similar to: Speex Problem

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

Well, I thought I had the problem solved. Ported everything over to PJSip and build RDNS records for the phones and the server, but I am still experiencing the problem on incoming calls. ** On 6/7/2016 1:00 PM, Faheem Muhammad wrote: > I've faced the same issue. The issue was related to DNS, the reverse > lookup query failure caused the delay around(7-9 seconds). The purpose >

I am having an issue with a couple of phones where they ring, but there is a long delay after the phone is picked up before the audio starts. My setup: * Server running Asterisk 13.9.1, Dahdi 2.11.1 w/ OSLEC * Server is CentOS 7 * Quad core CPU with 16GB Ram * 2 Snom 300 phones. * NO NAT. Server and phone are on the same subnet with only a gigabit switch between them. * Digium

Hi all, When trying to send a FAX I got the following error: Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/ 003228949469 at 80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534534 at 1.1.1.1.1 Where I should

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello fellow developers, I wanted to follow a more up-to-date llvm version and saw the google/stable branch that seems to be updated every two weeks or so. But it seems that the last update of the google/stable branch is from 2016-08-15 [1]. Did it move to somwhere else and I am lookint at the wrong spot, or maybe there is something terribly wrong with the current trunk? Regards. [1]

We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the

I'm having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO NAT*_ involved. Phones and server are plugged into the same network switch, all on the same IP range. The server is running a Wildcard AEX410 analog card with 2 FXO modules receiving incoming analog lines. Occasionally, in the middle of a call, the

Curious to know if anyone's created something similar to the following, if so and you'd care to share an AGI or dialplan, much appreciated. I will be eventually write a script to answer the nTH call. (if I can't find it (why reinvent wheels). Looking to do some testing sending anywhere between 50-200 calls to a machine. I'd like a Snom/Polycom/whatever to pick up after the nTh

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error:

Hi, I have been trying to figure out why the S100U is not performing very well on RH9.. Here is my thinking..( may be totally wide of the mark but here goes anyway) I remember reading somwhere that the sound system used by RH has changed... Does the S100U not depend on the sound subsystem?? So what I think is that the sound subsystem in RH9 and the S100U are not happy working together.. Does

I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that

Is there any possibility of DAHDI supporting Automatic gain control on TDM ports? I'm having issues at a couple of offices where calls made to local numbers are fine but a when a calls from or goes to a large percentage of long-distance or 1-800 numbers the person at the remote end cannot hear the person in my office. Boosting the gains in zapata.conf (I'm still using 1.4.21) to 8

Hey all, I have a client whose previous programmer ditched. I'm his webmaster, and now he wants me to have an asterisk system set up for serial number authentication...and I have a digium card from the previous guy. Are there hosts that will set this up for me? ie, rack space somwhere? Are there guides online I can look at? Thanks Razib

http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is working fine in my Setup, but I want Extern1 to talk to Extern2 directly whitout going over Asterisk as the uplink is slow. When I set for Extern1/2 canreinvite=yes it works, but "Intern-2-Extern" doesn't work because Asteisk gives out the private IP-Adresses of Int1/2 I defined localnet=10.0.0.0/255.0.0.0 (Private

Dear R Plots without par arguments do start not at zero (means, the box around is somwhere outside the specified plotrange). How to start really from zero, pe. basline y=0? every standart par works like this: x<-seq(1,10,0.1) y<-sin(x) plot(sin(x)) Thanks for help Marc -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden:

Hi list, I just installed the rcom package and tried to read/give out some values from/to Excel. Altogether it works great... but nevertheless I don't know how the syntax works or in other words: "Which command needs which parameters?" Is there somwhere a manual about this package with good examples? I've read the Package description... but there are not really good

Hi all, is icecast good for me? I am interested in delivering some audio + mpeg4 animation through the internet (live streaming). my mpeg4 animation is supposed to be smaller than the audio o my questions are: - would icecast, as it is, support a different kind of stream (my own type) or do i have to mux the audio and video before into a special format? or maybe it supports only audio? - it

I've gone ahead and updated both stable and testing for the branch. Sorry that it hasn't been updated in a while - I'll make sure it happens more regularly now. Thanks! -eric On Tue, Nov 22, 2016 at 8:25 AM David Blaikie via llvm-dev < llvm-dev at lists.llvm.org> wrote: > I think the process has bitrotted a bit and not sure it's going to > continue to be updated

This is not really a problem just something I noticed in my testing.. When two or more Asterisk servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2