field is SIP header is coming as sip:41.205.190.15 which is not > correct, instead it should be dialed >number:41.205.190.15 Then the problem seems to be between your asterisk box and your Cisco. Perhaps if you told us what you were trying to SIP dial, we would be able to tell us what you did wrong. > Has any one of you tried using Asterisk in this scenario yes. > and also to do LCR and Quality based routing of International calls? I don't know what that means. > Please let me know if there is any documentation /example of this kind > available There is. cisco.com you pay them, then you can use their documentation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, 15 Mar 2010, David Backeberg wrote: > >> and also to do LCR and Quality based routing of International calls? > > I don't know what that means. > Least Cost Routing. Asterisk doesn't have anything built in for this. We do it with an in-house AGI. Others have done similar things that you might be able to buy. Try on asterisk-biz. The question I have is - why the Cisco? Assuming you have SIP or H.323 capable phones, just dump the Cisco and use the asterisk box for the whole shebang. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExtenSpy Problem
On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote: > Hi > > I'm trying to get ExtenSpy to work but it wont, I'm dialling a number > from my mobile which comes into our server and answering the number on a > particular SIP extension which all works fine. I'm then dialling an > exten from my own SIP extension which executes the ExtenSpy for the > correct extension but I hear nothing. You didn't mention version. Could be relevant. Bumping a thread without adding anything useful is annoying. If you do it again, I won't be helping. Try turning off the recording before using ExtenSpy(). You tell us what happens. Try ChanSpy() instead of ExtenSpy() Type your version number and your results. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena wrote: > I have been trying to do this since 2 days but couldn't make itneed your > help.. Well, you could certainly ask Cisco for help. You did pay Cisco money, right? > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers > I am able to place call from cisco gateway to the asterisk box and also to > some softphones extensions but >when making a call from softphone from > asterisk box to PSTN, it fails. While I debug on Cisco gateway I found >that > the To field is SIP header is coming as sip:41.205.190.15 which is not > correct, instead it should be dialed >number:41.205.190.15 Then the problem seems to be between your asterisk box and your Cisco. Perhaps if you told us what you were trying to SIP dial, we would be able to tell us what you did wrong. > Has any one of you tried using Asterisk in this scenario yes. > and also to do LCR and Quality based routing of International calls? I don't know what that means. > Please let me know if there is any documentation /example of this kind > available There is. cisco.com you pay them, then you can use their documentation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as avaya definity recording server
Oh.. I didnt know that. Thanks Sam > Muro, Sam escribió: > What do you mean chief? What am looking at is ability for asterisk to > receive a call and recording until it tier down without bridging it to the > physical device > > Sam > >> Would you like the advice in all caps? >> >> > He means that you put the subject in all caps. He normally gets upset > with everyone that does this on the subject or in the body. I've > corrected the caps in the subject to avoid further upsetting. > Cheers, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as avaya definity recording server
Muro, Sam escribió: > What do you mean chief? What am looking at is ability for asterisk to > receive a call and recording until it tier down without bridging it to the > physical device > > Sam > >> Would you like the advice in all caps? >> >> He means that you put the subject in all caps. He normally gets upset with everyone that does this on the subject or in the body. I've corrected the caps in the subject to avoid further upsetting. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Android Phones ;-)
Jeff LaCoursiere wrote: > On Mon, 15 Mar 2010, Ishfaq Malik wrote: > > >> Conrad Wood wrote: >> >>> FWIW, just received an android-based phone and after installing >>> "sipdroid" found that it works very well with asterisk ;). >>> >>> It automatically dials numbers through asterisk if available and >>> otherwise through the gsm network. >>> >>> Contacts integrate well too. >>> >>> No ties to any telco or to google, just a happy user ;) >>> >>> >>> Conrad >>> >>> >>> >> I did the same last week and agree totally, a nice little softphone, well >> integrated with the rest of the phone and took about 1 min to configure >> without looking at any instructions. >> >> -- >> Ishfaq Malik >> Software Developer >> PackNet Ltd >> >> > > So which Android phone, and is it using the GSM interface for the SIP > traffic, or only if you are on wifi? > > Does anyone have a wimax android phone yet? > > j > > Mine was HTC Hero, I was using Wifi, the receiving end said the sound was perfect. It has a setting that you can set yourself to allow over GMS or not. I can't really envision myself using it over GMS so I haven't tried but it is able to. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Android Phones ;-)
On Mon, 15 Mar 2010, Ishfaq Malik wrote: > Conrad Wood wrote: >> FWIW, just received an android-based phone and after installing >> "sipdroid" found that it works very well with asterisk ;). >> >> It automatically dials numbers through asterisk if available and >> otherwise through the gsm network. >> >> Contacts integrate well too. >> >> No ties to any telco or to google, just a happy user ;) >> >> >> Conrad >> >> > I did the same last week and agree totally, a nice little softphone, well > integrated with the rest of the phone and took about 1 min to configure > without looking at any instructions. > > -- > Ishfaq Malik > Software Developer > PackNet Ltd > So which Android phone, and is it using the GSM interface for the SIP traffic, or only if you are on wifi? Does anyone have a wimax android phone yet? j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER
What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam > Would you like the advice in all caps? > > On 03/15/2010 01:20 AM, RESEARCH wrote: > >> Hi there >> >> I remember to ask this question in the past but now I have thought of something little bit difference. While I understand that asterisk dialplan >> accept the call to be answered[ Answer() ] in the dialplan, I wanna know if >> this is possible; >> i. A call on legacy PBX, extension to extension is made. >> ii. On call bridging, the legacy PBX initiate a third bridging to the recording system via an ISDN interface. >> iii. Conversation on Legacy continue but asterisk record this call until hangup is issued >> >> Please advice if this is possible. >> >> Sam >> >> > > > -- > Alex Balashov - Principal > Evariste Systems LLC > > Tel: +1 678-954-0670 > Direct : +1 678-954-0671 > Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 68, Issue 33
What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam > Would you like the advice in all caps? > > On 03/15/2010 01:20 AM, RESEARCH wrote: > >> Hi there >> >> I remember to ask this question in the past but now I have thought of >> something little bit difference. While I understand that asterisk >> dialplan >> accept the call to be answered[ Answer() ] in the dialplan, I wanna know >> if >> this is possible; >> i. A call on legacy PBX, extension to extension is made. >> ii. On call bridging, the legacy PBX initiate a third bridging to the >> recording system via an ISDN interface. >> iii. Conversation on Legacy continue but asterisk record this call until >> hangup is issued >> >> Please advice if this is possible. >> >> Sam >> >> > > > -- > Alex Balashov - Principal > Evariste Systems LLC > > Tel: +1 678-954-0670 > Direct : +1 678-954-0671 > Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Android Phones ;-)
Conrad Wood wrote: > FWIW, just received an android-based phone and after installing > "sipdroid" found that it works very well with asterisk ;). > > It automatically dials numbers through asterisk if available and > otherwise through the gsm network. > > Contacts integrate well too. > > No ties to any telco or to google, just a happy user ;) > > > Conrad > > I did the same last week and agree totally, a nice little softphone, well integrated with the rest of the phone and took about 1 min to configure without looking at any instructions. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Android Phones ;-)
FWIW, just received an android-based phone and after installing "sipdroid" found that it works very well with asterisk ;). It automatically dials numbers through asterisk if available and otherwise through the gsm network. Contacts integrate well too. No ties to any telco or to google, just a happy user ;) Conrad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL in 1.6 and Gosub
Klaus Darilion wrote:
> Hi!
>
> I just updated from 1.4 to 1.6.2.6 and Asterisk complains about my AEL
> dialplan:
>
>application call to Gosub affects flow of control, and needs to
>be re-written using AEL if, while, goto, etc. keywords instead
>
> What is the suggested replacement for an explicit Gosub() call? I use it
> like this:
>
>...
>Gosub(blacklist,${exten},1);
>...
>
> context blacklist {
>_+43900! => Hangup();
>_+43910! => Hangup();
>_+X. => return;
>
> }
In 1.6, AEL macro() is implemented using Gosub(), so you can use it as a
direct replacement. This is listed in the CHANGES file.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Check us out at www.digium.com & www.asterisk.org
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[asterisk-users] AEL in 1.6 and Gosub
Hi!
I just updated from 1.4 to 1.6.2.6 and Asterisk complains about my AEL
dialplan:
application call to Gosub affects flow of control, and needs to
be re-written using AEL if, while, goto, etc. keywords instead
What is the suggested replacement for an explicit Gosub() call? I use it
like this:
...
Gosub(blacklist,${exten},1);
...
context blacklist {
_+43900! => Hangup();
_+43910! => Hangup();
_+X. => return;
}
thanks
klaus
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Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
I have read 2 solutions (a) Changing the Dial plan and capturing DNID and inserting it into one of the existing column in CDR table. (b) Copy new CDR related .c & .h files which have added the functionality of recording DNID into MySQL. For this, CDR table structure needs to be changed and a new field has be created in CDR table. But I am still not very sure on how to go about doing this. Since I only have a production server, I do not have the options of experimenting. Can someone help with a step-by-step? Thx Sanjay On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer wrote: > Isn't the use of DNID separate to the userfield? I'd like to have this > working also. > > Lee > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex > Balashov > Sent: 15 March 2010 08:34 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield > (DNID) field into MySQL > > Use the userfield. > > On 03/15/2010 04:25 AM, RSCL Mumbai wrote: > >> Hi, >> >> I would like to see the DNID in my MySQL CDR logs. >> >> I have read one big thread in the Asterisk Developer List, but I could >> not figure out how to implement it ? >> Is there a simple step-by-step. >> >> Thx in advance. >> >> Vai >> > > > -- > Alex Balashov - Principal > Evariste Systems LLC > > Tel : +1 678-954-0670 > Direct : +1 678-954-0671 > Web : http://www.evaristesys.com/ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
Isn't the use of DNID separate to the userfield? I'd like to have this working also. Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: 15 March 2010 08:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL Use the userfield. On 03/15/2010 04:25 AM, RSCL Mumbai wrote: > Hi, > > I would like to see the DNID in my MySQL CDR logs. > > I have read one big thread in the Asterisk Developer List, but I could > not figure out how to implement it ? > Is there a simple step-by-step. > > Thx in advance. > > Vai > -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.2 to 1.6 and bristuff
Am 12.03.2010 13:17, schrieb Steve Davies:
> Hi,
>
> I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap
> :) I was wondering if someone could point me at 3 things that I appear
> to have "lost"?
>
> 1) ZapEC(off) - Is there an equivalent dialplan command to request no
> EC on a channel before dialling in DAHDI?
>
> 2) rxfax(file.tiff) - I have found ReceiveFax(), but I am aware that
> much has happened in the faxing stakes recently, is there a good
> starting point to read about how this works in 1.6 with T.38
> passthru/gateways etc?
AFAIK ReceiveFax does work on T.38 and non-T.38 channels, it is
transparent to the dialplan. ReceiveFax detects if the channels switched
to T.38 and takes care of it.
T.38 passthrough also works.
T.38 Gatewaying is not supported yet. If you need it you can try the
patch from the bugtracker:
https://issues.asterisk.org/view.php?id=13405
> 3) Bristuff came with its own version of PickupChan() - Does anyone
> know if native Asterisk supports their Pickup mechanism?
I needed to get rid of PickupChan and thus use Pickup() and the pseudo
context PICKUPMARK. Works well for me (1.4).
E.g.:
// incoming call
Set(_PICKUPMARK=${EXTEN});
Dial();
//pickup call
Pickup(${ext...@pickupmark);
regards
Klaus
> Or perhaps the Bristuff work has been ported to 1.6.2? I could not
> find it anywhere. Tzafrir used to run a GIT repo of that sort of work,
> but I have not dealt with it for so long I have lost the references to
> it :( If necessary, I can re-do the bristuff code that I need for
> 1.6.2, but thought I'd make sure I was not reinventing the wheel.
>
> Thanks,
> Steve
>
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Re: [asterisk-users] ExtenSpy Problem
Ishfaq Malik wrote:
> Hi
>
> I'm trying to get ExtenSpy to work but it wont, I'm dialling a number
> from my mobile which comes into our server and answering the number on a
> particular SIP extension which all works fine. I'm then dialling an
> exten from my own SIP extension which executes the ExtenSpy for the
> correct extension but I hear nothing.
>
> Here is the output in the CLI
>
> -- Executing Goto("SIP/hidden-081aba30", "pack-hidden|s|1")
> -- Goto (pack-hidden,s,1)
> -- Executing NoOp("SIP/hidden-081aba30", "")
> -- Executing Wait("SIP/hidden-081aba30", "2")
> -- Executing Set("SIP/hidden-081aba30", "CALLERID(num)=hidden")
> -- Executing Set("SIP/hidden-081aba30", "CALLERID(name)=Ish Test")
> -- Executing Dial("SIP/hidden-081aba30", "SIP/PACK504|30")
> -- Called PACK504
> -- SIP/PACK504-081a6b18 is ringing
> -- SIP/PACK504-081a6b18 is ringing
> -- SIP/PACK504-081a6b18 is ringing
> -- SIP/PACK504-081a6b18 answered SIP/213.166.5.133-081aba30
> -- Packet2Packet bridging SIP/213.166.5.133-081aba30 and
> SIP/PACK504-081a6b18
> -- Executing ExtenSpy("SIP/PACK501-081a80a8", "pack...@pack-local|bq")
> == Spawn extension (pack-local, 5504, 1) exited non-zero on
> 'SIP/PACK501-081a80a8'
>
> PACK504 does exist under the pack-local context and I get the same thing
> if I leave out the context part. I get the same thing whether I put in
> the b option or not and if I don't put in the q option I get the
> following. Also, you can see ExtenSpy being executed for the same
> extension that has answered the call.
>
> -- Executing ExtenSpy("SIP/PACK501-081acfe0", "PACK503")
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
> -- Playing 'beep' (language 'en')
>
> Does anyone have any thought/experience of this? Also, if a call is
> already being recorded by MixMonitor, can it also be spied on?
>
> Thanks in advance
>
> Ish
>
Bumping this in the hope that it is seen by a fresh pair of eyes...
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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[asterisk-users] Asterisk to be used with Ciscs media gateways
Hello Guys, I have been trying to do this since 2 days but couldn't make itneed your help.. The scenario is as under: PSTN-Cisco AS5350---Asterisk BoxVoIP Providers I am trying to use SIP on Cisco Gateways and Asterisk box for the connection. The configuration is as under: Sip.comf [PCCW-KPN] type=peer host=41.205.190.15 allow=ulaw qualify=100 nat=no canreinvite=no user=07028000709 extension.conf exten=07028XX,1,Dial(SIP/PCCW-KPN) Cisco Gateway: dial-peer voice 110 voip description Voip peer to test the server destination-pattern 1234 session protocol sipv2 session target ipv4:196.3.60.24 session transport udp incoming called-number .T dtmf-relay rtp-nte codec g711ulaw fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback pass-through g711ulaw clid strip I am able to place call from cisco gateway to the asterisk box and also to some softphones extensions but when making a call from softphone from asterisk box to PSTN, it fails. While I debug on Cisco gateway I found that the To field is SIP header is coming as sip:41.205.190.15 which is not correct, instead it should be dialed number:41.205.190.15 Has any one of you tried using Asterisk in this scenario and also to do LCR and Quality based routing of International calls? Please let me know if there is any documentation /example of this kind available/ Br, Mohit DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
Use the userfield. On 03/15/2010 04:25 AM, RSCL Mumbai wrote: > Hi, > > I would like to see the DNID in my MySQL CDR logs. > > I have read one big thread in the Asterisk Developer List, but I could > not figure out how to implement it ? > Is there a simple step-by-step. > > Thx in advance. > > Vai > -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
Hi, I would like to see the DNID in my MySQL CDR logs. I have read one big thread in the Asterisk Developer List, but I could not figure out how to implement it ? Is there a simple step-by-step. Thx in advance. Vai -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
