similar to: current cvs - zap failure with tdm?

Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3 Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the "lock-up", IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk. Box is debian w/ asterisk built from scratch. My setup is

Hello, Can anyone help with the output shown below? It?s running on RH9, recent CVS of Asterisk and with one X100P card (2 channels), a budget tone 102 and Xlite softphone. CLI> -- Starting simple switch on 'Zap/1-1' Jul 7 18:42:24 WARNING[1192437440]: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no

I have a TDM400 running telco lines on ZAP2-4 My after hours config is supposed to receive the incoming call then divert it to my home phone by calling out one of the other zap channels available. console output as such... Starting simple switch on 'Zap/3-1' -- Executing Dial("Zap/3-1", "ZAP/G1/0823274210") in new stack Aug 20 19:06:44 NOTICE[646]:

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: ----------------- fxsks=1-2 loadzone=us defaultzone=us zapata.conf: ------------------ context=inbound-analog signalling=fxs_ks group=1 channel => 1 channel => 2 I created a couple of test extensions: ; test extensions

Hi I'm running safe_asterisk, but get core-files in /tmp - how do I debug them ? I know gdb asterisk core.12370 and bt full But it didn't show anything usefull for me. Can anyone help me ? (Running asterisk 1.0.2 with ast_data /Hans-Henrik ----------------- Last from bt full: priority=200, callerid=0x81b8e90 "Dial", action=1134845864) at pbx.c:1384 e =

Hello All, I just discovered that the last I unfortunately I send only to Louis - not the list. So below are my answers included (and log outputs that were requested). Never the less in meantime I have investigated further into SAMBA & winbind. I was able to setup samba dc based on previous instructions and guidelines successfully. I additionally setup a debian samba member with winbind.

I'm really racking my brain trying to figure this one out here. I am running a pop3 server for remote offices on CentOS 5.2. We purchased a SSL cert from Verisign and installed it on our dovecot server, but I continue to get failure problems with the cert and I don't know where to go from here. here is some info about our config: dovecot version: # dovecot --version 1.0.7 hostname:

Dear R users, I'm having some problems with the stack() and unstack() functions, and wondered if you could help. I have a large data frame (400 rows x 2000 columns), which I need to reduce to a single column of values (and therefore 800000 rows), so that I can use it in other operations (e.g., generating predictions from a GLM object). However, the problem I'm having can be reproduced

Hi I have an asterisk box with digium hardware connected to a Siemens EWSD version 15 using a crossed E1 cable. The asterisk is serving as a h323 media gateway. When i start asterisk, the Siemens gives me this alarm: REPORTES GENERADOS EN LA EWSD AES/V15SBOL/BOLCBK1V51327079/013 05-06-06 11:25:38 7773 3062/03728 HF.ARCHIVE-80040

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

Hello, I'm traying to link 2 * servers using SIP and the following errors was show: "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13

#### I checked through every 3 factor * 3 loading case. #### While, 4 factor * 3 loading failed. #### the data is 6 factor * 3 loading require(sem); cor18<-read.moments(); 1 .68 1 .60 .58 1 .01 .10 .07 1 .12 .04 .06 .29 1 .06 .06 .01 .35 .24 1 .09 .13 .10 .05 .03 .07 1 .04 .08 .16 .10 .12 .06 .25 1 .06 .09 .02 .02 .09 .16 .29 .36 1 .23 .26 .19 .05 .04 .04 .08 .09 .09 1 .11 .13 .12 .03 .05 .03

Date: Fri, 31 Aug 2007 13:19:32 +0300 From: "Dovid B" < > Subject: Re: [asterisk-users] phone as control interface (was 99 bottles of beer) To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <05bf01c7ebb8$6ff3ffc0$0a00a8c0 at DovidLaptop> Content-Type: text/plain; format=flowed;

Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using

Is there a way to apply paste to?list(form1 = EQ1, form2 = EQ2, form3 = EQ3, form4 = EQ4)?such that I don't have to write form1=EQ1 for all my models?(I might have a list of 20 or more)? I also need the EQs to read the formulas associated with them. For example, below, I was able to automate the name assignment but I could not figure out how to?to set up the list using?paste or other

Hi, Thanks a lot. Is it possible to implement that functionality as a pragma? Ex. .. [code] ... #pragma A .. .. #pragma B .. .. Lets say I want to reserve registers X9 and X10 only within pragma A and B. Is it possible to achieve this in llvm? Thanks Nisal On Mon, Jun 12, 2017 at 9:54 PM, Tim Northover <t.p.northover at gmail.com> wrote: > Hi Nisal, > > (Adding llvm-dev back

Hi, Using S=1000 and simdata <- replicate(S, generate(3000)) #If you want both "m1" and "m0" #here the missing values are 0 res1<-sapply(seq_len(ncol(simdata.psm1)),function(i) {x1<-merge(simdata.psm0[,i],simdata.psm1[,i],all=TRUE); x1[is.na(x1)]<-0; x1}) res1[,997:1000] #????? [,1]???????? [,2]???????? [,3]???????? [,4]??????? #x1??? Numeric,3000 Numeric,3000

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,