similar to: Ouch ... error while writing audio data: : Broken pipe

Hi all, I have a problem starting the ztdummy. Here is what happens: [root@asterisk /]# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy After this, ztdummy is visible with lsmod, but when I try MeetMe, I get following: == Parsing

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

Hi, We have phones registered at another soft switch, and would like to use Asterisk as a Voicemail system. Is it possible and how to configure Asterisk to send NOTIFY messages (for MWI) to the endpoints that are not registered to the Asterisk? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have stress tested the Asterisk Voicemail. We have encountered problem with simultaneous calls that are sent to the same mailbox. It occurred that several calls were writing to the same file. It seems that there is a synchronization issue in the Voicemail application. Did someone else find this issue? What would be the solution/workaround for it? Regards, Stojan Sljivic

Hi, Is there a module within the Asterisk standard distribution that provides SNMP features? Is there any third party software for that purpose? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/918b5ebf/attachment.htm

Good day all How do I get my asterisk and sip to use the password.I'm using x-lite.If I use just the username and no password it still logs on? Here is my sip.conf entry? [101] type=friend callerid="Test User" <101> context = test_1 ; Default context for incoming calls username=101 secret=123456 host=dynamic dtmfmode=inband ; Choices are inband, rfc2833, or info

Ok, After following BJ's advice and removing ogg.so I then got a pbx_realtime.so error in the same fashion. I removed that file, and then the next and then the next as you can see in the log below. I think something is not right. duh here is my sign..lol...but I am not sure even where this ast_register_file_version flag is in a config file or what step I have missed. I am doing a VOIP only

Hi, I use Asterisk 1.0.5 and TDM04B. When an incoming call over ZAP channel hangs-up, it takes 10 seconds until Asterisk realize that. How can I shorten the time of hang-up detection? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050614/de3eec0f/attachment.htm

Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. >From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

Dear all after an upgrade to 1.6 from 1.4 (as explained in the UPGRADE-1.6.txt file) the | delimiter is not working by default. I've added a compat section in asterisk.conf a [options] dontwarn = yes [compat] pbx_realtime=1.4 res_agi=1.4 app_set=1.4 And restarted Asterisk, but i still have problem to have the | delimiter working, [Jun 9 23:20:54] DEBUG[11744]: pbx.c:3122

Hi: I upgraded SuSE to 10 and Asterisk to trunk and now after deleting all modules and previously compiled stuff and recompiling asterisk, zaptel, and libpri, I get this failure of asterisk to start: [pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style pbx_realtime.so (0x31) loaded RTLD_LOCAL => (Realtime Switch) [chan_mgcp.so]Apr 25 03:36:41

I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running ztcfg and asterisk fails. [root@asterisk asterisk]# ztcfg Notice: Configuration file is /etc/zaptel.conf line 3: Unable to open master device '/dev/zap/ctl'

Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ============ ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here:

Hello all! My basic problem is that we haven't been able to get realtime to use ODBC to store configuration data. Here are the details: We've installed Asterisk on a CentOS machine as follows: 1. Downloaded, compiled, and installed FreeTDS 0.63 2. Downloaded, compiled, and installed unixODBC 2.2.11 3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel from CVS

Hello people, Went to IRC and zoa helped a bit by saying me to Do a insmod zaptel and insmod wcusb When I did the insmod zaptel it worked fine, but when I did the wcusb it said asterisk:/usr/src/zaptel# insmod wcusb Using /lib/modules/2.4.24/misc/wcusb.o /lib/modules/2.4.24/misc/wcusb.o: unresolved symbol usb_submit_urb_R3d6f870e /lib/modules/2.4.24/misc/wcusb.o: unresolved

Hi all, this is another post about this problem. I installed from scratch a new Suse Linux 10.0, with latest stable asterisk. Moreover I add the lines to /etc/udev/rules.d/50-udev.rules, in order to let the driver create the /dev/zap....... When I plug into usb port my TigerJet adapter, I see on /var/log/messages Feb 24 14:55:02 srvlnx05 kernel: usb 1-2: new full speed USB device using

I' m using Asterisk 1.09 on an virtual pc (VMWare 4.5) for testing. I can make calls from a Softphone to softphone, Hardphone to Softphone and so on. I can hear both RTP Streams. But when I call prompst on Asterisk I can hear nothing. RTP Stream goning from Phone to Asterisk but not the other way. I I start the PBX for console I got an error [app_rxfax.so][root@asterisk1 root]# Ouch ... error

Hi! I am in the process of upgrading our 1.2 servers to 1.6. We have a lot of realtime extensions with app=Macro and appdata=stdexten|080512|SIP/080512 But this does not work in 1.6. It is expecting , and not | as the argument seperator. If I change the | to , then it does not work in 1.2. Is there any backwards compatible switch you can enable in 1.6, so it accepts | as a argument seperator

Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: ---------------------------- /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field ?owner? specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: ***