similar to: ZAP ISDN losing digits

Displaying 20 results from an estimated 500 matches similar to: "ZAP ISDN losing digits"

2006 Feb 15
5
Aasterisk large-scale deployment w/analog phones
hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar
2005 Sep 05
2
DTMF issue on IVR
Hi All, I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt and asks caller to dial four-digit extension. Caller has to dial slowly, otherwise, Asterisk cannot recognize the extension number. I look at the trace on Asterisk CLI and there are missing digit in the middle of string. ex, caller
2005 Oct 17
1
astcc missing to bill random calls?
Hello list, I just came into a strange problem wth astcc. the trouble is astcc.agi does not bill some calls. The calls are logged in the cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an empty dstchannel, and with a lastapp field of "hangup". I suppose that astcc.agi was not able to get the answeredime variable from the SIP channel... I have added a few functions to
2005 May 30
1
I865, HFC-S etc.
Hi, I'am having some problems with new mainboards and 3xHFC-S cards. The the first problem was with interrupts, I mean if HFC-S card was using interrupt i.e. 21 or higher - it didn't work. Solved by disabling APIC. However, still the driver behaves a little bit strange. If card 0 & 1 is TE and 2 is NT, TE works fine, but NT is not working at all. If card 0 is NT and 1 & 2 TE - all
2005 Oct 11
2
IAX or IAX2 ?
Hi, I have read the wiki entries on IAX(2), but I'm afraid, it still have some questions: I have a working connection between two Asterisk-Servers (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on Debian 3.1) via IAX. Does this connection work with IAX or IAX2? When I try to load chan_iax2.so, I get the error message chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource:
2006 Apr 05
2
chan_modem_i4l delay
Hi, I currently use? Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just
2005 Aug 26
4
system crash
We just had * crash on us - no calls could be made / received. We had to kill -9 the * process. Checking the error logs, I came across these two lines, with the times matching the crash: Aug 26 13:48:00 WARNING[19282] pbx.c: Local/6024@AgentQ-94ce,2 already has PBX structure?? Aug 26 13:48:00 WARNING[19282] channel.c: Thread -1105359952 Blocking 'Local/6024@AgentQ-94ce,2', already
2006 Feb 19
3
Cisco 7905 can't register
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k on Debian stable). It could, however, register with another installation of Asterisk and the settings on the phone (apart from the SIP proxy address) haven't changed since then. On the new Asterisk box my sip.conf contains this: [jeremy] type=friend regexten=801 allow=g729 host=dynamic secret=PASSWORD nat=yes
2005 Jul 28
1
different _source_ addresses for registrations?
Hi all, Can I choose a different source ip address that asterisk uses when sending registration requests, and another one when transfering calls to a destination with the DIAL() command? The thing is, when I register with my sip provider proxy at let's say 111.111.111.111, I want the registration packets to be sourced from one of my ip addresses - 222.222.222.222 for example. I will then use
2005 Oct 15
1
Problem with '#' key recognition
Hi, I seem to be unable to get Asterisk to recognise the '#' key being pressed to acknowledge an incoming call from a queue. No matter how many times I press the key to acknowledge, the Asterisk server acts as if I have not. I have installed the ztdummy module, and it seems that Asterisk is picking it up OK (mainly since it's not moaning about permissions for IAX timing any
2006 May 03
1
Running applications when a queued call is answered
Hello, I'm experimenting with Asterisk for possible use in a call center. I'm trying to figure out how to run applications when an agent answers a call in the queue. I see that the queue itself supports a very limited range of applications; for example, I can give a URL to the Queue() application to SendURL(), or an announcement to read to the agent. I'd like to do some slightly
2005 Sep 08
1
Hangup problem
i have a box running debian sarge with asterisk installed from distribution packages: CLI> show version Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by kk@nyx on a x86_64 running Linux I have managed to configure a simple dialplan (the PBX task is quite simple as this is a small office with just a few phones) I have one Zap (PSTN) line connected to it and one SIP to a local provider. After
2006 Feb 15
1
Asterisk large-scale deployment w/analog phones
I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com "Realize the Value of Customer Contact!"TM This e-mail is intended
2012 Sep 10
3
How to remove last comma when iterating through hash in erb template
I need to produce a line in a config file in the format x = "ip1,ip2,ip3" I am using the method below to sort the hash before iterating over it. However, as you can see there will always be a final comma which breaks the app that uses this config file. Does anyone know how I could remove the final comma? ipv4_bind_addresses = "<% routes.sort_by {|key, v| key}.each do |key,
2005 Mar 24
1
voicemail problems with CVS-HEAD
Hello, I have moved from Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k (debian pkg) to CVS-HEAD, and realtime. Compiled no problem and now running, with realtime extensions and sip users in postgres (ODBC connection) database, trunking also works. I have looked on google, wiki, and this mailing list, along with talking to some peers, but to no avail. My problem revolves around voicemail. I have looked
2005 Mar 29
1
HFC-S
Hi! I have just installed Redhat 9 and Asterisk to my computer, and now i have problems with my non-zaptel Card, I don't know how to set it up since all instructions are for digium's hardware. I have searched from the Internet for hours now, can you help me to understand all this HFC-s thing and how it is related to CAPI, ISDN4Linux, bristuff and so on. I have to say that I am not so
2005 Mar 28
1
Connecting quadbri to EuroISDN with 2 TE and 2 NT ports - what cables and settings ?
Hi, I'm trying to connect quadbri between powered ISDN phone and ISDN line: ISDN <---1---> TE - * - NT <--2--> Phone I use quadbri, suse 9.2 and latest 0.2.0-RC7k bristuff. I've used sample settings provided with package, but do get strange error (I think that I have wrong setting for P2P or P2MP setting and cables 1 and 2). If I connect phone to ISDN with straight cable
2005 Jun 01
1
rxfax problems - cont.
Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242
2005 Jul 19
2
No sound when bridging two single FXO cards
Wow ! No reply... May be I must talk about X100P instead of X101P ? Is someone has yet encountered this kind of "no sound" problem when bridging two FXO lines like this (first as input, second as output) ? Any idea ? TIA. Best Regards, Francois BERGERET, France. ----- Original Message ----- From: "Francois BERGERET" <f6hqz-m@hamwlan.net> To: "Asterisk Users
2005 Jun 14
2
# no longer working
Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel