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Has anyone successfully setup the Avaya IPOffice 500 with a sip trunk to Asterisk. If so can someone give some config examples? Thanks Rick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070825/50bd7d63/attachment.htm

Hi I'm new in this field, have been reading a lot, and have a little question. could it be possible to connect an Avaya IP office pbx to asterisk using a E1/T1/Pri? Original instalation: Telefone company|Pri--->Pri|IP Pffice My Question: Telefone company|Pri --->TDM|Asterisk|TDM --->Pri|IP Office I know that it can be done by using h323, but I need a card on the IPOffice my

Ok, to review, I'm trying to get Avaya 9608G to come up in a pure Asterisk environment-- no Avaya SBC or gateway or any other Avaya gear in sight. I have the phone working to the point where it boots up properly, then displays a Username and Password prompt, and says its extension is 123 and the time is 4:57p, which is wrong. But please don't tell me the only way to program up each

Hi Everyone We would like to connect our Splicecom Maximiser PBX to our Asterisk box via H323 so that we can send our US calls via a low cost carrier (e.g. Broadvoice). Has anyone managed to do this in the past (I remember seeing some companies also worked with this system in the UK). The Maximiser only speaks H323 (not SIP) and can act as an H323 Gatekeeper, so in theory we should just be

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) "[Nov 6 17:14:23] WARNING[6227]:

Hi folks! I bought a X100P a while ago and know I've tried to get it working here at home again ... but I can't manage to get my X-Lite client working with Asterisk (CVS from a day ago) ... I've downloaded the latest version of X-Lite and I believe that I've set it up correctly ;-) But I cant get it to register with my Asterisk - I only get "Login timed out, contact your

Anyone have an H.323 trunk tied between their Avaya and Asterisk box that works? I am having some issues trying to get the two systems to connect. I am using the ooh323 channel to try to make the connection between the two system. I have all my configs if anyone would like to look over them. If I do a trace on Avaya I get a denial event 1191: Network Failure. Thanks! -------------- next part

Hello folks! I have a little problem when I'm trying to set up a small VPN between two hosts. The problem is that when I try to sepcify an IP-adress in /etc/tinc/netname/hosts/hostfile using Adress = xxx.xxx.xxx.xxx syslog says; Invalid variable name on line 2 while reading config file /etc/tinc/boppen/hosts/melc (Line 2 is the Adress = xxx.xxx.xxx.xxx line, and melc is the host I'm

Hi folks! This is sort of OT but I thought maybe someone had a tip for me. What I'm looking for is a tool that I can install on two computer for example, put one on each side of the customers WAN and "try" the connection - simulalate x calls (using codec xxx) and get statistics out of it (delays, jitter, dropped packets and that sort of things). I have looked at ethertap (to debug

Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would

Hi! I have benn playing a little with quesues tonight and I found out if there are at least one member-extension free the announcement with p'the place in the queue wont be played to the person who called in. Is this possible to change so the announcement will be played even if there are free member-extensions? I think that would be nice (well it's not how ACD-groups usually works but

Hi! Me and two friends are trying to get a VPN working, but we cant get routing between two tunnels. This is how it looks, all servers (192.168.*.1) are running IP Masquerade to enable the other computers behind them to access the internet. Both elayne and glenn are connecting to melc, and the tunnel between melc and glenn are running TCPOnly because that glenn doesnt have a public IP (it's

hi, on our incoming E1-PRI from german telco Arcor the leading 0 for the (area access code in europe) and the 00 (country accescode in europe) are missing on incoming callerids. only prepending a single 0 is not the solution as suggested by some writers on this list, because there is no way to differ between national and international callerids and it's not possible to make the decission

Sorry, that name is a misnomer. However, I was excited to find that Ruby has a built in Observable module and I''m pretty bored, so I apologize in advance.... require ''observer'' # one who is observed class Celebrity include Observable attr_accessor :name attr_reader :is def is=(val) @is = val changed notify_observers(self) end end # one who

Hi List, I would like create a H323 trunk from Avaya IPOFFICE to Asterisk, but i would like activate a "direct media path" for the RTP transit directly between the phone and the Asterisk. Now, - H323 Trunk is OK - RTP from the phone transit directly to Asterisk (activate "strictrtp=no" in rtp.conf, and "Allow Direct Media Path" option in Avaya Ipoffice) H323: Phone

I have a client who has a Panasonic Hybrid system. They are taking in another company as a building tenant and the tenant will be on a new 12 station Asterisk system. This new asterisk system will have 4 FXO ports plus ITSP. The two systems will be separate except that they should tie together for the purposes of dialing extensions directly on the opposite phone system and for transferring

I have a few questions. Asterisk is a windows program why each time I try to find out how communicate with my Panasonic TDA 100 or with TDE 100 always read "use one card o use a box" why I can't use simply my network card, in the other side of Panasonic exist two types of cards one in TDA 100 with 2 trunks and in the other side TDE have internal Two trunks too. Why if I want to

Acording to the documentation in gsl "The physical row dimension tda, or trailing dimension, specifies the size of a row of the matrix as laid out in memory" I think that if I pass a matrix to C++ through .C as single ( or.double ), that is, .C ( as.single ( matrix )) then the tda is simply the number of elements of that matrix. Am I right? Thank you.

Hello! Accidentally deletes the file daemon, / usr / lib / samba / classic / smbd. Now I can not start Samba, Samba reinstalling again in the newer version I would lose my settings? It will overwrite existing files or just bring back the excluded? I use a distribution derived from Suse, when I try to install the yast he asks that some libraries are installed: *libstdc + + **libstdc + +-devel

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware