[asterisk-users] FW: question on how to connect 2 boxes

2009-12-16 Thread B.Masoud @ SH
Was my question not understood? Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: it’s connected to E1, and its purpose to

[asterisk-users] question on how to connect 2 boxes

2009-12-14 Thread B.Masoud @ SH
Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: it’s connected to E1, and its purpose to terminate calls. It will receive SIP

[asterisk-users] Need help with this conf

2009-11-27 Thread B.Masoud @ SH
Hello, I would appreciate if someone can give some help on what I want: When someone call my box (from outside), to a certain ZAP port, it will put him on hold, and immediately the box calls to outside SIP trunk to a preconfigured certain number, then when the other party picks up the phone,

[asterisk-users] IVR for asterisk

2009-11-24 Thread B.Masoud @ SH
Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] max call duration --- SOLVED

2009-11-17 Thread B.Masoud @ SH
PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] max call duration -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 B.Masoud @ SH wrote: How can I set a maximum call duration on a ZAP channel? Look at the parameters on the Dial application. Barry

[asterisk-users] max call duration

2009-11-16 Thread B.Masoud @ SH
How can I set a maximum call duration on a ZAP channel? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] asterisk SIP hangup

2009-11-13 Thread B.Masoud @ SH
Hello all, How can I ask Asterisk to ignore a sip hang-up request for XX seconds from the beginning of the session? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To

[asterisk-users] Termination Question

2009-11-12 Thread B.Masoud @ SH
Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through

Re: [asterisk-users] Termination Question

2009-11-12 Thread B.Masoud @ SH
So how can I let A makes a PEER connection between B C, and ONLY log the call information? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, November 12, 2009 6:10 PM To: Asterisk Users Mailing

Re: [asterisk-users] Termination Question

2009-11-12 Thread B.Masoud @ SH
don't have NAT traversal Kung-Fu, I suggest using IAX2 over SIP. -K - Original Message - From: B.Masoud @ SH mailto:i...@saudihome.com To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Sent: Thursday, November 12, 2009 6:10

[asterisk-users] outbound routing

2009-11-08 Thread B.Masoud @ SH
I have 2 questions: 1. Can I make outbound route rule based on the Source Channel? 2. Can I auto change the outbound route based on time/Day of week? Any help very appreciated.. ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] outbound routing

2009-11-08 Thread B.Masoud @ SH
Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] outbound routing -- Sent from mobile device On Nov 8, 2009, at 2:13 PM, B.Masoud @ SH i...@saudihome.com wrote: I have 2 questions: 1. Can I make outbound route rule based on the Source

Re: [asterisk-users] Dynamic DNS trunk --- SOLVED

2009-11-02 Thread B.Masoud @ SH
dnsmgr.conf: enable=yes refreshinterval=300 regards. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 30, 2009 3:28 AM To: 'Asterisk Users Mailing List - Non

[asterisk-users] Calls disconnects after short time

2009-10-31 Thread B.Masoud @ SH
Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup

Re: [asterisk-users] Calls disconnects after short time

2009-10-31 Thread B.Masoud @ SH
: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My

Re: [asterisk-users] Calls disconnects after short time

2009-10-31 Thread B.Masoud @ SH
...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Sunday, November 01, 2009 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time My server use public ip, so

[asterisk-users] strange dialing HELP !

2009-10-30 Thread B.Masoud @ SH
Hello I just found out this: I had a phone into the FXO ports to see why calls are not passing through, When I ask asterisk to dial a number of 10 digits, it dials the first 9 digits, then wait 2 seconds and dial the last digit! Any idea how to overcome this and dial the whole number 1

Re: [asterisk-users] Dynamic DNS trunk

2009-10-29 Thread B.Masoud @ SH
Discussion Subject: Re: [asterisk-users] Dynamic DNS trunk If the trunk is a dynamic IP you need the other end to register to Asterisk, so letting Asterisk know the new IP. Regards, Juan B.Masoud @ SH wrote: I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show

Re: [asterisk-users] Dynamic DNS trunk

2009-10-29 Thread B.Masoud @ SH
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Friday, October 30, 2009 1:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dynamic DNS trunk On 30/10/09 6:42 AM, B.Masoud @ SH wrote: Hi I tried with registration, it did not update the IP address I

[asterisk-users] Dynamic DNS trunk

2009-10-28 Thread B.Masoud @ SH
I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep

[asterisk-users] hangup from which side

2009-10-22 Thread B.Masoud @ SH
When Asterisk establish a call through an outbound trunk, Is there any way I can know who hang up the call first? The caller or the party called? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

[asterisk-users] polarity on some channels

2009-10-21 Thread B.Masoud @ SH
Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks

Re: [asterisk-users] polarity on some channels

2009-10-21 Thread B.Masoud @ SH
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, October 21, 2009 10:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] polarity on some channels B.Masoud @ SH wrote: Hello, I have : answeronpolarityswitch

Re: [asterisk-users] Cisco 1751 setup with asterisk

2009-10-20 Thread B.Masoud @ SH
I have tried more than 10 different branded/non branded, audiocodes was by far the best fxo device.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Wednesday, October 21, 2009 12:39 AM To:

Re: [asterisk-users] all our circuits are busy now

2009-10-20 Thread B.Masoud @ SH
- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Tuesday, October 20, 2009 4:35 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] all our circuits are busy now On 20/10/09 1:30 PM, B.Masoud @ SH

[asterisk-users] all our circuits are busy now

2009-10-19 Thread B.Masoud @ SH
I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message all our circuits are busy now then few second later, the phone rings, going to the second route! And the

[asterisk-users] ACD ASR

2009-10-14 Thread B.Masoud @ SH
Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source destination? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now:

[asterisk-users] delay to dial

2009-10-10 Thread B.Masoud @ SH
Hello all, Is there anyway that I can configure Asterisk to start dialing out from fxo after (xx) seconds from getting the dial tone? I don't want tdm card to send the number immediately because it fails many times. Thanks for any help. ___ --

Re: [asterisk-users] delay to dial

2009-10-10 Thread B.Masoud @ SH
] On Behalf Of Doug Lytle Sent: Saturday, October 10, 2009 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] delay to dial B.Masoud @ SH wrote: Hello all, Is there anyway that I can configure Asterisk to start dialing out from fxo after (xx) seconds from

Re: [asterisk-users] delay to dial

2009-10-10 Thread B.Masoud @ SH
I have done the changes exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}}) I am getting this: -- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592, DAHDI/r0/0559857826|300|) in new stack -- Called r0/0559857826 Is it now on work? Or I have to restart? Thanks. -Original

Re: [asterisk-users] delay to dial

2009-10-10 Thread B.Masoud @ SH
. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk Sent: Saturday, October 10, 2009 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] delay to dial B.Masoud

[asterisk-users] calls ansowered for 1 second or less

2009-10-09 Thread B.Masoud @ SH
Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, it’s just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : vendor SIP--à Asterisk ßIAX2---à Asterisk with

[asterisk-users] choppy sound

2009-10-09 Thread B.Masoud @ SH
Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 -

Re: [asterisk-users] choppy sound

2009-10-09 Thread B.Masoud @ SH
a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject

Re: [asterisk-users] choppy sound

2009-10-09 Thread B.Masoud @ SH
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart

[asterisk-users] limiting number of channels to be accessed

2009-10-08 Thread B.Masoud @ SH
Hello all, Assuming I have 1 asterisk with 24 channels fxo and another 2 asterisk boxes all connected iax2, I want to grand the first asterisk box to use all the 24 channels on the main, but I want the 2nd asterisk to use only 8 port, how can limit the second box from receiving more than

Re: [asterisk-users] Asterisk debug message --- stopped sounds ???

2009-10-08 Thread B.Masoud @ SH
Anyone pls I have seen this message stopped sounds while I am watching asterisk debug: -- Called 9/0532828384 -- Call accepted by 192.168.10.220 (format ulaw) -- Format for call is ulaw -- IAX2/9-69 stopped sounds -- IAX2/9-69 answered

Re: [asterisk-users] limiting number of channels to be accessed

2009-10-08 Thread B.Masoud @ SH
, October 08, 2009 12:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] limiting number of channels to be accessed B.Masoud @ SH wrote: I want to grand the first asterisk box to use all the 24 channels on the main, but I want the 2^nd asterisk to use only 8

Re: [asterisk-users] limiting number of channels to be accessed

2009-10-08 Thread B.Masoud @ SH
To: Asterisk Users List Subject: Re: [asterisk-users] limiting number of channels to be accessed /etc/asterisk/extensions.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Thursday, October 08

Re: [asterisk-users] tdm outgoing

2009-10-07 Thread B.Masoud @ SH
, October 05, 2009 10:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing B.Masoud @ SH schrieb: I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I

[asterisk-users] Asterisk debug message --- stopped sounds ???

