similar to: Configure Max TNT PRI to SIP with Asterisk

Displaying 20 results from an estimated 10000 matches similar to: "Configure Max TNT PRI to SIP with Asterisk"

2006 Nov 07
3
Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
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
2006 Nov 08
1
Re: Asterisk and Max TNT PRI to SIP Authentication Issue
> 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
2007 Jan 27
2
max tnt pri voice channels 56k or 64k, does it matter, selection parameter?
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
2007 Mar 16
0
MAX TNT Question
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
2007 Mar 08
0
Asterisk SIP to MAX TNT Gateway, Sporadic Echo
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
2007 Mar 30
0
Re: Lucent TNT - ring timer
> 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
2010 Jun 24
2
T.38 on a MAX/Lucent/Ascend TNT
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:
2008 Jun 18
0
T.38 Passthru w/ MediaGateway | Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN
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:
2005 Jun 22
1
OT: MAX TNT and PRI calling name (CNAM) facility message
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
2010 Jan 20
1
Using SIPPEER status with CUT function? SOLVED
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'
2007 Jun 27
0
Asterisk to Cisco 2600 GW DTMF Not Working, Working now
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
2005 Jun 15
2
Asterisk and Max TNT
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
2007 Apr 26
1
Re: Voicemail on Different Server, Voicemail with NFS
> -----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
2005 Sep 14
0
MAX PRI for single server (was:Not enoughlinesavailable for Asterisk implemetation)
> -----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)
2006 Dec 05
0
Re: regcontext, NoOp extension vanishes when extension reload, WORKING
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
2006 Apr 18
0
Asterisk Performance 350 Concurrent ChannelsWorking Nicely
Is this with Asterisk in the RTP stream? Is it doing any transcoding? > -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Tuesday, April 18, 2006 9:34 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Asterisk Performance 350 Concurrent > ChannelsWorking Nicely > > > Hi All, > > This is a performance
2008 Jan 29
0
Asterisk and MRTG, a little help please...WORKING
On 1/28/08, JR Richardson <jmr.richardson at gmail.com> wrote: > > You need to take a step back and first test the script without using > > MRTG. Execute it like this: > > # /opt/bin/asterisk-mrtg -h localhost -u XXX -p XXXX -1 SIP -2 Zap > > 10 > > 10 > > 10 > > 10 > > > > You should get 4 lines of numbers. That respresents your SIP
2005 Aug 09
1
inbound caller id name pri - tnt - asterisk
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
2006 Apr 11
0
TNT Max Config
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.
2005 Sep 08
1
MAX PRI for single server (was: Not enoughlinesavailable for Asterisk implemetation)
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