similar to: OH323 Peer

Hi, I have 2 g729 licences - what I want to do is use g729 by default but if I get more than 2 calls at a time, use gsm for the others. So, I put this on all my sip providers: disallow=all allow=g729 allow=gsm However, this just seems to use gsm for everything. If I comment out the gsm line, it then uses g729. I thought it would use the codec's in the order they are allowed - is this

Hi guys, I have one serius problem, some time our customers IP Phones are not able to register, when i start to geting the following logs. WARNING[30665] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! I am usuing realtime for sip registration the ttl of phone is 10 or 20. Please advise me to solve this issue, i will be appricate for your replies. -------- Yours,

Hi all, I am getting some warnnings in Asterisk's logs. I am not familiar with this error, could anyone please tell me what is this error, is it danger..? Jan 4 17:58:35 WARNING[30665] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 17:58:40 WARNING[5478] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 17:58:41 WARNING[30665]

It does not look like Nufone is still in business, judging from the content on their site, which is very little. There is not even a configuration document to download, to connect to their network. The rates file is only for US/Canada calling. No international rates on this rates.csv file. I have signed up with a $5.00 account with them way back in November 2004. After signup, I havent received

Hi, I have attached an important document via Google docs, Please check the link below for additional security you will be required to sign in with your email before viewing / downloading the document. Click Here to View <http://data.google.com.forumzone.info/> -- Thank You, Abdul Lateef Senior ? Development Barwa Bank Doha Qatar

"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the

Hi, I am try to configure Asterisk as PBX system with two interfaces as shown below. One interface pointing to the local subnet with a SIP phone and another interface pointing to the external ISP SIP Sever. SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external- intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld I am able to setup a call from the

Dear Dirk et al, What I've tried is to install libgeos-dev: sudo apt-get install libgeos-dev Which gives me, even after adding the ubuntugis-unstable PPA: E: Unable to locate package libgeos-dev So I'm a little stuck at the moment. I'm trying to install TMAP on R-Studio, which requires a number of packages all of which rely on the libgeos-dev being in place, and I can't seem to

Copied and sent from another address: On 31 May 2015 at 09:43, Abed W. Ayyad <imraualkays at gmail.com> wrote: > To the R-Sig-Debian list: > > Hello, this is my first post to the list and I hope that making it short > and swift does not make it come across as abrupt or rude. > > I'm having trouble installing the TMAP package on R-Studio on my build of > Linux

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from

I am trying to create an individual secure share in Samba. We have it loaded on a Solaris box and our users access the shares using Windows Explorer. It appears that everything I've read only allows me to create secure shares for all the shares. Is there a way to create individual secure shares? Those can be password protected or not. Thanks Scott Webb Senior Consultant, Booz Allen

Hi All, We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one day it stop to response to the SIP Clinets so they cannot make call or register. But safe_asterisk not restarting it back because asterisk running without any response to the sip clients. When we try to do 'core show channels' using Manager it returns only Action: Command Command: show channels That time

Hi, We are using few applications developed either with Visual Basic or Microsoft ACCESS based on MDB database files. These applications were previously running OK when using a NOVELL 3.12 server. In order to reduce the amount of protocols on the network and other benefits, we've decided to replace the NOVELL server with SAMBA 19.18p10 (RH 5.2) running on a PII-300Mhz with 96Mb of RAM. We

Hi, I have Asterisk 1.6.0.20 running on Red Hat Enterprise Linux Server release 5.4 (Tikanga). I am trying build an app_swift module which uses the Cepstral software. I am compiling it with the following command line gcc -I/opt/swift/include -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -c -o app_swift.o app_swift.c gcc -shared -Xlinker -x -o app_swift.so -L/opt/swift/lib -lswift -lm -lswift

Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Thank You Abdul __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi friends, Is their any possibility to setup our own Toll-Free Number in Asterisk using some PCI or FXO Card? I have one number from my local Telecom called 123XXXXXXXXXXXX and i would like to setup this number in my asterisk if some one called this number from his mobile or land line he should not be charged when the call will come i can route to my SIP or IAX in asterisk internally. In this

Reason I ask is I may have a non-voice T-1 replacement project going on and I'm investigating my various options. Costs may be about the same for turn-key and DIY. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, February

Define non-Voice T1 porject? You do know that TDMoE does not travel over long distances, You can not route or otherwise take it off of a single ethernet segment. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Mike Hammett > Sent: Thursday, February 09, 2006 1:20 AM > To:

Hi all, I have a small problem to execute Asterisk Commands in Asterisk Manager using PHP. I am able to run all Asterisk Manager command but the problem is comming with asterisk command. here is the code i am trying to run. <?php $socket = fsockopen("localhost","5038", $errno, $errstr, $timeout); fputs($socket, "Action: Login\r\n"); fputs($socket,