similar to: Asterisk Legacy PBX integration and fail-over question,

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

# 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"

Hello. I have a TE212 configured in E1 mode. This is shown in a cat /proc/zaptel/2 and 3 (where the card is configured): cat /proc/zaptel/2 Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 RED NOTOPEN 25 TE2/0/1/1 Clear 26 TE2/0/1/2 Clear 27 TE2/0/1/3 Clear 28 TE2/0/1/4 Clear 29 TE2/0/1/5 Clear 30 TE2/0/1/6 Clear

hi after some testing with A@H, i've decided to install my asterisk server on a slackware (because it's my favourite distro and it is still suggested here http://www.voip-info.org/wiki-Asterisk+Linux+Slackware) , so, i've installed the last 10.2 release, and i've recompiled the 2.6.15.4 kernel. then i've downloaded asterisk1.2.4 and zaptel1.2.4 i load module zaptel and

Hello, I have been struggling with the configuration of this card on my box. I have a T1 line and I am trying to setup asterisk with it. I followed all the instructions and I still see a blinking red light on the card. I use fedora 12. If everything is fine should I see a green line when I plug in the T1 line ? I want to isolate the issue so I di not start asterisk. When I run asterisk I get

Dear All; I am afraid from IRQ misses: 1 The ISDN E1 was working fine on the machine, the electrical disconnected and then the Red Allarm. I checked the dahdi and I found that I have to reinstall dahdi again and I did. But still not becoming UP. The output of the cat /proc/dahdi/1 is following (I am afraid from the IRQ misses: 1, so if it is a problem what is the solution)? [root at CC

I am having problem with T1 configuration. Following r the config.. Zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order span = 1,1,0,esf,b8zs # fxsks=1-24 bchan = 1-23 dchan = 24 # Span 1:

Hi, I've searched and searched on the possible problems. If anyone can help me that would be great. Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) B8ZS/ESF CRC4 error count: 4750 E-bit error count: 5023 Timing slips: 72 1 TE2/0/1/1 Clear (In use) (SWEC: MG2) 2 TE2/0/1/2 Clear (In use) (SWEC: MG2) 3 TE2/0/1/3 Clear (In

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, On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), users are complaining for bad audio. My setup is: PSTN <--E1/PRI ---> Asterisk <--- E1/PRI---> Siemens HiPath <---E1/PRI ---> PSTN asterisk -rx "dahdi show version" DAHDI Version: SVN-trunk-r10414 Echo Canceller: HWEC asterisk -rx "pri show version" libpri version: 1.4.12 A

I've done a decent size update of the OpenSSH portable from the upstream tree. There is an AES upgrade that needs to take place, but I need people to test and tell me what endedness changes need to be applied (if any). Attached is the patch. It is geared towards the latter snapshots..Unsure how it will patch (if at all) against 2.9pX series. Thanks. - Ben "This is dark day in

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

Hi, I am having problems with an Asterisk with a Digium TE205P card. The issue is that the Ring Back Tone is noisy. I am making modem's calls and this noise influences on the initial negotiation protocol, so modems have to recall. My configuration is: Asterisk version: Asterisk 1.4.21.2 Linux version: CentOS release 5.2 (Final) Card: Digium TE205P

I see from the archives that Siggi Langauf was wanting to do exactly what I want to do back in November 2003. Here is what he asked: I would like to do a pilot with some legacy gear, however. Accordingly, I'd like to be able to have * dial 1000X where X is the box that has a new voicemail message and 1001X when the user of mb X deletes the new message(s). The dialing should occur

My Setup looks like this: Mitel 200 SX (1st T1) -------- Bell South (2nd T1) | | | Digium TE110P Asterisk MITEL CONFIGURATION Primary Timing Source: 1st T1 Card Secondary Timing Source: 2nd T1 Card ASTERISK CONFIGURATION span=1,1,0,d4,ami (Look to the Span for timing) We are getting a lot of Frame and Slip errors.... Time Slip Frame 7:00 736 950 8:00 690 1200 9:00 437

Hi list! I managed to come reasonably far (farther than I thought I would) but have two problems. I still need to pass calls to the Legacy PBX for Fax (I need it as a channel bank). I have calls coming in into asterisk, that works fine. Based on the DID I can route calls to the Legacy PBX but I'm puzzled how. I guess I need a new dial command for that? All fax calls are now coming in a

I hope this isn''t too far off topic. I did a little bit of pricing homework a few months ago on new T1 cards. The idea was to build a Linux based router/firewall. After all, Ethernet NICs are easily available for less than $10 today. But all the T1 cards I found cost a fortune. And about a year ago, I learned the list price for a complete Adtran Netvanta 3200 router with T1 card is

I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel:

Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

I posted this earlier on the development list. For those of you who watch both lists, please pardon the duplication. Currently, in our * lab we use all SIP phones so the MWI NOTIFY works perfect. I would like to do a pilot with some legacy gear, however. Accordingly, I'd like to be able to have * dial 1000X where X is the box that has a new voicemail message and 1001X when the user of mb X