similar to: Asterisk SIP to MAX TNT Gateway, Sporadic Echo

> what is the sip.conf for 1239 > which I'm going to assume is a extension on the TNT > > Barry > > JR Richardson wrote: > > 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 exten 1239 is the CID Number from the

Hi ALL, I'm using this TNT to front-end an asterisk cluster, working pretty well so far. Some T1's are inbound from PSTN PRI's and others are Outbound to PSTN PRI's. Specifying what traffic to send out what PRI is pretty easy, we have unique trunk numbers assigned to specific T1's or groups of T1's, so when I send SIP traffic to the TNT, I prepend the dialed call with

Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the

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

Hi All, Any of you Max TNT Guru's out there have some sample configs for a Max TNT running 11.0.6 code? The example on the wiki was for 10.0 code, it doesn't quite match up with the newer 11.0.6 TOAS release. Any help will be greatly appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

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

Anyone have experience with T.38 passthru in Asterisk 1.4 to a MAX TNT Media Gateway? We're experiencing sporadic results... Topology is described below... Thanks in advance.. -Joe Traditional Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN -------------- next part -------------- An HTML attachment was scrubbed... URL:

> 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

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 am looking for someone who know what they are doing with a TnT MAX to help me get started with configuring the thing. The unit will have 6 PRI's and 18 E&M T1's going into it and sending the calls out VoIP to asterisk boxes and to upstream voip providers. Has 3 x 8T1 cards and 8 x 96 VoIP DSP cards. Willing to pay for your time. Email me at mezzmor at aim dot com.

Does anyone have a MAX/APX with working ingress PRI calling name? I recently acquired a MAX TNT on the cheap and it's integrating fine except for one thing. In the 11.0.0 release notes, it is stated that ISDN calling name will, if present and permitted by presentation flags, be added to the From: and Remote-Party-ID: headers of the INVITE. I'm not able to make this happen. Pcap

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

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

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

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to any of the call control or not. I'd also like to setup a print queue and have outbound

We are in urgent need of some help getting Asterisk to gateway between an incoming H323 connection and SIP to a Lucent TNT. We have the incoming H323 already set up and the SIP going to the TNT but the media stream is getting lost somewhere as no audio is heard. We are willing to pay $$$ for an extra set of eyes to get this resolved fast. It's probably something quick and easy and we are just

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Troy Settle > Sent: Wednesday, September 14, 2005 7:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not > enoughlinesavailable for Asterisk implemetation)

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

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

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'