similar to: codec problems with asterisk

Hello. I have been searching the archives for a simple, clear listing of the available codecs with total size, plus the data and overhead sizes. Does anyone have this handy, and can it be added somewhere, even the wiki. Regards...Martin -- The system will be down for 10 days for preventive maintenance.

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

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates

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, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.

I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many "answers." I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be

I'm trying to get Asterisk to pass thru calls using the G729 codec. I've got a 7960 phone and my gateway is an AS5400. I got the following messages when debugging SIP (7778881000 is the 7960): WARNING[1872]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/7778881000-2874(4) to SIP/as5400-35c1(256) WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Development Lab in Richardson, Texas testing our setup. We have a three server system

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems,

Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:.

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

Hi there, If I may keep this short, I would like your permission to use the drivers: ISOLINUX and MEMDISK, ISO 9660/El Torito CD-ROMs bootloader, to make a FreeDOS bootable CD-ROM. I plan to distribute the CDs for troubleshooting purposes and to cover my costs i plan to sell them for $10US each. I noticed that the most of the troubleshooting Cds are breaking copyright licensing so I wanted to

Hi, We recently installed asterisk too replace our office PABX, however we are finding it hard to get documentation on the way Agents login. In the agents.conf we have setup a user of agent => 1003,4444,Test. In the extentions.conf file we have added in exten => 9,1,AgentLogin. We can dial 9 and key in the uid and password which logs us in. The asterisk CLI confirms it is login, but when

Hi all, First of all, I want to say that I really appreciate all the work that's gone into LLVM, clang and the ORC API. I would be very unlikely to be doing my current project if they weren't available. I'm using the ORC API to JIT some C code (soon to be C++) on Windows. (I was on LLVM 8.0.0 but it didn't handle the 'allocator' attribute so I upgraded to LLVM 9.0.0.)

We are using a windows server 2003 active directory as our single sign on server. I have been able to get our RHEL4U6 servers to authenticate with active directory. My concern is that the RID mapping to unix uid/gid range (15000-20000) is stored locally on each machine in a tdb database. So far all of the servers have produced the same mapping, but I do not think it is guarantied. I think the

Dear Ladies and Gentlemen, I have a fairly large database (N=13,000) and a single main categorical discriminator between the groups. I want to look at the time course of a number of continuous biochemical variables over chronologic age. Therefore I believe I need to prepare hexbinplots in two columns with simple regression lines in them (with useOuterStrips (in library(latticeExtra) if

Unsure what exactly the problem is, but when using 4.8-STABLE I was unable to compile MySQL (it failed during configure phase).. On further investigation, the following code segment when compiled with phtreads, results in a segmentation fault: #include <stdio.h> typedef long long longlong; main() { longlong ll=1; float f; FILE *file=fopen("conftestval", "w"); f