[asterisk-users] SRTP enabling
Hi everyone, I was trying to support SRTP in asterisk for our Linksys IP Phones to prevent of ISP blocking issue. I compiled successfully SRTP from http://srtp.sourceforge.net/srtp.html But i don't know from where i should start to configure in Asterisk. Could someone please give me the example sip.conf for the way how i can support? You replies will be high appriciated. Abdul __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and VAD
Hi all, does Asterisk 1.2.7.1 supporting VAD? because i am running my asterisk on VPS and i want to save badwidth. Khan, __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggesstion Required
Hi all, I want to setup asterisk box to do the following jobs. 1- 100 cuncurent calls 2- 1000 User Registration 3- MySQL Realtim 4- PerlAGI Here is my question could u please reply it: 1- No RTP only singnaling, Is it possible? Ans: 2- How much RAM? Ans: 3- How much bandhwidth per month with G729 Ans: 4- Proccessor? Ans: I will be appriciate for your kind of replies. Abdul, __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Multi Call Generation
Hi all, Is there any such as tools for multi call generation to test, how much call can be done via Asterisk? _ Best Regards, --- Abdul Lateef Nepal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Not Disconnecting
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 =
h,1,DeadAGI(/var/lib/asterisk/agi-bin/main-stop.pl)
AGI Script:
my $dialstr = $gwtype/$gwip/ . $dialednum .
|350|tTL( . ($credit_time*1000) .:7000:5000);
$AGI-exec('Dial', $dialstr);
Could please advice me how i can prevent such kind of
issue?
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[Asterisk-Users] Asterisk MySQL
Hi all,
I am using MySQL query inside my extentions.conf. i
have more than 200 agents using the same extentions
and i can see in each request asterisk try to connect
mysql.
My question is, Is there any way to make only one
connection for all users who is using the same
extentions.
Here is my example working extentions:
[mysqlt]
exten = _X.,1,MYSQL(Connect connid 192.168.1.65
username password database)
exten = _X.,2,MYSQL(Query r ${connid} INSERT\ INTO\
Userstabl\ set\ user=921)
exten = s,n,MYSQL(Disconnect ${connid})
Please advice me how i can make one connection for all
users?
Thank You
Abdul
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[Asterisk-Users] DID Provider via Asterisk
Hi all, I have my asterisk server in USA. and i want to be a DID provider, not the reseller from any other provider. i need to connect my server via T1/E1 line, after that i can sell the DID to my customers, and they can route the DID where they want. I do not have much information about DID, so i am not sure T1/E1 connection can help us to be DID provider. Please give me some information and some USA telecom web site, who can provide us these connection? Thank You Abdul Lateef __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum ASM400 FXO configuration
Hi All, This is my first day i brought ASM400 for Calling Card porpuse, I created AGI script for calling crad, so if some one is dialing 12345 our Calling Card AGI script will start to asking PIN,Phone number etc The Script is working well with SIPURA 3000. But i wanted to configure in quintum because this model is already having 4FXO line. So if any once can give me some usefull link or the idea for FXO configuration i will be appricate. I am looking the following diagram: PSTN FXO Line (Quintum) FXO Line [EMAIL PROTECTED] Thats all. Please help me for this issue. Thank very much in advance. Thank You Abdul __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Accept Unregistered GK Calls
Hi everyone, Could any tell me How i can accept unregistered Gatekeepers calls to my Asterisk Box? My customer is using another Gatekeeper and he want to use my Asterisk as a gateway for him to terminate the call using SIP protocol. and his Gatekeeper is not supported as end point to register my Asterisk Box. Here is waht i did the configuration but getting error: Error : SIP/2.0 404 Not Found sif.conf [from-SIPGK] type=friend host=cutomer_SIP_GK_IP_Address port=5060 nat=yes qualify=yes context=ivr-bal disallow=all allow=g729 extentions.con [ivr-bal] ;exten = _x.,1,Answer exten = _x.,2,AGI(ivr-bal.pl) Where ivr-bal.pl file is having very semple gsm file to play some voice. I will be appricate for your replies/ Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 Peer
Hi, i treid this OH323/ipgateway:port and working well for me. But i need to add some more featurres, like some of my H323 GW supporting only G.7231 codec and some one G.729 and others feature like rtptimeout etc So if i am direct dialing without these feautres, the GW are not able to handel my calls. Any more suggestion..? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 Peer
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: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OOH323 Configuration
Hi all, I am using OOH323 channel to dial our H.323 carriers. I downloaed it from the latest svn. this my extentions.conf how i am dialing to h.323 destination. exten = _x.,1,SetCallerID(700700) exten = _x.,2,Dial(OOH323/[EMAIL PROTECTED]) This is the error what i am getting in h323_log 05:32:08:878 Trying to connect to remote endpoint(:0) to setup H2250 channel (outgoing, ooh323c_o_1) 05:32:08:878 ERROR:Failed to connect to remote destination for transmit H2250 channel(outgoing, ooh323c_o_1) 05:32:08:878 ERROR:Failed to create H225 connection to :0 05:32:08:974 Cleaning Call (outgoing, ooh323c_o_1)- reason:OO_REASON_NOUSER I will be very thankfull if anyone give usefull hint. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : RE : [Asterisk-Users] Codec Selection
What will be the g729 and g723 codec capacity from Intel IPP liberary without License? Because still i am developing all billing and other application for asterisk so first i want to use these codecs for test once all our system become stable i will buy the license. S0 please let me know how many cuncurent calls can be handel using Intel IPP? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec Selection
Hi All, I have one Carrier which is supporting only G.723.1, how i can put in my extentions.conf to send calls to this GW using G.723.1, because for Clients i can specify the codec from sip.conf but i am little confiuse how i can give specific codec for carriers. your ideas will be appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI-get_variable(DIALSTATUS);
if ($discr == CONGESTION || $discr == NOANSWER ||
$discr == CHANUNAVAIL)
{
my $dialstr = $gwtype/$gwip/ . $dialednum .
|30|tTL( . ($crdeit*1000) .:7000:5000);
$AGI-exec('Dial', $dialstr);
$discr = ;
}
Any idea?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
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RE : [Asterisk-Users] Codec Selection
Hi,
Is there any special configuration for transcoding on
asterisk? Or Asterisk will do it automatically?
---
Olivier Taylor
Sun, 05 Feb 2006 11:51:51 -0800
Hi,
Just forget to choose the Codec on asterisk :(
Only solution is :
Disallow=all
Allow=YourCodec
If client doesn't have that codec you will need to
transcode on asterisk.
If client has that codec,asterisk will do pass-thru
and it will work.
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 20:00
À : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI-get_variable(DIALSTATUS);
if ($discr == CONGESTION || $discr == NOANSWER ||
$discr == CHANUNAVAIL)
{
my $dialstr = $gwtype/$gwip/ . $dialednum .
|30|tTL( . ($crdeit*1000) .:7000:5000);
$AGI-exec('Dial', $dialstr);
$discr = ;
}
Any idea?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
__
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[Asterisk-Users] How to Unregister?
Hi all, When i am using database show command, i can see more than 100 users are registered but actually they are not 100 some IP Phones are continue registered even i closed and switch off the IP Phone. Actually i am doing Windows based GUI, so i want to display all real registered users. I am using mySQL relatime for authuntication. I will be appriciate if any one can tell me how i can unregister so i will make some code to do unregisteration which ip phones are not registered. I will be appriciate for your replys. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gateway TIMEOUT
HI All, I have three a-to-z gateway from different terminators, I want to add in extensions some timeout condition. for the example my timeout=2 seconds if first gateway will not response in 2 second automatically it should dial using second gateway, respectively I will be appreciate if any can provide me the configuration how I should add it. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Gateway TIMEOUT
Hello All, Is there any idea please? HI All, I have three a-to-z gateway from different terminators, I want to add in extensions some timeout condition. for the example my timeout=2 seconds if first gateway will not response in 2 second automatically it should dial using second gateway, respectively#133; I will be appreciate if any can provide me the configuration how I should add it. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP IP Phone is not registering [urgent]
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, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Server Specification
Hi All, I was making plan to set an VoIP Gateway in India. And found some copanies who offered me to host my Asterisk server. I will be appriciated if anyone can suggest me how much simultaneous calls can be handeled with the following server specification? CPU : Dual Intel® Xeon® Processor at 2.8GHz Memory : 512 MB Hard Drive : 2 x 40GB 7.2K RPM Serial ATA Hard Drive Bandwidth : 100GB/MONTH HD Configuration : 2 Hard drives, Motherboard SATA RAID1 : Yes Port : 10/100MBPS SWITCHED VLAN Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some WARNINGS
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] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 17:58:49 WARNING[5478] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 17:58:57 WARNING[5478] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 12:27:46 NOTICE[5482] chan_sip.c: stale nonce received from '30 sip:[EMAIL PROTECTED]:1220' Jan 4 12:27:46 NOTICE[5482] chan_sip.c: stale nonce received from '30 sip:[EMAIL PROTECTED]:1220' Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco PGW-2200 OR Asterisk
