similar to: Mobilephone users get echo of them self when calling in to our asterisk server.

Qestion hvow can i use my SE w800i to reed flac files from a meorycard Best regards Fredrik

This file is 13 years old and NV01/03 isn't even supported by Nouveau. This patch remove README.NV1. --- a/README.NV1 2009-08-02 18:19:25.000000000 +0200 +++ b/README.NV1 1970-01-01 01:00:00.000000000 +0100 @@ -1,42 +0,0 @@ - Information for NVidia NV1 / SGS-Thomson STG2000 Users - - David McKay - - 20th March 1997 - -1.

Dear all I am intrested in making iterative estimation (thro' loop statements) of, say, linear regression model. For this purpose, I have written the following programme and that I have made use of a sample data (viz., exp.txt): ? Programme: ? # Linear regression modelling with sample data (try5.txt) # Repeated estimation through loop statement

Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Notice which appear when voice prompt is played. NOTICE[1217602880]: File sched.c, Line 209

Hello, I have a Nvidia Riva 128 ZX graphic card. But renouveau need the nvidia proprietary driver to work and there is no nvidia proprietary driver for Riva 128 or Riva 128 ZX. How can I send you informations that can help you to write a free and open source driver for this nvidia graphic chip. At http://users.tkk.fi/~jpakkane/ren/, I found nothing for NV3. But there is some information for NV1

A patch is attached to fix this problem. It removes the deprecated reference to README.NV1 and properly adds src/nv_rop.h Thanks, Rico Tzschichholz -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-Fix-make-dist.patch Type: text/x-patch Size: 976 bytes Desc: not available URL:

I have two asterisk machines. One of them has a Digium board (server A) and the other is simply using ztdummy (server B). Server A is running on Debian and Server B is running Gentoo. Server A is running Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running Asterisk 1.0.7. The problem I have is that when I try to transfer a call into a meetme room in server B, it simply hangs

Hi all , In my pbx ,there is no zaptel card ,so i loading ztdummy,but problem appear,when i dial the number into the pbx,sometimes i can not listen to the ivr ,and no rtp create. if i unload the ztdummy driver,the proble will disapper. I guess may be the timer source problem, but i dont't know how to solve it . anyone can give me some advices will be appreciated. asteirsk-1.4.21 and

Yeah I do use ring groups at the moment. But the problem is that I can't control "the flow". Let's take your example. dial(SIP/dev1&SIP/dev2&SIP/dev3) If I dial these 3 numbers, and dev2 is already one the phone. How do I check for that? I only want one of the three phones active at the time. But if no telephone is busy, they all should ring until the call is

I am still getting an error compiling the 1.2-Beta version. The tarball works fine, but I have never been able to compile the 1.2beta from CVS. I have been compiling CVS-HEAD on the machine for quite some time. It goes into this loop: if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \

Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com

1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all

hello, i got this error when i run make after downloading asteirsk from cvs. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/include -D_REENTRANT -D_GNU_SOURCE -O6 -Wcast-align -DSOLARIS -DBUSYDETECT_MARTIN -fomit-frame-pointer -c -o term.o term.c In file included from

hello i am trying to use res_odbc for sipuser. my connection is working. i have checked using isql. even cdr_odbc is working but i hav problem in res_odbc. i have created user in sip_buddies table but asterisk is no getting user from this sip_buddies table. /etc/asterisk/extconfig.conf [settings] sipusers=>odbc,asterisk,sip_buddies sippeers=>odbc,asterisk,sip_buddies

Question is quite easy: How am I supposed to configure Asteirsk to have the same extension, in 2 differents domains. In the general section of sip.conf, I add the domains, But how to say to Asterisk : user1@domain1 > Pasword1 user2@domain2 > Pasword2 Thanks for your help !!!!! JM -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

Hi, can i set up my asterisk for: - receive a skinny call in a specific context (yes, i have already compiled asteirsk with h323 support) - forward the call to a sip user A - make the sip user B join the call and create a conference between skinny caller, A and B maky thanks

I have read the wiki and several other internet documents. Can anyone make a comment as to what kind of functionality will you loose if you use Cisco 7940 phones with asterisk 1.4 things like: MWI, call transfer, conference,etc,etc. I have a customer with 6 of those phones that he like to use with the asteirsk PBX. thanks, -- ------------------------------------------------------------ Erick

http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi-box-is-official-named-airave-300451.php The above is quite interesting, it would be interesting to see if it uses sip, which I have no reason to believe otherwise, and if it does, can it be hacked to talk to Asteirsk? In which case one could have a very good extension to asterisk using any Sprint Cell phone, or maybe even

Hi there. We're going to develop a call centre app for internal use in our office. The call centre is probably going to be a java-based client installed on a windows machine that our secretary can use. Features should be a way to see incoming calls, answer them, and then transfer the calls to our different users/groups/divisions. If it also could be possible to have a way to see if the user