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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

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 --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040416/8e1d3036/attachment.htm

...sterisk server for our global help desk. We have put together a comprehensive reporting package for static's from the CDR. I'm not able to calculate the time a call is in the queue before it goes to an agent? I would appreciate help with working this out. Warm Regards and Thanks Shad Mortazavi ------------------------------------------------------ Nexus Global Technical Manager n|m Nexus Management Inc

...g solution and would like to have the ability to initiate a 3 way call based on an incoming call. One party will be an AGI that I have other will be an outbound call via a second T1 interface. Does anyone have a working configuration for an Asterisk initiated 3 way call? Thanks and Regards Shad Mortazavi ------------------------------------------------------ Nexus Group Technical Manager n|m Nexus Management Inc SIP: 0401@pbx.nexusmgmt.com ?

...know that within Astman I can define an extension 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 --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040916/41f5fc67/attachment.htm

...instability around the changes made in 4 and 5. I suspect the problem to be either caused by 4 or 5, in which 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 Netural Bay Sydney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040409/1c4574aa/attachment.htm

...Every 60 seconds 'Your call will be answered as soon as possible' 2. Tell the user how many calls are on the queue. I would then like 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040114/7a3212e1/attachment.htm

...communication. E.g. I will play some hold music, and every 5-6 seconds I drop some packets. I don't have the same issue with SJPhone. I'm 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 Inc Neutral Bay Sydney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040417/864991ef/attachment...

...tions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi

...tself, but what about someone on another system on the internet? Lets say I wanted to call someone at IPTEL.ORG? How do I tell Asterisk to forward calls that are not on it to ser? 2. 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040531/86d2abbc/attachment.htm

...rted in the CLI. In fact the CLI comes back and tells me everything is OK. Is this a known issue? Has someone else run into this? Is there a fix or patch? It looks to me like some type of timing issue with the AGI? Any help would be appreciated. I have a production deadline. Warm Regards 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',...

...ever the number I dial starting with a 9 goes out of the local interface. If I comment out the 'exten =>_9.,1,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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/a0c785fc/attachment.htm

...centers, both centers have a near identical setup. However, when connected to one of my data centers, I call a user, I can see on the CLI that the phone is ringing, but I hear no ringing on my SIP soft phone? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040721/503fcf64/attachment...

...Is there a way that I could configure the system so that the agent is called back. i.e. the agent logs into the system, a call is destined for them and their phone rings. If some one has this setup I would be very interested in hearing from them. Warm Regards and Thanks --------------- Shad Mortazavi US Technical Manager Nexus Management -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031229/6780b08b/attachment.htm

...92437440]: chan_zap.c:5834 zt_pri_error: PRI: Read on 49 failed: Unknown error 500 May 13 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|m Nexus Management Inc Sydney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040513/4b25e010/attachment.htm

I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Netural Bay Sydney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040407/4e1a63e4/attachment.htm

...ating this functionality. Does 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 Manager Nexus Management -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031230/80fd5118/attachment.htm

...be able to share their ideas/code with me. I have written a number of large scale web based administration tools based on Perl and MySQL and would like to release this code to the Asterisk community once it is completed, to act as a call center tool. Warm Regards and Thanks --------------- Shad Mortazavi US Technical Manager Nexus Management -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/981a0c53/attachment.htm

...vices - 1) "DefaultCapture" = "2" My x-ten lite behaves as it does on my XP box. I connect to my Asterisk PBX and I can make calls and receive calls. The problem is there is choppy sound one way :( Is there something I can tweak in wine to cope with this? Warm Regards Shad Mortazavi

...n of Asterisk to enable me to measure the number of minutes a user has been held in a queue. This function was not available in version 0.7.1. I remember a similar problem with version 0.7.2. Anyone else run into the same issue? Is it a known issue/bug? What is the fix? Thanks and Regards Shad Mortazavi ------------------------------------------------------ Nexus Global Technical Manager n|m Nexus Management Inc