similar to: No Asterisk Sounds on SUSE ES 9/Linux 2.6

I know this is probably way out there but............ Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an

On a new * asterisk install onto new install Gentoo 2003.4 upon startup of asterisk: WARNING[16384]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/res_perl.so: undefined symbol: PL_thr_key WARNING[16384]: loader.c:429 load_modules: Loading module res_perl.so failed! perl -v = v5.8.5 built for i386-linux-thread-multi <= I installed ithread support in perl Have not been able to

Hi Peoples I am having problems with meetme, in that it responds with "conf-invalid" when I dial a conference number. I notice that there is a note with regards to ztdummy, and the need for that to be loaded. Is this still the case? Is meetme dependent on this module? I do NOT use zaptel cards in my system, and there for zaptel is not loading. Can anyone shed some light

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

I would like to use one of my Snom 200's 5 programmable keys to park calls. I am using image SIP 2.04g. I have tried a variety of combinations and have come to the conclusion that: 1) On the Key Mappings administration page, I must select the "Transfer" under the "Break Keys" option box to be able to successfully transfer calls using the "Transfer" button. 2)

Using an example provided by "The Hitchhiker's Guide to Asterisk", I made the following addition to my extensions.conf file: [inbound-analog] exten => s,1,Wait(1) exten => s,2,SetVar(counter=0) exten => s,3,Answer() exten => s,4,Wait(1) exten => s,5,DigitTimeout(15) exten => s,6,ResponseTimeout(10)

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

Hello All, Can someone please tell me about Zaptel? Is it only needed if you are going to have an interface card like TDM400P installed on the Asterisk server? Do you really need it if you do not have the interface card? Thanks, Lonnie

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Hi, I've been trying to bring our Asterisk server to the latest version. I've grabbed the latest CVS and upon trying to compile zaptel, I get the following errors: gcc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c gcc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB

Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM

Hi to everybody, seems that I cannot load the zaptel modules: ztdummy says the following: [root@Hayabusa misc]# modprobe ztdummy /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_register

We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at nt_jnewman at yahoo.com. Justin

Hello System: Debian etch with kernel 2.6.18-4-686 or 2.6.18 custom. Asterisk Version: SVN-branch-1.4-r55483M Zaptel Version: SVN-branch-1.4-r2302 modules all ok in compilation time. And modules loaded: ztdummy 5928 0 rtc 13364 1 ztdummy zaptel 181540 1 ztdummy crc_ccitt 3200 1 zaptel In /dev/zap directory I have:

Hi there Just a quick question. I have been playing around with asterisk CVS-1.0.02 on fedora core 1 (2.4 kernel) and I would like to have a look at asterisk v 1.0.6 but am still a little uncertain which linux kernel is best to run on ?, can I use Fedora Core 3 (is it the preferred kernel) or should I stick with FC1 Ps - the only additional hardware in the box will be a digium single port E1

Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes

Is ztdummy still required with Asterisk 1.4 when no zaptel cards are available to use for timing? In all the beta releases I used to get a warning when Asterisk started up, saying that no timing device was found. The warning seems to have gone away with the full release of 1.4, which prompts the question... Is it still required? Does 1.4 do something different for timing? Regards, David

Friends, Please send all offers for help *off list* to us at info@astricon.net. Do not disturb the list with offers of your services, please. I repeat: Only the Developer's Meeting will be considered for broadcast at this time. In order to enjoy the conference, you will simply have to be there. It's an IRL experience - meeting all the other Asterisk user's from around the globe,

What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the