2009-10-07 Thread B.Masoud @ SH
I have seen this message stopped sounds while I am watching asterisk debug: -- Called 9/0532828384 -- Call accepted by 192.168.10.220 (format ulaw) -- Format for call is ulaw -- IAX2/9-69 stopped sounds -- IAX2/9-69 answered SIP/xxx.xxx.xxx.xxx-b7d009a0 What does it

Re: [asterisk-users] Networking Concept

2009-10-06 Thread B.Masoud @ SH
- Non-Commercial Discussion Subject: Re: [asterisk-users] Networking Concept B.Masoud @ SH wrote: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling

Re: [asterisk-users] Networking Concept

2009-10-06 Thread B.Masoud @ SH
- Original Message - From: B.Masoud @ SH mailto:i...@saudihome.com To: 'Asterisk Users Mailing List - Non-Commercial mailto:asterisk-users@lists.digium.com Discussion' Sent: Tuesday, October 06, 2009 1:14 AM Subject: [asterisk-users] Networking Concept Hello, I would like

Re: [asterisk-users] tdm outgoing

2009-10-05 Thread B.Masoud @ SH
send a call to the lines attached to the card? PaulH B.Masoud @ SH wrote: Hi I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls to that trunk, I am getting all circuits are busy now, do I have to do something specific?? I am using elastix. Thanks

Re: [asterisk-users] tdm outgoing

2009-10-05 Thread B.Masoud @ SH
Are you series??? My card is FXO TDM2400, I am sure its designed to forward calls to pstn!!! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: Monday, October 05, 2009 5:07 AM To: Asterisk Users

Re: [asterisk-users] tdm outgoing

2009-10-05 Thread B.Masoud @ SH
Man, thanks a lot! I just changed the name to g0 instead of DGTDM24 and it worked!! I would like to know where I can set the configuration for line tones( dial tone, call and busy tone) and where I can change different setting for polarity / current disconnect etc.. of the line? I cant find

Re: [asterisk-users] tdm outgoing

2009-10-05 Thread B.Masoud @ SH
Thanks, I made the zone, and now call disconnect works ok! i have one last problem, I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I balance the calls over all ports??? Any suggestions appreciated.

[asterisk-users] Networking Concept

2009-10-05 Thread B.Masoud @ SH
Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in

[asterisk-users] tdm outgoing

2009-10-04 Thread B.Masoud @ SH
Hi I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls to that trunk, I am getting all circuits are busy now, do I have to do something specific?? I am using elastix. Thanks. ___ -- Bandwidth and Colocation Provided by

[asterisk-users] Trunk Sequence

2009-09-15 Thread B.Masoud @ SH
I have added 2 trunk sequence in my outbound routes, The problem is that: 1. If the call was busy on the first trunk it will go to the second (i.e. the called party hung-up without answering the call) How to overcome this??? ___ --

Re: [asterisk-users] RESET CDR

2009-09-10 Thread B.Masoud @ SH
-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR B.Masoud @ SH wrote: Yes that is the problem. So what do you do when the line doesn't support polarity?? What is the best solution in this case? What kind of gateway do you use to connect to the PSTN? -- Iván Stepaniuk Alba Fotónica S.L

[asterisk-users] ASR ACD

2009-09-10 Thread B.Masoud @ SH
Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now:

Re: [asterisk-users] ASR ACD

2009-09-10 Thread B.Masoud @ SH
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Thursday, September 10, 2009 3:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ASR ACD Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments

Re: [asterisk-users] RESET CDR

2009-09-09 Thread B.Masoud @ SH
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Wednesday, September 09, 2009 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR On 9/09/09 5:14 PM, B.Masoud @ SH wrote: A little more help is appreciated, I know about

Re: [asterisk-users] RESET CDR

2009-09-09 Thread B.Masoud @ SH
Can you provide me some code for that? I am NOOB -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Wednesday, September 09, 2009 5:31 PM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] RESET CDR

2009-09-09 Thread B.Masoud @ SH
Yes that is the problem. So what do you do when the line doesn't support polarity?? What is the best solution in this case? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk Sent:

[asterisk-users] RESET CDR

2009-09-08 Thread B.Masoud @ SH
Hello, How can I reset CDR time , let's say after 30 seconds of answer signal, reset CDR to 0 , any idea ?? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona

Re: [asterisk-users] RESET CDR

2009-09-08 Thread B.Masoud @ SH
] On Behalf Of Matt Riddell Sent: Wednesday, September 09, 2009 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR On 9/09/09 4:34 PM, B.Masoud @ SH wrote: Hello, How can I reset CDR time , let's say after 30 seconds of answer signal, reset CDR