similar to: Avaya UUI

Can you put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards ---------------------- > >jean-marie.goupil@telintrans.fr wrote: >> OK, so I'll do that... Is there any infos I need to know

Hi, I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using chan_ooh323 from asterisk-addons. I am able to make a call from SIP Phone -> Asterisk -> Avaya -> Station (phone) and vice versa. I am also able to make a call from SIP Phone -> Asterisk -> Avaya -> PSTN. However I face problems when I make DID calls from the PSTN. The DID calls are made through

On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi <shinacalypse at gmail.com>wrote: > Hi! > I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide > a conference bridge for an existing Avaya PBX. I have no control over the > Avaya system, but I am able to speak with the admin in charge when I need > stuff done. I am running all this in a VirtualBox

Dear all, How do I set the UUI informations for outgoing calls and read the UUI information for incoming call in asterisk? Thanks in advance.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100531/ffcceeee/attachment.htm

Hi, I came thru ISDN UUI (User-User Information) protocol which is defined in this RFC - http://www.ietf.org/id/draft-ietf-cuss-sip-uui-17.txt But I don't understand how to use this with Asterisk. Any idea would be much appreciated. Thanks. Gopal. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

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

I just connected an Asterisk server to an Avaya Pbx using the instructions at: http://www.voip-info.org/wiki/index.php?page=Asterisk +Avaya Everything seems to be working as I can send and receive calls. The only detail I am having a problem with is that when an extension on the Avaya calls an Asterisk extension, I only get the name part of the CID, not the number. From Asterisk to the Avaya

Hi everyone, I was wondering if anyone on this group has successfully integrated Avaya 8300 or 8700 and Asterisk using H.323 trunk and would be willing to share configurations and/or comment on the voice quality achieved. Currently we have Avaya 8300 integrated with Asterisk over a Q.SIG trunk, but we need to put Asterisk in a different geographical location from the PBX and need to explore

Hi there, Is it possible to manage UUI with asterisk and ISDN (T0 Fritz card). Basically, is it possible to send User to User Information using the D-channel, while making a call?

OK, so I'll do that... Is there any infos I need to know about chan_sip.c (because I suppose it's it that I need to play with)? Does anyone know an interesting website where I can find infos about UUI in ISDN (with CAPI maybe?) ? Thanks for your help.

I'd like to start configuring my Avaya 9608G phones for use on Asterisk / FreePBX / PBX-In-a-Flash. I'm using a variety of other phones on my system without major issues. I've read the discussion back in March, May and August of 2016, but unfortunately, my difficulty is much more basic. I think it has to do with DHCP, specifically, what options I'm offering the phone via DHCP.

Right-- I've seen the Avaya document you cite below. It says "To administer DHCP option 242, make a copy of an existing option 176" but I don't have any example of option 176 or 242 to copy, and don't know what to do to /etc/dhcpd.conf to make it offer option 242. Then there's this long table of parameters to use with (presumably) option 242. I was hoping someone had a

I have added H.323 support for ASTERISK PBX using the OpenH323 library. I thought that since the Avaya IP phones are H.323 phones I will be able to make them register to Asterisk server. Does anyone know if this is possible ? Katerina

Hi I have a client using a Avaya IP Office PBX that is taking a SIP trunk from me terminating on a * box. It all works perfectly apart from DTMF. Although you can hear the tones they don't seem to get recognised. I have tried DTMF mode auto, inband, out of band and rfc2833 but no luck. Any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK

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

The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether it's a SIP phone or

May be force be with you hello, We want to make H323 calls beteen asterisk and avaya(lucent) pbx. We create node-name,H.323 signaling group,trunk, but we can not make H.323 calls to asterisk. Also no warnings exist in debug. Instead of giving the IP of Asterisk ,i give my computer's IP and run SJPhone ith H.323 GUI. In this time, connection is established. SJPhone accepts H323 calls but

Hello, We want to make H323 calls between asterisk and avaya(lucent) pbx. We create node-name,H.323 signaling group,trunk, but we can not make H.323 calls to asterisk. Also no warnings exist in debug. Instead of giving the IP of Asterisk ,i give my computer's IP and run SJPhone ith H.323 GUI. In this time, connection is established. SJPhone accepts H323 calls but Asterisk does not. Do

Thanks for the reply John. About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department. I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip