similar to: configuration asterisk zap module

hello all, i'm using TE110P and all configuration prosess is done. but after i do the 'modprobe' for zaptel and wcte11xp, follow by ztcfg -vv and asterisk -vvvgc i have this error. *CLI> May 27 14:44:33 WARNING[3059]: chan_zap.c:8534 pri_dchannel: PRI Error: We think we're the network, but they think they're the network, too. May 27 14:44:34 WARNING[3059]: chan_zap.c:8534

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

hi all, i want to ask about caller id in asterisk. if i make outgoing call, people who receive my call can see the pilot number. what i must do if i want that people receive my other number (DID number)? is it posibble? thank to everybody.. regards, shahdan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

I figured it out. The old system (Televantage 3 and 4 I think) has limited specifications on the T1. After setting up the system, I was able to send and recieve calls. I still have some work to do like figuring out faxing and a floating receptionist, but this is a nice start. ----------------------------------------------------Televantage T1 Requirements: Framing: D4 Superframe or Extended

I can't seem to get any ZAP trunks on my TE110P to match any extensions for incoming DID. I've even used the exten => _X.,1 And it still will not match that. All I get is: -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at zap-custom,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at zap-custom,s,1 still failed so falling back to

Hi, This error I got it just when I gonfigure zaptel support isdneuro 31 channels. But if I configure zaptel to support T1 and just 24 channels I have no problem. ######################################################################## #### # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # loadzone = it span=1,1,0,ccs,hdb3,crc4 bchan=1-23 dchan=16

Helo, I have a troubles getting to stable work of Digium TE110P card (mailed some time earlier in the list) - I can't get 100% pseudo zap interface accuracy (zttest), so getting HDLC aborts and call drops. I tried number motherboards, hardware and software configs according to info in wiki, thisl list and number of websites - no luck. So I ask everyboby who successfully use Digium TE110P card

Hello that might would be an easy question for someone, but im in doubt Is there any possibility to pass a call from one asterisk to another and then to ZAP channel. For instance I have "A" asterisk with numbering 45670 "B" asterisk with numbering 45680 second asterisk has TE110P card with single PRI port connected to Siemens EWSD. When I originate call from asterisk

I have a strange problem: if I put on hold an incoming call from my Digium TE110P, I can't resume it and the person at the phone continues to hear MOH until the line falls. My TE110P is connected with an italian E1 NT. If I put on hold a call on a SIP channel I can resume it without any problems. Is there someone that can help me? These are my configurations: zaptel.conf

Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion.

Hello, I'm using asterisk for a while, fully ip, and i'm now moving to ISDN, so i got a TE110P. The first problem is the "Unexpected Channel selection 3" message. I'm unable to find, on the web, the meaning of such message. Can somebody explain me? My second problem is when a call come in, and hang up. Asterisk shows " -- B-channel 0/31 successfully restarted on

Hi, I need little help with setting up new server i recently bought TE110P/E1. I have 1 PRI and I will be using Soft phones on the agents site. This is 1st time for me to work with Zaptel card and i am having some problem with that. if any one call help me with this or care to share extensions.conf with zap>sip sip>zap with me will be great. I want to be able to make call through SIP

TDMoE was it. Thank you!!!! Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Friday, June 24, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Exposing Zap Channels on Server A to be UsedByServer B

Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp (bhcompile@decompose.build.redhat.com) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps on digium website...no errors reported. The modules seem to have loaded...here's what lsmod shows: Module Size Used by

Robert, Essentually I want to be able to have Server B dial the extensions connected to server A as well as route calls to the outbound route on Server A. Server B will have little to no knowledge of what is on Server A. I just want it to dump the calls off. For some reason I keep thinking this was a PRI type of thing. Like there was a module that loaded up as a fake PRI that your

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

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP