similar to: Multiple Digium cards with one NFAS trunkgroup

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

I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get "Ring requested on unconfigured channel". How can I busyout these these channels so that incoming

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been

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

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.

Can anyone describe the asterisk implementation of this any better than the sample config files do? from zapata.conf ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ; group => <trunkgroup>,<dchannel>[,<backup1>...] ; ; trunkgroup is the numerical trunk group to create ; dchannel is the zap channel which will

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

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hi guys, I've been installing and configuring a TE110p card. The compile and install went very well. I'm using this on FC4 and I compile with linux26 as well checked I on the udev configs. zttool and ztcfg both indicate that the card is ready. But when I try to "load chan_zap.so" then I get the following Unable to load module chan_zap.so Jan 5 21:43:33 ERROR[6808]:

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

Hi, I wonder if anyone is using Digium's TE110P card on an E1 connection. I have been try to, but so far it wasn't much of a success. It only works more or less in EuroISDN as PRI CPE. And even that config gives me some trouble with channel negotiation. My current config: *zaptel.conf:* span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=be defaultzone=be *zapata.conf:* [trunkgroups]

I've got a system with 5 pri circuits configured into a CFAS group with a primary and secondary d channel. There are three TE410P cards in the system. The 5 circuit span are located as follows: circuit 1 on span 5 circuit 2 on span 1 circuit 3 on span 6 circuit 4 on span 2 circuit 5 on span 9 primary d chan is on chan 24 of span 5 (chan 120) secondary d chan is on chan 24 of span 1 (chan

While testing a problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon this problem: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1210770512 (LWP 11346)] 0x002e3fe1 in pri_release_timeout (data=0x93a0478) at q931.c:2589 2589 q931.c: No such file or directory. in q931.c q931.c is in libpri, function

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX* A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

Does anyone have a working NFAS config for Zapata and zaptel for 2 NFAS trunks? First two DS1s on tg 1 and other two on tg2? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050825/95961694/attachment.htm

I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. Most calls are fine, all incoming calls are fine, but I am getting echo on a significant number of outgoing calls. The person on the other side hears a perfect call, but the SIPphone side gets to hear themselves. It happens 100% of the time to some numbers (outgoing only), and only sporadically to others. Has anyone

[18884732963 at from-fax-machine:... - your call is hitting the from-fax-machine context - yet your 'fax' exten is in the from-pstn-4 context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax extension" line. When Asterisk detects an incoming fax tone - it tries to automagically route the call to the 'fax' extension in the SAME