search for: abdul_zu

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

...ut 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 Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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

...x.xxx:1220>' Jan 4 12:27:46 NOTICE[5482] chan_sip.c: stale nonce received from '111130 <sip:111130@212.xxx.xxx.xxx:1220>' -------- 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 Qatar http://www.hatif.com __________________________________________ Yahoo! DSL ? Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com

...ing 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, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: abdulzu@hotmail.com GoogleTalk: lateef.np@gmail.com YM!: abdul_zu Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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

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 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 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

...Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <ea18e54a0801141603i2569a9d2i58011e5000fcfec at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On Jan 14, 2008 6:23 PM, Abdul <abdul_zu at yahoo.com> wrote: > >> 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 >...

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

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

"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

Hi all, We are running more than 40 active calls on our Asterisk Box. But some time we are facing problem, call is not disconnecting for a long time more than 2 and 2 hrs. in this cuase our customers charged for 1,2 hrs. even they made very small calls. i have already set rtptimeout = 60, but not disconnecting Here is my extentions. [main-ext] exten => _x.,1,AGI(main-ext.pl) exten =>

Hi all, I was coding for Callback application in Perl. I have small question to get the variable name of duration. I seen in CDR table of mysql there is two filed one is duration and second is billsec the billsec value variable is $AGI->get_variable('ANSWEREDTIME') But could you guys tell me the variable name of "duration" field. In this way i want to capture when the first