search for: pa168

Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget...

...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, Kanuri, Seshu (Company IT) <Seshu.Kanuri@morganstanley.com>: > have you configured the STUN server on the phone to any one of the > available stun servers like stun.xten.net? > > > -----Original Message----- > From: aster...

For those of you who may be interest.... IAX2 loads are now available for the standard builds... http://www.aredfox.com/edownloadsiax2.htm Just a word of caution... Remember to change the ports over to 4569 from whatever. And don't forget to grab the palmtool from http://www.aredfox.com/download/tools/PalmTool.zip My own testing of IAX2 with both a phone and an ATA is that IAX2 is

Are there any IAX speaking "hardphones" out there? If so, can anyone offer comment on their quality? Thanks! -Dorn

I purchased an IAX2 hardphone, X100 otherwise known as a Netweb X100 or YWH100 with a PA168 chip and the latest firmware 1.45 available, from a US retailer. I was able to configure the phone to work with my Asterisk box, except the hold and transfer buttons do not work. When you press the hold button, it rings endlessly, the transfer button, displays "transferring" but it does...

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...d point build, but this > is > >> really impressive, almost too good to be true. This is all straight > >> compiled C, after all (though the compiler has been updated also). Judging from these numbers it looks within the bare realms of feasibility to get speex working on the PA168, which includes a ADSP2181. This is a popular chip for voip applications (see http://www.voip-info.org/wiki/view/PA168for more details) Has anyone explored this chip? -- Mike Taht PostCards From the Bleeding Edge http://the-edge.blogspot.com -------------- next part -------------- An HTML att...

...phone 10916: behind a router, public IP = 219.xx.xx.9 phone 10920: behind a router, public IP = 218.xx.xx.24 *************************************************** Sip read: REGISTER sip:60.xx.xx.164 SIP/2.0 Via: SIP/2.0/UDP 219.xx.xx.9:5060;branch=z9hG4bKVfTeS1TbJ Max-Forwards: 70 User-Agent: PA168S From: "10916" <sip:10916@60.xx.xx.164>;tag=ks8tHkBudRSI7Ydz To: "10916" <sip:10916@60.xx.xx.164> Call-ID: l50TKpxGbtLGYIvi@218.xx.xx.24 CSeq: 25463 REGISTER Contact: <sip:10916@219.xx.xx.9:5060> Expires: 60 Content-Length: 0 11 headers, 0 lines Using latest r...

Hullo :) I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from sipgate.co.uk to any other extension. My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind transfer, simply dial the number you want to transfer to, and press 'FWD'... This is what happens when I start the sip debug after the initial call setup... 01618313800 is the callerID of the person making the call, 1301 is the interna...

I'm looking for an ATA\Voice Gateway that runs IAX and has several ports (8 would be nice). I am looking to avoid devices that use the same firmware as the ATCOM devices as I found them to be buggy (and a PITA to find the proper update). ---------- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed...

...terisk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk 1.0.8 *CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new stack -- Called 2391 -- SIP/2391-79a0 is ringing -- Saved useragent "PA168S" for peer 2319 -- SIP/2391-79a0 answered SIP/2339-4da6 -- Attempting native bridge of SIP/2339-4da6 and SIP/2391-79a0 -- Started music on hold, class 'default', on SIP/2339-4da6 -- Stopped music on hold on SIP/2339-4da6 == Spawn extension (local, 2391, 1) exited non-...

Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of asterisk@home installation. I have tried to use the oh323.conf content listed on WIKI but it is just not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot register. I need a working example of this file for similar phone. Seshu -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.

Message: 1 Date: Mon, 14 Feb 2005 09:53:36 +1100 From: "PHP Mechanic" <oliver.bode@phpmechanic.com> Subject: [Asterisk-Users] Who makes these phones? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <08d401c5121e$dbea4750$0200a8c0@oliver> Content-Type: text/plain; format=flowed;

...risk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk 1.0.8 *CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new stack -- Called 2391 -- SIP/2391-79a0 is ringing -- Saved useragent "PA168S" for peer 2319 -- SIP/2391-79a0 answered SIP/2339-4da6 -- Attempting native bridge of SIP/2339-4da6 and SIP/2391-79a0 -- Started music on hold, class 'default', on SIP/2339-4da6 -- Stopped music on hold on SIP/2339-4da6 == Spawn extension (local, 2391, 1) exited non-...

Back to the ACTUAL TOPIC of this thread... This phone looks kinda nice, where can one get hold of it? How about it's * compatibility? I realize that it says it does things like 3-way conference and attended transfers, but how about in *? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

John Novack wrote: >And the only IAX2 box made is the Digium one, with it's current shortcomings ? >From reading through the archives, it seems there is currently no way to reset to factory default, no >written MAC address on an individual box, and some other instabilities requiring frequent resets. Yeah, there's the rub. Dunno if it's worth it, I'm willing to give it

Are their any good chooses for IAX Adapters? -Thanks

I have been hearing a lot about the new Gnet SIP phones. Is anyone using them? How do they perform? Sean

...terisk/mohmp3. When i set a call on hold everythinghs seem ok, but i cannot hear music. I'm using asterisk 1.0.8 *CLI> -- Executing Dial("SIP/2339-4da6", "SIP/2391|60|Thtr") in new stack -- Called 2391 -- SIP/2391-79a0 is ringing -- Saved useragent "PA168S" for peer 2319 -- SIP/2391-79a0 answered SIP/2339-4da6 -- Attempting native bridge of SIP/2339-4da6 and SIP/2391-79a0 -- Started music on hold, class 'default', on SIP/2339-4da6 -- Stopped music on hold on SIP/2339-4da6 == Spawn extension (local, 2391, 1) exited non-...

Hi, I'm using asterisk 1.0.6 and I would let media path be connected directly between the phones without going through Asterisk. I have to it with an AtCom320 (with pa168s chip). I just saw and tryied to do what this page http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%2 0clients%20connect%20directly says. Before going on (with sniffer eth traffic between * and two phones) I'd like to known if it can works. Does anyone just did it? T...