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...establish 1 G.729 call and 3 G.711 with RT31P2-NA (using three-way calling). In PAP2-NA if I mark "Use Pref Codec Only" and there is one call established, when I call the PAP2 it replies with "488 Not Acceptable Here". Thanks, Leonardo -- Leonardo Gomes Figueira sabbath@planetarium.com.br

Hi All, I have installed chan_unicall and MFC/R2 successfully, and is runnign fine. But I noticed that once unicall is installed, asterisk CPU usage as reported by 'top', jumps to 99% every few seconds. I have no incoming calls, and I have even removed the E1 lines from card and I tried almost everything possible but I was not successful in determining the cause of this high cpu

Hello Has anyone been able to get the Soyo G668 phone to work with Asterisk? I've been playing with it for two days and I haven't had any luck. I've got a soft IP phone(xten) to work fine. If anyone has a sample config page that would be great. Thanks Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, thanks for great job do with xen .deb, we have a problem on xen 3.4.2 (certainly not just us) how is impossibile do hvm domu work because there isn't qemu-dm, i read a few months ago discussion of this but also now there is the problem, will be resolved soon? it possibile rebuild the current package with the simple addition of qemu? thanks in advance for any reply -------------- next

I first downloaded Vorbis about a month ago, around the time Rok Papez posted comparisons of Ogg and Lame [*]. Here's my much rougher test on a single file. CPU: AMD K6-233 MP3: LAME 3.70, mp3blaster 2.0b6 Ogg: Vorbize 0.6 (Xiphophorus libVorbis I 20000718), ogg123-0.1 Data set: Ministry, "The Fall" (51,955,724 byte WAV) * * * Ogg Vorbis command line: time vorbize -v -w

Brendan Cully wrote: > > The format of ices.conf changed somewhat in 0.2, to support multiple streams > from the same playlist with different reencoding options. In short, these > settings have been moved into the Stream node. For more info have a look > at the sample ices.conf included with the distribution. I did modify my ices.conf when I made the upgrade. Here's my

Hi, hoping that experienced hands will quickly show me the right way: after a fruitless web search i am turning to this list with my rather elementary question: is there any other way to communicate with the IAXy besides using special utility software that needs to be compiled under UNIX? Here is the story: about two months ago, after some not very satisfactory attempts at using SIP (my phone

...be used for simple redirections as well,              this example will redirect all requests for http://server:port/ to              the status page         -->         <alias source="/" destination="/status.xsl"/>         <ssl-certificate>/search/path/to/back-sabbath.pem</ssl-certificate>     <ssl-allowed-ciphers>ECDH+AESGCM:DH+AESGCM:ECDH+AES256:DH+AES256:ECDH+AES128:DH+AES:ECDH+3DES:DH+3DES:RSA+AESGCM:RSA+AES:RSA+3DES:!aNULL:!MD5:!DSS</ssl-allowed-ciphers>     </paths>     <logging>         <accesslog>access.log</acces...

Hi all, I have begun the release of my MFC/R2 protocol software. At http://www.opencall.org/installing-mfcr2.html there are instructions for installing what I have released so far. This is the MFC/R2 protocol software, and a test program. The software to interface Asterisk to the MFC/R2 code will be released shortly. It used to work, but it hasn't been touched for a while, and Asterisk

I am getting the following on my 'new box' for my DNS server: Date: Wed, 20 Feb 2013 13:58:04 -0500 From: To: Subject: [abrt] full crash report User-Agent: Heirloom mailx 12.4 7/29/08 abrt_version: 2.0.8 cmdline: ro root=/dev/mapper/vg_onlo3-root_01 rd_NO_LUKS rd_LVM_LV=vg_onlo3/root_01 LANG=en_US.UTF-8 +rd_NO_MD SYSFONT=latarcyrheb-sun16 KEYBOARDTYPE=pc KEYTABLE=us

Hi, I'm trying to disable the internal dial plan of an Avaya 4602 with SIP firmware 1.1 but couldn't find how to do it. Even if I configure a custom Dial Plan it keeps adding other builtin rules to my dial plan. Ex: Configured dial plan: DialPlan 19xx|7[8-9]xx|0xxxxxxxxxxxxxxxxxxx+ Reboot. On the syslog it shows: Aug 17 12:42:12 192.168.0.115 DigitMap:

Hi, I wonder if on Asterisk 1.2.X calls from queue answered by Agent channel still must be transfered only by Asterisk internal transfer (features) like on 1.0.X ? The wiki says on http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Queue "Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option

Hi, after upgrading to Asterisk 1.2.8 from 1.2.7.1 I got a problem with Grandstream BT100 after making an attended transfer (FLASH + NUMBER + SEND + WAIT ANSWER + TRANSFER). After the transfer, the display clears all the info except the clock, there is no dial tone, the WEB admin stops working. Incoming calls make the display light turn on but there is no ring and no callerid on the

Hi, executing the reload command on Asterisk make the Unicall channel unblock all Unicall channels dropping active calls. Is this just a local error (maybe in the other end of my E1 ?), an option in the source (maybe a debugging option ?) or it must be this way ? Asterisk: 1.0.9 unicall: 0.0.3-pre4 protocolvariant: br Thanks, Leonardo Aug 31 15:00:55 WARNING[28411]: Already have a