similar to: Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI.

>Also, I'm sure you've probably checked on this one, >but are the phones registered with asterisk? >You can make outbound calls on them without them >actually being registered. I'm assuming you can >still get in and see the CLI. What does "sip show peers" >look like? What does "sip show peer xxx" show? >What does the CLI show when you

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)

Steven Critchfield Wrote: ------------------------ Just a quick thought here, as the vast majority doesn't have access or at the minimal don't use the software you are using to do config and as it is an agi script outside of asterisk, you should go to the vendor of PBXWare and see what they say. /SNIP/ Let me rephrase my question - Has anyone faced any problem with DeadAgi script where

>> We have bought PBXware GUI from Bicom systems and configured >> extensions >> with Polycom Phones as UAs. >> >> The Polycom Phones can dial out and make calls but I cannot make >> extension to extension calling. >> >> Googling did not help much. >> >> As you may be aware PBXware is a closed source software GUI from Bicom >>

Hi, I want to start managing my asterisk boxes with a centralized graphical based interface so I can (due to customers request) give control to customers to add/change extensions to their current PBX intallations such as (not complete list) Add/del/mod extensions sound recordings (ivr or voice attendants) email to fax/ fax to email voicemail to email SIP and ZAP, no IAX needed configure calls

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

Hi asterisk users, I am in need of information about how to configure the sip.conf and extension.conf for subscribers to support the dialog event package rfc 4235. I am using asterisk 1.6.0.1 version. The below are the configuration of sip.conf and extension.conf files which I have done. I have three subscribers as one from my application(App) and other are x-lite1 and

>> Also, I'm sure you've probably checked on this one, >> but are the phones registered with asterisk? >> You can make outbound calls on them without them >> actually being registered. I'm assuming you can >> still get in and see the CLI. What does "sip show peers" >> look like? What does "sip show peer xxx" show? >> What

Seshu, Whats with people who work at Morgan Stanley? You on the one hand bash the opensource software, and then on the other hand a few weeks later, ask for assistance from the open source community for assistance??????????????????????????????????????????? Today, May 11, you request assistance in installing the Areski Calling Card platform, after posting a couple of weeks ago, April 20, this

While we have not been able to get our hands on this model yet (we have the other Linksys models) the Linksys voice products are based on the sipura technologies ATA and will work. I was under the impression that the only shipping WRT54GP2's are provisioned and locked to ATT (for now). Also keep in mind that you must be an ITSP and get authorized by cisco/Linksys to be able to buy all Linksys

Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Why don't u attach the setup page of the phone ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di FaberK Inviato: gioved? 13 ottobre 2005 17.56 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] PA168S/AT320P Right now, but nothing changed. 2005/10/13,

Hi everyone, i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal: http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 ). Does anyone know a better and more documented management interface for * ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

For the benefit of everyone, having H323 Configuration problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

Hi list; I am looking for an application that can be used with call center, in this application we can integrate the telephony part of the call center (like CTI Client ad so on), any one can advise for a good application to be used with Asterisk Call Center? - Note: The application to be customized easy, to be able to use it with Banking, Telecom, Oil, .. etc. Regards Bilal

List, The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk. Seshu Kanuri "G

Hi all, I was wondering if anyone has any recommendations for *.conf generators for Asterisk. Creating *.conf files manually for Asterisk requires too much effort for what I do other than minor tweaking. I run Asterisk as a network appliance (Astlinux on CF) so something like FreePBX that needs to run on the Asterisk server itself is not an option. I have been using IPmanager up until now

Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass,