similar to: three way calling and cisco ata 186

Hi, Does anybody have ATA 188 working with any kind of fax machine? I've tried many different configuration following the Cisco Online Manual and I couldn't get this working with Asterisk. I were trying do change the ATA Connect Mode and Audio Mode reading the (http://www.cisco.com/en/US/products/hw/gatecont/ps514/ products_configuration_example09186a00800d698e.shtml) and allowing

I see that there's now a 2.16.1 upgrade path for Cisco ATA-186 devices, dated (variously) July 11 or July 14 2003. Here are some interesting bugs that claim to be fixed. Most notable is CSCeb17953, at least from my perspective, as I've hit this bug before. CSCea42480 The Cisco ATA ignores the Require:100rel header and processes call. CSCea69889 The Cisco ATA builds a 302 Moved

Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the

ðÏÞÅÍÕ ÂÙ É ÎÅÔ? ÷ÏÐÒÏÓ ÔÏÌØËÏ × ÔÏÍ, ËÁË Õ ÔÅÂÑ ÂÕÄÅÔ ÓÏÅÄÉÎÑÔØÓÑ ÐÁÎÁÓÏÎÉË Ó ÁÓÔÅÒÉÓËÏÍ. Date: Wed, 30 Jul 2003 20:06:17 +0400 From: Pavel Zheltouhov <pavlo@comlink.ru> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk,ata186 and Panasonic TD1232 Reply-To: asterisk-users@lists.digium.com I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk.

I strange error messages when using mgcp and ata186 . This session is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/1@10.0.1.19 Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/1@10.0.1.19-1' observed '0' -- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode: sendrecv Posting Request: RQNT 306

hi guys, i'm in need of several 24port or higher fxs device which supports sip, aside from mediatrix and audiocodes (cisco's vg248 doesn't support sip), do you have any idea who else manufactures such device? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030801/67eb12dd/attachment.htm

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

Newbie to Asterisk I've been looking around for a little while, can't seem to find some sample configs for using a Cisco 7206 as a gateway. The below link is an initial plan of an Asterisk solution that may replace our Cisco Call Manager 3.1/ IPCC / IVR setup. We currently have all of the hardware below. Just take a peak and see if there is anything that is off base. I don't know

I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk. Can I dial from asterisk into ata, then indicate phone number playing tone (use DISA feature at panasonic) and connect to any analog phone connected to panasonic ? I think some of Playtones application within Dial application can help me. But I don't know how. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia

I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? Thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Hi, I'm having trouble getting caller*id to appear on my phone connected to an ATA186, and being called from Asterisk. Does anyone out there successfully see callerid on their ata186-connected phone? The "From:" header in the INVITE to the ATA seems to have the "right stuff" - eg From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061 But

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

I am trying to do a "factory reset" of an ATA186 using the widely-available instructions (basically dialing "FACTRESET#" on the keypad while at the menu prompt). I have done this a number of times before with success, but on this unit the lady spells out "P A S S W D" when I finish up the entry. Does anyone know what to do next? Hitting the star key (which is

Hi all, I have a Cisco ATA186 connected to an Asterisk Server (SIP) The dialplan uses 1XX for local extensions and XXXXXXX for external numbers, where the first digit is always different than 1. In this moment, when I dial 123 for example, ATA waits till timeout before dialing that number. The same for the longer one. How can I do to make it dial imediately when 3 digits starting with 1 are

I'm trying to register with net2phone. I've already changed chan_sip.c, User-Agent: string to say "User-Agent: Cisco ATA 186 v2.16 ata18x (030401a)". But still I'm getting the error msg. Here is the debug msg: IP Address is xxx.xxx.xxx.xxx 11 headers, 0 lines Reliably Transmitting: REGISTER sip:66.33.146.12 SIP/2.0 Via: SIP/2.0/UDP

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

Hi guys! Maybe I found a very annoying bug: by holding the ALT key and clicking (anywhere) with the right mouse button, Compiz crashes. I found this bug (or a **very**strange behavior?) by accident... ;) I tried many times, and always Compiz to crashed. Here's my machine configuration: [start] OS: Ubuntu 6.10 "Edgy Eft" GNOME: v2.16.x Compiz: v0.3.6 (repository gandalfn, with

Hi, I have one X100P card connected to the PSTN line and two analog phones connected to an ATA v2.16, SIP mode). When a call is received from PSTN, after about 5-10' the call is closed without apparently any reason. I have increased the value busycount to 10 and the problem persist. A pure IP call (local or even remote) is never closed by himself. There is something else to check? Thanks,