[asterisk-users] Linksys SPA941

Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi

[asterisk-users] Changing the Caller ID

? Many Thanks Shad Mortazavi n|m Nexus Management ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

[asterisk-users] Mobile SIP Client

Dear All, I'm looking for a mobile SIP client to use with Asterisk. Has anyone got experience in this area and can you advise me of a product? Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management plc SIP: [EMAIL

[asterisk-users] Mobile SIP Client

Thank you for the information. I'm specifically looking for a Windows 5.0 Mobile SIP agent for a Qtek 9000. Many Thanks Shad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] Problem With Transfering Calls.

it sometimes starts the transfer process and does not acknowledge the call. Can someone please explain what the issue is here and how I can overcome this? Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management plc

[Asterisk-Users] Automatic 3 Way Call

initiated 3 way call? Thanks and Regards Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc SIP: [EMAIL PROTECTED]   ___ --Bandwidth and Colocation provided by Easynews.com

[Asterisk-Users] Automatic 3 Way Call

initiated 3 way call? Thanks and Regards Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc SIP: [EMAIL PROTECTED]   ___ --Bandwidth and Colocation provided by Easynews.com

[Asterisk-Users] New SkypeSIP gateway

Message: 24 Date: Mon, 03 Apr 2006 19:21:57 -0500 From: Michael Graves [EMAIL PROTECTED] Subject: [Asterisk-Users] New SkypeSIP gateway To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain;

RE: Re: [Asterisk-Users] Routing SIP calls via URI

server using IAX2 and then call any external VoIP number. Warm Regards Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc -Original Message- From: Shad Mortazavi Sent: Thursday, March 30, 2006 10:30 AM To: asterisk-users@lists.digium.com

[Asterisk-Users] How do you perform a Variable Substitution In Asterisk

the % with @. Can this be done natively in Asterisk? My production version is Asterisk CVS-v1-0-07. I have read through http://www.voip-info.org/wiki/view/Asterisk+variables and could see no obvious method for this. Many Thanks Shad Mortazavi - Nexus Group Technical

Re: [Asterisk-Users] Routing SIP calls via URI

We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX for Voicemail, 2xxx for Meetme, etc. Do I need to setup a prefix to dial the internet? And then route all calls to the External(*) based on this prefix? Thanks Shad Mortazavi -- Nexus Group

RE: Re: [Asterisk-Users] Routing SIP calls via URI

of this phenomena and how to preserver my URI going form the internal(*) to the external(*). Warm Regards and Thanks Shad Mortazavi --- Nexus Group Technical Manager n|m Nexus Management Inc -Original Message- From: Shad Mortazavi Sent: Thursday, March 30, 2006 10:30 AM

[Asterisk-Users] Routing SIP calls via URI

on our internal server; exten = _sip.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN}) However this does not seem to work? How do I change my dial plan so that SIP calls are routed from my internal Asterisk box to my external Asterisk box over IAX2? Warm Regards and Thanks Shad Mortazavi

[Asterisk-Users] Possible AGI Bug in Asterisk?

Shad Mortazavi #!/usr/bin/perl # CCVR Recording Module Shad Mortazavi December 15, 2005 use Asterisk::AGI; use DBI; #Create a DB-DNS my $dbh = DBI -connect('DBI:mysql:CCVR;192.168.6.56','CCVR_User',''); my $dbh1 = DBI -connect('DBI:mysql:CCVR_ADMIN;192.168.6.56','CCVR_User',''); my $dbh3

[Asterisk-Users] RE: Possible AGI Bug in Asterisk?

Dear All, As a quick update. Switching back to Asterisk CVS-v1-0-02 seems to have fixed the problem. In addition ... The command.. print STDERR Bla Bla Bla :\n; Seems also to be broken in the new version see below; New Version.. -Original Message- From: Shad Mortazavi Sent

[Asterisk-Users] Asterisk and Inter-tel

? Can someone confirm whether or not a T1 from an Inter-tel platform will carry DNIS and/or caller ID.? Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc

[Asterisk-Users] Hold Music is breaking up

'. If I call in over the T1 or using my Softphone there are no such problems. I was wondering if anyone could point me in the right direction. Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc

[Asterisk-Users] Voicemail while in queue.