Hi all, I need your golden openion about to set an VoIP softswitch. We decided to set Asterisk either Cisco PGW-2200 SS7/C7 PTSN SoftSwitch. Till now i am not fimiliar with cisco but Asterisk i did well configuration. My question is: Which will reliable to handel more than 600 cuncurent call with all kinds feature like CallBack,Calling Card,SS7 etc... I don't mean about the cost because Asterisk is open source and cisco is commercial, just i need to know which one will be better and why? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid
Hi, I am using SIPS softphoe. and i tested with another SIP Gatekeeper and i can see callerid in plain format. But when i am trying using Asterisk it is apearing callerid, username. So i don't think this is from client side or softphone. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Voip provider
hello, You can check this compnay. http://www.hatif.com Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk cdr mysql
Hi all, Did anyone installed asterisk-addons successfull? Becuase i am getting some error in installation. Error: cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:292: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 rm app_saycountpl.o Please help me how i can load this mysql cdr module? -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGIphp Installation
Hi friends,
I was trying to execute ring.php using AGIphp but i am not able to
ring another extention i am getting this error:
- Executing AGI(SIP/123456-6e57, ring.php) in new stack
Failed to execute '/var/lib/asterisk/agi-bin/ring.php': Exec format error
-- Launched AGI Script /var/lib/asterisk/agi-bin/ring.php
-- AGI Script ring.php completed, returning 0
Here is my Configuration
[sip] ; i want to ring this ext.
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
[ppp]
exten = 111,1,agi(ring.php)
this my ring.php code.
?php
require_once('phpagi-asmanager.php');
$number = '9745405022';
$asm = new AGI_AsteriskManager();
if($asm-connect())
{
$call = $asm-send_request('Originate',
array('Channel'=SIP/$number,
'Context'='sip',
'Priority'=1,
'Callerid'=$number));
$asm-disconnect();
}
?
Please anyone can explain me why i am getting this error?
--
Best Regards,
Abdul Lateef Khan
Computer Programmer
Mobile No. : +974 - 5405022
ICQ : 276-994-704
YM! : [EMAIL PROTECTED]
MSN : [EMAIL PROTECTED]
Google Talk : [EMAIL PROTECTED]
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Re: [Asterisk-Users] RE: return Credit Time
Hi, I already install the agiphp from the following steps, i want to be sure, is my agiphp installation is correct or not. i copied all following files into /var/lib/asterisk/agi-bin folder phpagi.php phpagi-asmanager.php phpagi-fastagi.php dtmf.php ;For test i crated one extention [ppp] exten = 111,1,agi(dtmf.php) These are all modification which i did for phpagi, Is another configurations need to be done to work properly? When i am dialing this 111 extentions i am getting the error: Nov 21 06:44:30 WARNING[8266]: Timeout, but no rule 't' in context 'ppp' i will be very thank full if anyone can help me. -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] return Credit Time
How i can return maximum credit time to terminate the call under his credit. In CISCO NAS i found h323-credit-time which is returning maximum credit time for calls when the call reached to this time, it will disconnect automatically. I did a lot of google but i am not able to find the commond which can return max calling credit. i will be really apriciate if any one can tell me this commond. -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Abdul Lateef Khan wants to talk to you using Google Talk
I've been using Google Talk and thought you might like to try it out. We can use it to call each other for free over the internet. Here's an invitation to download Google Talk. Give it a try! --- Abdul Lateef Khan wants to talk to you for free using Google Talk. If you already have Gmail or Google Talk, visit: http://mail.google.com/mail/b-96175a09a9-18176bcf70-0af60adddc366227 You'll need to click this link in order to add Abdul Lateef Khan to your Friends list and talk with each other for free. To try Google Talk (and get Gmail, a free Google email account with over 2,500 megabytes of storage) visit: http://mail.google.com/mail/a-96175a09a9-18176bcf70-dcb87e27db Google Talk is a downloadable Windows* application that lets you send instant messages to your friends and make free phone calls over an internet connection. Google Talk offers excellent voice quality and works with any computer speaker and microphone. Gmail is Google's free email service, offering lots of free storage, powerful spam protection, built-in search for finding your messages, and a helpful way of organizing email into conversations. And there are no pop-up ads or untargeted banners -- just text ads and related information that are relevant to the content of your messages. Once you sign up, we'll notify Abdul Lateef Khan of your new Gmail address and add you to each others' Friends lists so you can start talking right away. Gmail and Google Talk are still in beta. We're working hard to add new features and make improvements, so we might also ask for your comments and suggestions periodically. We appreciate your help in making our products even better! Thanks, The Gmail and Google Talk Teams To learn more about Gmail and Google Talk, visit: http://mail.google.com/mail/help/benefits.html http://www.google.com/talk/about.html (If clicking the URLs in this message does not work, copy and paste them into the address bar of your browser). * Not a Windows user? No problem. You can also connect to the Google Talk service from any platform using third-party clients (http://www.google.com/talk/otherclients.html). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: return Credit Time
