similar to: T.38 Passthru w/ MediaGateway | Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN

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 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:

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

> 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

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

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

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

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

Hi All, I'm trying to track down an intermittent echo issue. My setup is <phone>sip<asterisk>sip<tnt>pri to carrier less than 10ms latency on the network, 100% SIP, ULAW I have several different phones; cisco, linksys, polycom, snom. It's difficult for me to reproduce the problem regularly so I'm really having trouble isolating anything. I'm wondering if this

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, 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

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 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.

> 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

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.

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

Hi, I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI. Everything works fine with one exception. I seem to be unable to figure out why I cannot hear PSTN intercepted announcement ("number is not in service" etc.) when I'm calling a disconnected number through

https://bugs.freedesktop.org/show_bug.cgi?id=68835 Priority: medium Bug ID: 68835 Assignee: nouveau at lists.freedesktop.org Summary: Unknown opcode 0x43 + protection fault on a Graphics Blaster Riva TNT card QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: