similar to: Urgent: Unable To Execute after updating from SVN

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Hi, I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Maybe there are some other non commercial applications which work under

Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate

Hello list, I have been running Asterisk CVS for a good while. When I try to install 1.0.3, asterisk won't start. Below are the last few lines of output before Asterisk crashes. I ran "make samples" to start with a fresh setup. [app_read.so] => (Read Variable Application) == Registered application 'Read' [app_alarmreceiver.so] => (Alarm Receiver for Asterisk) ==

Anyone using the AlarmReceiver? Does it work? Mine doesn't seem to communicate properly. How can I tweak the DTMF settings? Is it in the zaptel.conf or somewhere else?? -- Executing AlarmReceiver("Zap/1-1", "") in new stack > AlarmReceiver: Setting read and write formats to ULAW > AlarmReceiver: Answering channel > AlarmReceiver:

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for

Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

I have built Asterisk 1.4.4 on my Solaris 10 x86 box: LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib' CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw --without-oss --without-vpb --prefix=/opt/asterisk-1.4 The build and install go fine but the asterisk executable reproducibly dumps core with a segmentation violation. If I start it as: asterisk -gc and

Hi, I've brought this up in the past and there was a good discussion - am wondering if there have been any new developments. Our dialtone service, like I am sure is true for most ITSPs, touts the ability to drop your POTs lines for significant savings. For businesses we have a low-cost Atom based PBX and a "fax relay" setup locally with hylafax/iaxmodem to solve that issue,

> Message: 1 > Date: Tue, 15 May 2007 23:01:24 -0400 > From: Frank Tarczynski <ftarz at mindspring.com> > Subject: [asterisk-users] Asterisk 1.4.4 reproducibly dumps core on > Solaris 10 > To: asterisk-users at lists.digium.com > Message-ID: <464A7404.5000706 at mindspring.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I have

Hi, I finally found an ATA which works really well with asterisk and its application alarmreceiver. Frankly it works just like the TDM card. It is Soundwin S800 series ATA. Cheers Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060512/7f7175cf/attachment-0001.htm

Hi, I've been trying to get my ADT alarm panel ( Lynx24R ) to work with asterisk and sipura SPA 2000. So far the alarm company has been recieveing lot of undefined events. Someone suggested I should change communication format to 4x2 instead of default Ademco CID. Unfortantely ADT is reluctant to hand me over the information I need. I can change the communication format on the panel since

I'm trying to setup DATA calls with Dial(Zap/g1d/12345678), but with PRI DEBUG SPAN 1 on, it seems to connect a regular SPEECH call. I'm using 1.0.6. Is this feature broken in stable release? There seems to be support in the source, but it doesn't work. Does the Telco set what each PRI channel support? Like DATA or SPEECH etc.. Do I have to specify in zapata.conf or zaptel.conf that

Hi, on a configuration with one external ISDN S bus (to telco) and one internal S bus (to ISDN telephone), where Asterisk is in the middle (using HFC hardware), I noted the following: - when a GSM phone or ISDN phone calls in, the Transfer capability is Requested transfer capability: 0x00 - SPEECH - when an analog phone calls in (either from an analog line or an analog ISDN

I used: cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds iaxyprov astcc and in the same order I try to compile it. Asterisk ends with the lines below. It complains of a newer libpri, but I just did it a step before! What do I miss? chan_zap.c:62:2: #error "You need newer libpri" chan_zap.c:128: error: parse error before '<<' token chan_zap.c:133:1:

I installed FC4, ran command, # yum install asterisk. A bunch of stuff happened, but can't locate .conf files. I have a list of files: /usr/share/doc/asterisk-1.2.4/configs/features.conf.sample /usr/share/doc/asterisk-1.2.4/configs/rtp.conf.sample /usr/share/doc/asterisk-1.2.4/configs/extensions.conf.sample /usr/share/doc/asterisk-1.2.4/configs/logger.conf.sample