search for: lateef

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

...(Z.Z.247.106) of the Asterisk PBX when it should actually (Y.Y.47.149) <--- Transmitting (NAT) to X.X.141.32:5060 ---> SIP/2.0 183 Session Progress^M Via: SIP/2.0/UDP X.X.141.32;branch=z9hG4bK87468d20000002f44b86a00400006f2b00000166;receiv ed=X.X.141.32;rport=5060^M From: "Irfan Lateef" <sip:2005 at Y.Y.47.149>;tag=327f290e2e7^M To: <sip:99084611234 at Y.Y.47.149>;tag=as24228e21^M Call-ID: 876BAA6B36F644F7B4EF7BE5D4B7E8BD0x87468d20^M CSeq: 2 INVITE^M User-Agent: Asterisk PBX 1.6.0.17^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,...

...;0x9106ef8', 10 retries! Jan 4 12:27:46 NOTICE[5482] chan_sip.c: stale nonce received from '111130 <sip:111130@212.xxx.xxx.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.c...

...tart 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: 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...

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 ----------------------------------------------------------- Please do not print this e-mail unless it is absolutely necessary. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-...

"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

...pile logs to see the flags. The question is how do I find the compile-time options that were used to build the asterisk binaries. Or the other question is, if you have come across this problem building other modules how do you generically solve it. Thanks in advance for any help. -Regards, Irfan Lateef

...MoE (Mike Hammett) > 8. SIP-H323 Help and Multiple Listening Port (Kenige Ho) > 9. RE: TDMoE (Alexander Lopez) > 10. Re: Mitel 5220 IP phones (tracinet) > 11. Polycom dialplan restriction (Carlos Chavez) > 12. SER + Asterisk (Nick Hoffman) > 13. OOH323 Configuration (Abdul Lateef) > 14. Re: Bandwidth: to seperate or not to seperate (Rich Adamson) > 15. RE: PRI indications. (Mark Edwards) > > > ------------------------------ > > Message: 9 > Date: Wed, 8 Feb 2006 23:59:18 -0500 > From: "Alexander Lopez" <alex.lopez@opsys.com> >...

...8. SIP-H323 Help and Multiple Listening Port (Kenige Ho) > > 9. RE: TDMoE (Alexander Lopez) > > 10. Re: Mitel 5220 IP phones (tracinet) > > 11. Polycom dialplan restriction (Carlos Chavez) > > 12. SER + Asterisk (Nick Hoffman) > > 13. OOH323 Configuration (Abdul Lateef) > > 14. Re: Bandwidth: to seperate or not to seperate (Rich Adamson) > > 15. RE: PRI indications. (Mark Edwards) > > > > > > ------------------------------ > > > > Message: 9 > > Date: Wed, 8 Feb 2006 23:59:18 -0500 > > From: "Alexander...

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,

...scking index file /home/cpmailuser/mail/cpanel.net/aaron.stone/storage/dovecot.map.index Jan 2 13:15:13 mx1.cpanel.net dovecot: lmtp(aaron.stone at cpanel.net): