similar to: No RTP from asterisk?

I've been unsuccessfully trying to get nss_ldap to work. I've chased down hundreds of google searches over the last 3 days, and I can't seem to get a centos system to authenticate against ldap. Every daemon on the system is running into the same problem: nss_ldap: could not search LDAP server - Server is unavailable sshd, nscd, httpd, you name it.. slapd is clearly running, telnet

I have a Xen LVM based VM with two partitions: /dev/VolGroup00/vm01p1 /dev/VolGroup00/vm01p2 p2 is about 50GB, I gave it's LVM an extra 30 or so using lvextend. How can I extend the partition to fill my free space and the filesystem? I don't mind offline, if I can get Xen to boot off an alternative media. Peter -- Peter Serwe http://truthlightway.blogspot.com/ -------------- next

I recently came across the need to convert jpg images with IM, did a standard install of "yum -y install ImageMagick" and found that images converted with CentOS's base port of IM would actually corrupt the images, yet using the same (albiet different version, different distro) software didn't corrupt the images at all. By corrupted, I mean, the bottom portion of the image under

We've been going through some growing pains fumbling through porting some custom C modules that used to live under apache 1.3 into the 2.0 spec, even though none of us really considers ourselves a 'C developer'. We've basically got them all working now under the 2.0 API, and the question was posed about whether or not we should consider attempting to upgrade everything to run

I decided to blow out the FC2 installation I inherited on my workstation when I started here and installed CentOS-4.4-i386. Server side stuff I pretty much have down cold, but getting two heads on two seperate video cards doesn't seem to work for me. I'm using the built in Intel 865/810 card as primary, and an ATI Radeon 9200 PCI card as secondary. Initially, my system bios on this

I have a brand new poweredge 2900 with 10 SAS drives configured in two arrays via the built-in PERC 5 raid controller as: raid 1: 2x73GB raid 10: 8x300GB It's got 4GB of ram, and it's intended to be an NFS filestore. For some strange reason, logging in with ssh works great, it returns a prompt, all seems well. I go to run a simple command like 'top' or 'yum -y install

I've got httpd and mod_perl installed, but I seem to have an application that requires part of mod_perl that doesn't seem to be properly installed into perl, notably, the 'Apache' portion. To 'fix' this, cpan would like to install either mod_perl-1.29, or alternatively to install mod_perl-2.0.2. 1.29 doesn't seem smart, and 2.0.2 absolutely will not install over the

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

My full backup script errorred out the last two times I ran it. I''ve got a full Bash trace of it, so I know exactly what was done. There are a moderate number of snapshots on the zp1 pool, and I''m intending to replicate the whole thing into the backup pool. After housekeeping, I take make a current snapshot on the data pool (zp1). Since this is a new full backup, I then

I'm trying to sort a DATAFRAME by a column "ID" that contains alphanumeric data. Specifically,"ID" contains integers all preceeded by the character "g" as in: g1, g6, g3, g19, g100, g2, g39 I am using the following code: DATAFRAME=DATAFRAME[order(DATAFRAME1$ID),] and was hoping it would sort the dataframe by ID in the following manner g1, g2, g3, g6, g19,

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

Some of the run commandlines don't seem to be the *default ref* configuration. For instance, in 099.go, what's in the script is: LEVEL = ../../../.. BM=099.go ifeq ($(RUN_TYPE),test) STDIN_FILENAME = /dev/null STDOUT_FILENAME = null.out RUN_OPTIONS = 40 19 else STDIN_FILENAME = 2stone9.in STDOUT_FILENAME = 2stone9.out RUN_OPTIONS = 50 9 endif include ../../Makefile.spec95

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

Here is the setup: Phone A (in NYC) on own bandwidth. Phone B (in LA) on own bandwidth. Asterisk box in Houston,TX on own bandwidth. Both phones contact asterisk to register. Not much bandwidth used for this as it is a few packets every hour or so. Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk calls phone B. Both phones are connected and both people are talking.

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and

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

We are evaluating starting a small VoIP consumer based platform. What is the best codec to use with customers using primarily DSL as internet connectivity? I know that g729 is the king-all, but I want to know what the rest of the professional are using out there. g729 has a cost involved, so does the cost really offset the performance? Or is it better to go with g711 to start off? We plan

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?