similar to: No Caller ID Name PRI NI2.

Problem: Zap Channels Disappear @ random intervals. (Channels have disappeared on both gateways twice this week). My Setup: I have 2 Dell 1850 Power Edge Servers - Configured as. P4 2.8Ghz 512 ECC Memory SCSI Array (2 Drives Mirror) Configuration is really simple the boxes are setup as PSTN termination gateways: (SIP/ZAP) - We are running a SER/Asterisk Hybrid. My asterisk

Hi list, Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we only have the first 6 plugged in right now). Everything works fine until we fail the primary D channel (D's are on 24,48) the secondary then picks up and outbound calls do not work, if we reboot Asterisk the D on 48 comes up and it

---------- Forwarded message ---------- Date: Tue, 4 Jan 2005 17:12:32 -0800 From: Bill D'Anjou <danjou@rocketinternet.net> To: steve@daviesfam.org Subject: FW: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server Can you post this to the group for me? I've joined & I am receiving e-mails from the group but when I try to post I get a bounce saying "waiting for

I am terminating a PRI, NI2 signaling into a PBX (My company's PBX not the PSTN) from an Asterisk server. Caller ID number appears to be transmitted caller id name is not being transmitted is their a compile time flag on libpri that I need to un-comment to enable this feature or is this is unsupported for now? Thanks Chris. -------------- next part -------------- An HTML attachment

Hi, I'm a lab manager / supervisor at our labs. We've had Asterisk in use for over a year directly hooked to the PSTN - a no brainer for configuration (although I had to fix some AT&T specific things in libpri). Right now I have two big challenges. One is to hook our box up lineside to a Lucent Definity G3R. Avaya is chasing this from their end, but we had the highest level of

Hi Guys, I have several * servers connected to T1 PRI's from various service providers in multiple locations the US. All the * servers use the same hardware with the same OS and * version. When connected to 5ESS Switches, using the NI2 (national) PRI protocol, the CID name and number come across fine and populate into the * CDR fine. I connected to a DMS-500, NI2 (national) protocol and

List, I have a TE410P (T1 mode, all PRI) and a T100P (fxoks, for fxs channel bank). I cannot seem to get the T100P to send any clock to the channel bank. I prefer that it use the same clock source as the TE410P, but it doesn't matter if it's not in sync just as long as it's there. The TE410P is configured 3x pri_cpe, 1x pri_net. The three cpe go to XO Sonus switch, the net to

Maintainer: Express Line Date opened: January 17, 2006 Status: Open Value of bounty: $5000.00 Licensing for code: We retain intellectual rights to the underlying source code. We need Asterisk (stable version) to be able to perform a 2B channel transfer for a NI2 B8ZS PRI line. We can't use a channelized T1 at the time for our work. This feature is commonly called a call transfer on analog

I have a question about how Asterisk Parses the Dial Plan. To create a hunt-group which would be the appropriate dial plan: [CompanyABC] exten => 7228888,1,Dial(SIP/8017228888,60,r) exten => 7228888,102,Dial(SIP/8014361234,60,r) exten => 7228888,103,Dial(SIP/8014362345,60,r) exten => 7228888,104,Dial(SIP/8014363456,60,r) exten => 7228888,105,Dial(SIP/8014364567,60,r) exten

We've been working for the past 2 weeks to get a new V400P working with our PRIs from the telephone company. We're trying to get the Asterisk server setup as a VoIP gateway for SIP and AIX. We can make SIP-SIP calls, but all calls from or to the PRI fail. This is the applicable entries from the Asterisk log (configuration files follow) for a call coming from the PSTN on the PRI. I

Is there a way to match the last 7 digits of an extension? So that 1008014445454 8014445454 4445454 Would all match? I have looked at extension matching and I can't figure out how to do this:-) Thanks in advance! Chris -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest

Question: I am PBX multi-hosting several customers on one of my * servers, what the best way to setup call parking to prevent company A from picking up Company B's parked calls ? Any basic examples would be greatly appreciated. Thanks Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm having difficulty getting a PRI working between asterisk and a Merlin Legend. I have been attempting to get it going as esf, b8zs & 5ess (although I have tried other switchtypes...). I found the following info in the Legend docs: If the switch type is 5ESS, the protocol MUST be set for ATT Custom. If switch type is DMS-100, the protocol MUST be set for NTNAPRI The README in

I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server. When I use Asterisk version 10.0.10 everything works perfectly, however when I use 1.2.4 I lose the ability to receive calls from the PSTN. All I get is the following error in my SIP Proxies error logs: SIPSession::proxyResponseImmediately(): Failed to

Hi, I have a client that has a 24 channel voice T1 that I have been using e&m signalling over for a number of years. The local telco finally got an ISDN switch and wants to move them to PRI. I didn't see this as a big problem - I've done a few others on this particular Caribbean island without issues, but this would be the first time with a Xorcom unit involved. We tried to do

I did some searching to find out what I need to do to get CID&NAME inbound on one of our PRI circuits. Right now, we're only getting number (well, asterisk shows the number in both the name and number portions of CID). Is there anything special I need to do with Asterisk to have it accept the name portion of CID along with the number on a PRI that is provisioned NI2? Thanks, John

I currently have two channel groups in my zapata.conf file. I would like one group to be immediate=yes and the other immediate=no Does not seem to matter which way I go, the first entry in overrides my explicit setting for the second group. I am running * 1.0.9 on FC1 [trunkgroups] ;trunkgroup => 1,24 trunkgroup => 1,48,72 ;spanmap => 1,1,0 spanmap => 2,1,0 spanmap => 3,1,1

Hi all, I've got a TE410p connected via T1 to four channel banks (2 FXO and 2 FXS), no PRIs. Some users are complaining that they hear clicks and pops on the FXS lines, generally when they pick up the phone it's noisy. This happens only after a while, e.g. after a fresh restart of everything, all is fine but after some time these noises start appearing. From what I've read on the

I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. Prior to the resets I am getting similar notices to the following chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Telco claims the PRI's are fine on their end and that it is my unit. Is this timing? (google somewhat leads to this) I am running 1.08 asterisk zaptel