similar to: Codec Selection

Hi all, I have H.323 Gateway, and i want to make a peer to route calls to this GW. But i don't know is oh323.conf supporting to add peer type entry with all feature. Please let me know how i can add H.323 GW type peer? -------- Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: abdulzu@hotmail.com GoogleTalk: lateef.np@gmail.com YM!: abdul_zu Doha

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 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]

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

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, 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

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

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, 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

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

Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [cybercall at ip-208-109-177-212 ~]$ asterisk -vvvvvvc -bash: asterisk: command not found [cybercall at ip-208-109-177-212 ~]$ I checked modules and other configuration files which are installed correctly. Please help me

Hi All, i have created one prepaid PERL AGI script to integrate asterisk users in our current Oracle Billing System. I am using $AGI->exec('Dial', $dialstr); in script after getting the MAX time out for the priticular call. But when the channels increase on my asterisk more than 50-60 asterisk get crashed and i am suspecting the cause is of AGI Script. because when i check ps on

Hi all, i am wondering if i can make two leg cdr in mysql cdr table. 1st Leg : Registrar the ATA which registered to the asterisk and it normally logging in cdr table. 2nd Leg : The CDR of carrier for the example if i send call like exten => _x.,1,Dial(SIP/${EXTEN}@AT&TIP) I this cause i can get the accrue duration of call because currently we are facing some call missing not coming

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

#define GET_BIT(lll) \ // blah blah #define G(label1,label2) \ { \ // decent amount code \ ... while (1) { \ GET_BIT(label2); \ }; \ } void f() { if (..) G('c', 'd'); while ( .. ) G('a','b'); } After perf sampling, a lot