similar to: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

hello, sorry that I post it here again, as I have already posted it on www.redhat.com/archives/fedora-xen/. Now I am working on fedora6Xen,i.e. 2.6.20-1.2944.fc6xen .Several questions: 1. After I have installed the windowsXP, the XP OS of course needs to reboot, but then the WindowsXP is automatically closed. How can I restart it? And I use #virsh start , the domain will be running, but when I

I keep running into the dead end that it can't find config.h in the source tree. It looks like newer kernels don't use it anymore. Someone ran into this awhile back when compiling 1.2 and it looks as though the issue was never resolved. Any ideas? Last time I tried this, I was on fedora core 5 64bit and all went well. It's not working on this newer setup Any ideas on what I can do

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off from Automation Alley, which is SE Michigan's version of Silicone Valley. -- Steven

I finished to try to install Asterisk Now 1.4.0 on an AMD 3800 dual processor machine. The install lookups on the search for the Sata drive, since however it loads the sata_sil driver it doesn't work. Did someone knows what version of Linux is using on Asterisk Now? Thanks, Carlos Alperin -------------- next part -------------- An HTML attachment was scrubbed... URL:

Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio. Is their any difference in the inbound & outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc? Thanks, Ross ---------- Original Message ---------------------------------- From: "Carlos Alperin"

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

Dear All, I need to bind two different ports at the same time for SIP. 5060 and another port number. Is it possible ? It would be something like port=5060,5062 Isamar

Hello Everyone, This is my first post, and this is my problem :-). I have a asterisk@home, work excellent (only internal users), but i need outbound calls. One person give me an access to his "Cisco 5300 Media Gateway", he give me a dial rule and the router ip address. I've created a SIP Trunk, and a outbound routing, with all the info (the rare thing, the

I need to find someone to work with me in the Grand Rapids Michigan Area. Someone good with Linux and Asterisk would be ideal. Please get me contact info if you are interested. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/315eff3f/attachment.htm

This is all the reference to PTHREAD_MUTEX_RECURSIVE on lock.h #ifdef __APPLE__ /* Provide the Linux initializers for MacOS X */ #define PTHREAD_MUTEX_RECURSIVE_NP PTHREAD_MUTEX_RECURSIVE #define PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP { 0x4d555458, \ { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0x20 } } #endif #ifdef BSD #ifdef __GNUC__ #define

I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the

Before I go ahead and spend $40.000 on a Cisco 5400, just because my clients need T.38 faxing, I want to ask the community if there is any chance of having Asterisk receive G729+T.38 and sending the call via Zaptel to its final destination. Any answer will be appreciated. Federico

Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday,

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

Hi all, I'm newbe with asterisk and i'm facing with this problem that i'm not able to solve. I've to put an asterisk box after an analogic pbx wich require a 0 digit to give the dialtone. So when a client ask asterisk to dial an extension it should 1) send the 0 digit 2) wait for the dialtone 3) dial the extension the client send. How can i obtain this result? Thank's in

Hi list! I have several boxes running asterisk as I want, no problems there but the one thing I haven't sorted out is how to properly start asterisk on boot time. This is probably a n00b class question but how do I properly set this up (I didn't find any docs on this). The zaptel script alone does not load the proper driver on boot time, I guess I need to do some thing with the

Hi all!, i search for some information about to setup my asterisk box with e-mail notification when a I call the voicemail application. Voicemail application works fine in the Dial Plan but nothing happens with email notification ...so what i need to know about this?...wiki pages did not help me ....thanks! G. ----- Original Message ----- From: <asterisk-users-request@lists.digium.com>

Hi, I am facing problem in sync time within DomU with Dom0. After gone through mailing lists, I have set /proc/sys/xen/independent_wallclock to 0. But still DomU are not syncing their time :-S Machine using kernel-xen-2.6.19-1.2911.fc6 xen-3.0.3-3.fc6 xen-libs-3.0.3-3.fc6 Please advice to solve this weird problem, Thanks _______________________________________________ Xen-users mailing list

hi, i´m using fedora core 6 and want to build xen 3.1 from source for the 2.6.20 kernel. how can i build xen with the existing kernel-config from /boot (e.g. config-2.6.20-1.2948.fc6xen) and an installed kernel-devel (source) package? is it good to use this config or should i create a new one? i searched the net and haven´t found any howto to build xen from source with newer kernel versions and