Dear Group, I have the following requirement; I would like our users to be able to press 0 while they are in a call queue and have the option of leaving a voicemail, also when nobody is logged in, drop directly to a voicemail box. Is this possible? Thanks Shad Mortazavi

[Asterisk-Users] Outbound Mediatrix 1204.

the Asterisk box the call come back! I have seen that several people have this feature working and would be very grateful if you could share your configuration with me or point me in the correct direction. Thanks and Regards Shad Mortazavi -- Nexus Group

[Asterisk-Users] Mediatrix 1204 and Asterisk

on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9

and Regards Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] RE: Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9

exten = 0420,1,Dial(SIP/phone21,20) exten = 0420,2,VoiceMail,u1021 exten = 0420,3,MusicOnHold(default) Removing the tr has done the trick. And the problem is gone. The agent can still transfer the call. Thanks for the idea. Warm Regards Shad Mortazavi

[Asterisk-Users] I don't get a queue_log with my version of asterisk (0.7.1).

Dear All, I would like to add this feature to my version of asterisk. Is there a patch I can apply to get this function? Does anyone have any instructions for this? Thanks Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management

[Asterisk-Users] Calculating the lenght of time in a call queue?

this out. Warm Regards and Thanks Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

[Asterisk-Users] RE:Calculating the lenght of time in a call queue?

I don't get a queue_log file? At what stage was this introduced? Thanks Shad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Asterisk queue_log

. Is this release dependent or is there configuration file I'm missing. Thanks and Regards Shad Mortazavi -- Nexus Global Technical Manager n|m Nexus Management Inc ___ Asterisk-Users mailing

[Asterisk-Users] SIP/SIMPLE, Jabber and Asterisk

Asterisk and Jabber for quiet some time and would love to see these two working with each other. Would welcome any input on this. Shad Mortazavi Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney

[Asterisk-Users] (Another) Queue log analyser

Title: (Another) Queue log analyser Ben, I would definitely have use for this application, fantastic start. When will you be making the source available? In my reports I use the CLID to look at calls for different agents i.e. call volume by agent. Warm Regards Shad Mortazavi

[Asterisk-Users] Limiting use of an account

supply me with an example of the required configuration?   Warm Regards and Thanks Shad Mortazavi - Nexus Technical Manager n|m Nexus Management Inc Sydney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

[Asterisk-Users] Limiting use of an account

Title: Limiting use of an account Thanks for the reply. I'm familiar with the use of contexts for user restrictions. I would really like to know if anyone had written an agi for this. Thanks Shad Mortazavi Nexus Technical Manager n|m Nexus Management Inc

[Asterisk-Users] RE: RE: Creating conference calls from within Astman.

- From: Shad Mortazavi Sent: Friday, September 17, 2004 1:03 PM To: [EMAIL PROTECTED] Subject: Creating conference calls from within Astman. Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay

[Asterisk-Users] RE: Creating conference calls from within Astman.

Title: RE: Creating conference calls from within Astman. Dear All, I sent this question a while back and was wondering if this was possible? Thanks Shad -Original Message- From: Shad Mortazavi Sent: Friday, September 17, 2004 1:03 PM To: [EMAIL PROTECTED] Subject: Creating

[Asterisk-Users] Re: RE: Creating conference calls from within Astman.

Title: Re: RE: Creating conference calls from within Astman. Thanks for this information. Can this be done from within Gastman as well? Warm Regards Shad ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Creating conference calls from within Astman.

and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a number of calls from within the conference room? Thanks for all your help. Shad Mortazavi

[Asterisk-Users] Logging into Multiple Call Queues on two * Servers and Voice Mail option.

Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney

[Asterisk-Users] No Ringing.

? Has anyone had a similar scenario? How as it resolved. Warm Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney

[Asterisk-Users] Asterisk and SER Setup Questions.

that are not on it to ser? How do I append the caller ID so that my calls do not appear to come from Asterisk? Thanks and Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney

[Asterisk-Users] Losing my PRI Interface every 20-30 minutes???

03:27:55 WARNING[1192437440]: chan_zap.c:5834 zt_pri_error: PRI: Read on 49 failed: Unknown error 500 I would appreciate any insights from anyone who may have had and resolved a similar problem. Thanks Shad Mortazavi --- Nexus Technical Manager n

[Asterisk-Users] Problem with x-ten lite

dont have the same issue with SJPhone. Im sure this is a configuration issues, but I can work out where? Can someone point me in the right directions? Thanks and Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management

[Asterisk-Users] SoundPointR IP 300

Title: SoundPointR IP 300 Dear Group, Does any one have experience using SoundPoint(r) IP 300? I have one call center on Snom 200's I'm adding a second and was looking at the SoundPoint, but needed some input. Thanks Shad Mortazavi

[Asterisk-Users] Latency and 'Scratchy' Voice...

is contributing to the 'Scratchy' sound? I have not had this issue in the 4 months of running the product. Warm Regards and Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Neutral Bay Sydney

[Asterisk-Users] Asterisk Server Crashing with New Application

case they will be very easy to rectify. I would however like to know if anyone else has had a) the same experience and b) has been able to isolate the issue. Warm Regards and Thanks Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc

[Asterisk-Users] RE: Asterisk Server Crashing with New Application

, 2004 4:46 AM To: [EMAIL PROTECTED] Cc: Shad Mortazavi Subject: Re: [Asterisk-Users] Asterisk Server Crashing with New Application Shad, I don't remember how far in the past, but a while back at least one person if not more reported instability in asterisk caused by more than one manager

[Asterisk-Users] Presence

. There is always MSN. Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney

[Asterisk-Users] IAX2 Problem and Question

context? Or does the context on each pbx need to be unique. Thanks and Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney NSW 2089

[Asterisk-Users] Extension Questions

,Dial(Zap/g1/${EXTEN:1}) then it works. How can I make the line starting exten = _9001 take precedence over the line starting exten = _9001? Kind Regards Shad Mortazavi - US Technical Manager Nexus Management

[Asterisk-Users] Troubles with the System Attendent Patch.

in exits in var/lib/asterisk/sounds. I'm running Redhat 9.0, Latest CVS, on a Dell 650. I have recompiled the Asterisk program several times and get to the same point. Any help or suggestions would be most welcome. Warm Regards Shad Mortazavi - US Technical

[Asterisk-Users] System Attendent

them put back on hold music. Does someone have a configuration for this or something similar? Your help would be greatly appreciated. Kind Regards Shad Mortazavi US Technical Manager Nexus Management

[Asterisk-Users] Call Queue and Agent Statistics

as a call center tool. Warm Regards and Thanks --- Shad Mortazavi US Technical Manager Nexus Management

[Asterisk-Users] AgentCallbackLogin.

into the directions of 'dial a null extension and press * to logout'. I don't seem to be able to translate this into Syntax. Can some one help? Warm Regards --- Shad Mortazavi US Technical Manager Nexus Management

[Asterisk-Users] Routing calls from a T1 based on DNSI.

anyone have experience in routing calls from a T1 based on a DNSI number? If so would you mind; a) Confirming this functionality and b) giving me a sample of what this would look like in the configuration file? Warm Regards and Thanks --- Shad Mortazavi US Technical

[Asterisk-Users] Agent setup

has this setup I would be very interested in hearing from them. Warm Regards and Thanks --- Shad Mortazavi US Technical Manager Nexus Management