similar to: Zap Channels Disappear???

Okay, now I am really confused. I have two PRI's coming in from two different Carriers (QWEST and ELI), both of them are supposed to be setup to pass name and number on incoming calls. Problem that I am having is that I am not receiving inbound caller id name on either PRI, the only thing that both carriers have in common is that I am terminating into a DMS switch at the carrier.

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 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

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

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

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:

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

I have just built a new system using an HP DL360G7 with a TE420 T1 card, and this is the first system using a generation 7 server. I'm not sure whether that is an issue or not. I am using Asterisk 1.2, and Zaptel 1.4.12.1 with patches for GEN5 of the TE420 card. I have successfully used this combination on several systems based on the DL360G6 and TE420(gen5), which have been in production for

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 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

All, I'm using a Digium TE410P w/ Asterisk 1.2.18. Trying to connect it to our NACT STX switch via PRI, d channel is up, T1 shows normal, but I'm getting crazy errors. I rewired this thing three times, then I connected the same cable from the STX to a Cisco AS5300 (same pri settings as asterisk), and all slip events and frame sync errors went away, so the cable is good. STX --> DSX

# Span 1 span=1,1,0,esf,b8zs bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2 span=2,2,0,esf,b8zs bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Span 3 span=3,3,0,esf,b8zs bchan=49-71 dchan=72 echocanceller=mg2,49-71 # Span 4 span=4,4,0,esf,b8zs bchan=73-95 dchan=96 echocanceller=mg2,73-95 # Global loadzone = us defaultzone = us Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1"

I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72

Hi all I'm trying to configure a TE410P in Europe with three E1s and a T1 channel bank (already bought in the US) to receive and make calls through *. I managed to get the E1s to work, but I'm having trouble with the channel bank (a Rhino). I have tried the bank in spans 1 and 4 changing jumpers in the card (always with no PRI connected, as I have no PRI where I'm testing now,

Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.

Hi, I running into the following problem on my Asterisk setup: --snip-- [Sep 3 01:40:59] NOTICE[9170] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep

When I send a call from my TE410P using E&M, the legacy PBX answers the call but doesn't route it. Any idea what this could be? I assume the digits aren't being delivered properly to the legacy pbx. Any suggestions on what config settings to muck with? Asterisk SVN-branch-1.2-r40901 built by root @ pbx04 on a i686 running Linux on 2007-01-14 14:05:02 UTC zaptel.conf

We have install a DS3 with 28 DS1's we have an Adtran MUX breaking out the DS1's, we are trying to setup the system with 2 dchannels for each 4 DS1's. Everything looks fine when modprobe zaptel and wct4xxp and ztcfg -vvvvvv but when I asterisk asterisk it says: Aug 10 16:33:32 ERROR[8954]: chan_zap.c:6750 mkintf: Channel 24 is reserved for D-channel. Aug 10 16:33:32 ERROR[8954]:

I'm installing TE405P card. This is my zaptel.conf. -------------------------------------- span=1,0,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 loadzone = us defaultzone= us -------------------------------------- When i modprobe wct4xxp, -------------------------------------- PCI: Found IRQ 11 for device 02:0b.0 PCI: Sharing IRQ 11 with