Hi Are, Thank you for your reply, Actually i have my own billing system with freeradius which is running for our customers. and i wanted to integrate Callback system with our Billing System. So if i am going to use AstBill or any others billing system i cannot make connection to my real billing system. For this i start to work with Asterisk and PHP to work with my old database. First as i am begner in Asterisk i wanted to ask how i can include PHP file and retrive the value from PHP variable into sip.conf or extentions.conf. for the example as you give me the example to send max calling time, if i want to take this time value from php variable how i can define into Dial format, is this configuration will work? #include myphp.php Dial(SIP/70103-dc7a, SIP/70108|30|tTL($phpvar:$phpvar1:$phpvar2)|20) -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Forwarding
Hi all, I have one external VoIP terminator, I need to forward all calls to that terminator i did some configuration in sip.conf but i am confiused what will be the configuration in extentions.conf to forward all calls to that terminator. sip.conf [general] register = 450102:201079:[EMAIL PROTECTED]:5060/450102 i found that 450102 user successfully registered on terminator. Now i want to register Grandstreem using 450102 user on Asterisk Server and using this want to forward call using the same username to the terminator. [user] type=friend username=450102 secret=201079 fromuser=450102 authuser=450102 context=allcall allow=g729 extentions.conf [allcall] exten = Please advice me how i can run this configuration. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A-Z carrier Registration
Hi all, I have 1 a-z carrier i want to forward all calls to that carrier, can any one hint me where i should add this carrier information? I will be appricate if any one give me direction way? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP = H.323 Terminator
Hi all, I have H.323 Terminator and i want to terminate our all SIP clients to this terminator, Is it possible to add H.323 Terminator in Asterisk? Please give me a little hint os i can start to configure. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP = H.323 Terminator
Hello Reli, If i am going to install chan_h323 with different port instead of 1719 and 1720, is it will work? Becuase already i have MVTS (Mera Softswitch) which is running on 1719 and 1720 port on the same server. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changing 5060 port
Hi friends, I want to change the standard 5060 sip port to our any defined port. i made some change in sip.conf but it is not working, I have 2 softphone which are able to register with 81 port but the any kind of hardphone is not able to register using 81 port. here is my sip.conf configuration [general] port=5060 [123456] type=friend username=123456 host=dynamic port=81 ;the hardphone should be register with 81 port context=voip allow=g729 allow=alaw allow=g723.1 Please help me how i can register with 81 port? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Configuration
Hi friends, I am new in asterisk, i installed the Asterisk on my Redhat EP. But i am not able to register any SIP softphone. i am getting Unathurize message when in SIP debug. Here is my sip.conf configuration [general] context=default realm=asterisk port=5060 bindaddr=0.0.0.0 srvlookup=yes [123] type=friend username=123 secret=123 nat=yes host=dynamic ;port=81 reinvite=no canreinvite=no qualify=1000 dtmfmode=rfc2833 disallow=all allow=all context=inbound-from-local Please help me to find the problem. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/H.323 suggestion
HI all, Is Asterisk able to work as SIP and H.323 Gatekeeper same time? If it has the capability to work which i should open? Yours suggestion will be high appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallBack Suggestion
Hi friends, I am new in asterisk, i came for CallBack purpose, i read from Voip-info.org aboue callback with asterisk and i am near to collect all information about to start developing callback system. Just i have a samall question, Is Callback needs some special hardware? i have my PSTN phone number i want to call this number after two ring the call will be disconnect and the Callback will start to call back to the caller ID and it should prompt to enter pin id which will authunticate via freeradius.if the authuntication is valid it will give some beep for dialing the international number. Any kind of suggestion will be hearty appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
