similar to: Re: Lucent TNT - ring timer

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

> 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

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, 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

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

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 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

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, 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

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

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

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'

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

> -----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)

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

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

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

OK this was an easy one to fix. All I had to do is RTFM. Note on the wiki: ATTENTION: Make sure you take a look at bug report 7144 Just do what Kevin said, include the regcontext in whatever static context you have the priority 2 extension and don't make a static regcontext in extension.conf. Let sip module do the rest. Works great. Thanks Guys. JR On 12/5/06, JR Richardson