similar to: Asterisk and unknown codecs and GSM

Hello, are the gsm codecs in Cisco as53xx or 36xx (gsm, gsmfr) copatible with Asterisk's gsm codec? Thanks in advance, Thomas

Hi all. I'm currently having a following problem: I have only ssh connection to a CentOS 5.2 system, there are two harddiscs on it. One stores the system (/ filesystem) and the other should be used to help restore the system in case of first disks' failure. I thought that maybe dump would be a good utility to make it. But in only works on read-only filesystems. In one article I've read

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

Hi all, I am wondering whether there is a small bug in the binom.test function of the ctest library (I'm using R 1.6.0 on windows 2000, but Splus 2000 seems to have the same behaviour). Or perhaps I've misunderstood something. the command binom.test(11,100,p=0.1) and binom.test(9,100,p=0.1) give different p-values (see below). As 9 and 11 are equidistant from 10, the mean of the

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Hello, I''m trying to setup Xen on a machine with 2 physical network devices using linux bonding without success. I haven''t received any answer on my question on xen-users, the only discussion on this seems to be at http://lists.xensource.com/archives/html/xen-devel/2005-10/msg01174.html Could someone shed some light on state/ plans regarding Xen and bonding? Regards --

Does anyone know of a cross browser solution to print a page/url after a user clicks a button? Currently, I''m using a "hidden" iframe to do my bidding. But from my experience, IE requires that the iframe''s src attribute be set initially to the url, in order for the page to open properly. I wasn''t able to add the iframe to the page dynamically, either. So

Hi all - This isn't really directly Asterisk related, but has anyone successfully set up a Polycom phone to register two lines on two different Asterisk boxes? I can get the first line to register, but the second one does not. I can still place calls from that second line, which indicates to me the server, user, and secret are correct. I'm running the newest 2.6 series firmware with the

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

I''ve seen a few demos online, for php, but I''m looking for a file upload progress bar implementation with Prototype. Can anyone point me in a good direction for one? I''m not using Rails, so I can''t use the upload_progress addon, unfortunately. Thanks for any info Jim _______________________________________________ Rails-spinoffs mailing list

Just out of curiosity, is there a plan to support Comet (name coined by the dojo guys) w/ prototype? Comet represents persisting an http connection for low latency data operations. It also represents a nice alternative to polling. Jim _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org

Hi, Anybody out there running Adit600s? I have in my care an Adit600 channel bank connected to an old (version 1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk (1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail. I have attempted to add the "slowsequence = yes" line to mgcp.conf. (It seemed to be the only likely candidate in the example files I found

Hey all, I''m not sure if IE can support this, but I''m trying to get the dimensions of an element that has percentages for height/width. In the example code below, I''m trying to get the dimensions for the ''MyCell'' element. Firefox supports Element.getDimensions(''MyCell''), but IE returns 0 for the height & width returned by that

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've

LLVMers, The LLVM 2.1 pre-release version2 is now available: http://llvm.org/prereleases/2.1/version2/ You can help test this pre-release by doing one of the following: 1) Download llvm-2.1, llvm-test-2.1, and the appropriate llvm-gcc4.0 binary. Run "make check" and the full llvm-test suite (make TEST=nightly report). 2) Download llvm-2.1, llvm-test-2.1, and the llvm-gcc4.0 source.