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I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax machine. Both are connected to a DAHDI board. I'd like to route incoming PSTN fax calls to the extension of the fax machine and process non-fax calls through different dialplan.logic. What's the best way to go about doing this? I've looked into Fax for Asterisk, bit I'm not sure that I want it or NVFax

----- Original Message ----- From: "chris" <chris@fivestartel.com> To: "Frank Tarczynski" <ftarz@mindspring.com>; <seruser@iptel.org> Sent: Tuesday, July 26, 2005 9:54 AM Subject: Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found > hi Tzafrir > > any idea how we can fix this error? > > include/asterisk/strings.h:232: parse error before `va_list'...

I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've

I'm looking to build the zaptel drivers on a Solaris 10 X86 box. I've found the driver source code on https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted along with Asterisk 1.2.7.1 Does anyone know of a fresher version? Is this code considered "somewhat ready for prime time use"? Thanks, Frank

I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at my end. There's enough delay time that I hear an additional ring after the PSTN number has

I'm trying to set-up H.323 support under Asterisk. I built a recent CVS release and the ooh323c code from the asterisk-addons. Everything built and installed and the H.323 stuff loads OK when asterisk starts. What is the easiest way to check if the H.323 code is working? I've edited the h323.conf and extensions.conf files but I'm sure that things aren't right. I've

I'm having a problem with asterisk-1.4.4 dumping core under Solaris 10 with a SIGSEGV error. gdb gives this stack trace: #0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1 #1 0xfec2a386 in _ndoprnt () from /usr/lib/libc.so.1 #2 0xfec2d4bb in vsnprintf () from /usr/lib/libc.so.1 #3 0x080e86de in ast_dynamic_str_thread_build_va (buf=0x8172763, max_len=0, ts=0x81482a0, append=0,

My AsteriskNow box was updated to Centos 5.6 (2.6.18-238.5.1.el5) and DAHDI doesn't want to load. I've tried building it from the sources, but get this error message: CC [M] /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/card_bri.o In file included from /root/Desktop/dahdi-linux-complete-2.4.1.1+2.4.1/linux/drivers/dahdi/xpp/xpd.h:31, from

I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help?

I'm having a problem with my IAXy not always connecting to my Asterisk box. When I pick-up the phone plugged in to the IAXy I get a busy signal. I have to hang-up the phone and wait a few seconds after the orange LED goes out and then try again. When this happens I don't see any connection attempts in the Asterisk -r output. When I do get the IAXy to connect to Asterisk I get a

>Message: 11 >Date: Sun, 28 Aug 2005 11:46:29 +0800 >From: "chris" <chris@fivestartel.com> >Subject: [Asterisk-Users] error compiling on solaris 10 >To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> >Message-ID: <003a01c5ab83$149a9e30$650fa8c0@acid> >Content-Type: text/plain;

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

> 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...

...cohen at xorcom.com > +972-50-7952406 mailto:tzafrir.cohen at xorcom.com > http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir > > > > ------------------------------ > > Message: 12 > Date: Sun, 10 Apr 2011 09:32:03 -0400 > From: Frank Tarczynski <ftarz at mindspring.com> > Subject: [asterisk-users] AsteriskNow updated to Centos 5.6 and DAHDI > doesn't work > To: asterisk-users at lists.digium.com > Message-ID: <4DA1B153.20703 at mindspring.com> > Content-Type: text/plain; charset=windows-1252; format=flowed > >...

I've built Asterisk from recent CVS sources on a Solaris 10 X86 box. I tweaked the makefile to get the build to run using gcc. And most recently ran into va_args problems with new code in asterisk/utils.c. It seems to run OK and register with my VoIP provider, but I'm still having trouble setting-up connections to my Sipura SPA-3000. Does anyone have some experience under S10 X86?

I'm running a recent CVS build under Solaris 10. In the shell than I'm running the Asterisk console I have TZ=US/Eastern and in my voicemail.conf I have tz=eastern and eastern=America/New_York|'vm-received' Q 'digits/at' IMp. The voicemail envelope information seems to be exactly 4 hours ahead. No matter what I try I can't seem to find the cause. Any ideas? Frank

I'm running a recent CVS build under Solaris 10. In the shell than I'm running the Asterisk console I have TZ=US/Eastern and in my voicemail.conf I have tz=eastern and eastern=America/New_York|'vm-received' Q 'digits/at' IMp. The voicemail envelope information seems to be exactly 4 hours ahead. No matter what I try I can't seem to find the cause. Any ideas? Frank

I need some help generating configuration files for Asterisk. Since I'm running under Solaris I'm having trouble with some of the utilities that are more linux-centric. Can anyone recommend a free/low-cost package to generate conf files that is not linux-dependent and will handle a IAX2 and SIP trunk? Frank

I'm looking for a recent asterisk package for the Linksys WRT54G. Has anyone know of a 1.2.X build for this box? Thanks, Frank

I'm looking for pointers towards building and running the zaptel drivers under Solaris 10. Can anyone help? Frank