similar to: The plain old PBX functionality

Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone. Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone. <-- SIP read from xxx.yyy.142.234:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP

Q: What is the most annoying thing in e-mail? Spam and useless replies when I've already asked for this topic to be closed *sigh*. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Gergo Csibra -->> Sent: 05 December 2008 14:41 -->> To: Asterisk Users

Apart from the "virbiage" FT-201 which hasn't seen the light of day yet, just wondering if there are any other IAX phonesets around available NOW. Don't get me wrong, the Virbiage unit seem to be in the right price range, its just not available yet... Gary .

Hello asterisk-users, My SIP phones are in pickupgroup, and if some of them ringing from other phone can pick up with *8 as usual. But I want to know if this happen. I've tried the a extension, but seems not working. Any other idea? -- Best regards, Gergo mailto:csibra at gmail.com

Hello asterisk-users, There are .call files, with their own syntax, ant they works. But I have a problem. The voip-info.org says: "... If the call answers, connect it here ..." that means, if the called people picks up the phone, he/she hear ringing, until the "caller" picks up the phone. But what can I do, to connect the call before it answers, so the when the called

Hello Asterisk, This is only a test, because I can't start new thread in this list... -- Best regards, Gergo mailto:csibra at gmail.com

Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in

Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap

http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from Thirdlane) looks like a great program for "eye candy configuration" of Asterisk. However it costs lost of $, and I'm currently only an "experimenter" so to speak. Anyone advice of a decent alternative that is similar?? Currently, we only have VOIP connections, but will have a couple of Digium

Hello! My business is a wireless ISP. I would like to offer voice to several business customers. I have a * server, but still need some hardware cards for it. I would like to provide individualized billing to these customers for their usage. Most of these customers have Nortel PBX's. Could I use a Cisco ATA or a similar manufacturer to provide 1-2 line service? I would like to get

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto:

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

http://www.pcphoneline.com/skype "The VPT1000 is NOT a simple last generation USB phone audio device but is rather a next generation integrated gateway and USB phoneset with simultaneous dual mode Skype and SIP calling support. Skype is not forecast to have "SkypeIn" available until June 2005 but you can have the capability now via its built in SIP capabilities." Is this a

Does Asterisk support Radius accounting?.... -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-Users mailing list

Hi, If my memory serves me right, up to Asterisk 1.6, Queue app internals kept the application from working some other apps such as PickUp. I wonder if such things are possible (and if possible, still keep useful Queue Logs ie logs in which picked up or transfered calls are shown as such): 1- a call enters a queue, a phone rings, and a non-Queue member dials some digits and speaks with caller 2-

https://bugs.freedesktop.org/show_bug.cgi?id=91354 Bug ID: 91354 Summary: [Quadro K610M] "xset dpms force on" blinks the screen Product: xorg Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

I have a brand new IAXy I'm playing with. I do not get a dialtone on the phone, or any response at ll on the phone. No sound, no dialing, no ringing. The phone and wire are tested and known to be good. I think I have it setup correctly. When I give the iaxprov command I get this: #iaxyprov 192.168.1.90 iaxy.conf 02: c0 a8 01 5a 05: 11 d9 03: ff ff ff 00 04:

Hello, I am having some troubles with Snom phones and maybe someone can help me. Let me say this: BLF and pickup works great with Polycomes and Grandstream etc... So I think my problem might not be Asterisk related but I am not 100% sure. The snom phones subscribe to my extensions (hint priority) as expected. The light blinks (ringing) or is turned on (in the call) as expected. My problem is to

Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. I did take the power cable from the CDROM to put on the card - I don't need the CDROM right now.. I'm looking for direction in getting this card working - I currently

Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards,