similar to: Fax to Email with Asterisk and Lucent TNT

I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to

Hello folks, I've been trying to get T.38 over SIP working with calls terminated by a MAX/Lucent/Ascent TNT. As far as I can tell, SIP and T.38 are actually working perfectly; however, I can't get the TNT to properly terminate a FAX call. Does anyone have a working configuration for SIP and T.38 for calls from a TNT or APX? Here's a brief description/diagram of my test setup:

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications,

Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the

Hello, I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4. I installed the following -rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0- beta2.tar.gz -rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons-1.4.0- beta1.tar.gz -rw-r--r-- 1 root root 80019 Sep 21 13:25 libpri-1.4.0-beta1.tar.gz -rw-r--r-- 1 root root 1523413 Sep 21 13:25

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

Anyone out there have success getting caller id name from a pri, through a lucent tnt, to asterisk? What about from other media gateways? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050809/d3f02c3d/attachment.htm

> I've got a Lucent TNT that I'm using for a gateway. Its working fine, but I > have one problem. I cannot find any place to set a ring timer, or number of > rings. The calls seem to timeout (Goes to all circuits busy) after about 15 > seconds - which isn't enough time for some voicemail boxes to pickup. I > found a setting called ringing-timer under sip-options, but

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

I'm currently running 1.4b3 with a Digium card and 23 g.729 licenses. Is there a way I can get the g.729 codec work off the CPU and onto a DSP? Any T1/PRI cards with onboard codec DSPs? -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

we're evaluating the use of a Lucent APX8100 E3/SS7 to SIP gateway for use in conjunction with asterisk, serving something like 4000+ lines. does anyone have experience with the APX8100 and it's integration with SIP on asterisk ? does the APX8100 handle SS7<->SIP signalling well enough to be used ? any anecdotes would be well appreciated. -- Regards,

Hey gang, We currently have a class 3 switch (CSX) that..well..it sucks. It does terrible CDR writes, doesn't support LCR, the list goes on and on. We want to replace this with several asterisk boxes each running one or two 4 port PRI cards. The problem is: I can plug in 20 PRI lines into the CSX (from PSTN) and have 1 come from CSX into asterisk. If 1 call comes in on each of the 20 pris,

Is anyone out there using Lucent brand equipment to handle an incomming DS3, converting all 672 calls to SIP (as G729) and sending those to Asterisk/SER over ethernet? If you are and are willing to speak to my boss about your experiences (over the phone) with it, please contact me off list. We have a possible contract with a local CLEC to handle their long distance, and they want to send to

I have officially engaged in a pissing contest with the local Telco over PRI calling name delivery. The telco publishes their calling name delivery over PRI feature as being bellcore gr-1367-core compliant. The gr-1367-core spec states that the calling name is to be included as a facility IE in the setup message, or sent in a subsequent facility IE message with an indicator in the setup message

I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid="MaxTNT" <maxtnt> context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing

I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great !

Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the

Hi All I have posted this problem many times on the list but no reply, trying one more time may be someone will response this time When I call from 1 RTC Client to another without Asterisk everything use to be fine but when asterisk is there as a Registrar a problem use to occur in more than 90% calls, Caller can hear the voice of the receiving side but the receiver cant be able to hear the