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...h 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

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

Hi all, I am attempting to call from softphone to softphone, I am using X-lite to call another X-lite. I get the phones to call each other and finnaly connecting, but cannot hear the voice at all. Is there any ideas as to why this is happening. (I don't have sound card in my linux server. I need one in my linux server ??) PS: callonhold is working but cannot hear the music too. look at

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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 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

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