similar to: Asterisk connect to Cisco As5400 gateway

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------

We've got this configuration : Cisco as5400 --- asterisk main server ---- asterisk for cells ---- gsm gateway cisco and the gsm gateway are connected to asterisk via sip, the two asterisk servers are connected via iax. On a succesful call the cisco (not always, 60% of the times) will keep sending a ringtone to the connected phone, even if the call is answered, actually if the user behind

Hi, this is the second time that i post this, may be a wasnt clear the first time. Im having problems with an incoming peer after i upgraded asterisk from 1.0 to 1.2.4, in 1.0 i used to configure the incoming peers like this: register => @prepago-in [prepago-in] type=friend host=192.168.10.102 ; this is the cisco's ip context = from-external dtmfmode=rfc2833 insecure=very ; required for

Hello, I heard that since asterisk 1.6.0.6, now you can send faxes with t.38 through asterisk to a PST gateway that supports t.38 too. Is that true ? If so, what elements you need to make it work beside asterisk and the PSTN trunk ? Thanks all.- -------------- next part -------------- An HTML attachment was scrubbed... URL:

i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part --------------

I'm trying to get Asterisk to pass thru calls using the G729 codec. I've got a 7960 phone and my gateway is an AS5400. I got the following messages when debugging SIP (7778881000 is the 7960): WARNING[1872]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/7778881000-2874(4) to SIP/as5400-35c1(256) WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Hello, Thnks; You are absolutely right.I changed it to point to file:/root/ttylinux/ttylinux-xen , unmounted mnt and had succeeded to create and log in to the new created domain. Silly of me. John >Are you supposed to mount the filesystem under dom0 before you start >ttylinux? >From: Marius Hårstad Kjerkreit <mkjerkreit@gmail.com> >Reply-To: Marius Hårstad Kjerkreit

I've configured SIP softphone to work with asterisk n it's working fine..... but i m unable to connect my netphone IP phone.... I've connected my phone to LAN and assigned an IP address to it.... but how can i make call... plz tell me step wise..... _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today

hi, I am trying to get a webbased IMAP mail module to work. http://nocc.sourceforge.net/ But it seems it doesn't connect to it. When i try to login it says: "login is not allowed for connection" i guess it has to do with the settings of the conf file. I have dovecot installed on Centos and installed it with yum. All (basic)settings are usually set correctly during install.

thanks Billy. I replaced print "STREAM FILE $filename \"\"\n"; with print "EXEC PLAYBACK $filename \n"; and it worked fine. Interestingly when I did print "STREAM FILE beep \"\"\n"; within the script, it worked. If I wasnt a newbie to asterisk I wouldve thought this to be strange. >From: "William Piper"

Hi I am looking for a functional light-weight wiki which has a wysiwyg front-end for a rails application. I have been trying wikiwyg for the front-end, but the proto subclassing is lost on me. Any suggestions? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it''s FREE!

I have an InPlaceEditor, when it is clicked on it loads the unformatted textile text from the server. When i click on menu items, I change the content to the formatted text and i want to change the file the InPlaceEditor will load. I thought a simple loadTextURL: "getPage.php?page=" + currentPage; would do. But currentPage is always empty... It''s baffling :p Is there a way

hi i think my example is very simple and straightforward so i''m not sure if it meets your needs (change revert:true to revert:false from a draggable after I drop it on a droppable so it doesn''t return to its original place.) i did this # 2 divs created here <script language="javascript"> new Draggable(''drag'', {revert: false});

Hello Tom, I''ve been using Shorewall for years without problems. My previous version of shorewall was 1.4.6b-1. Everything worked just fine. Today I upgraded using rpm to 2.0.8-1. After update no one can connect to any interface from net. Server can connect to outside world fine and those described in routestopped have no problem connecting. Any help correcting this problem would be

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,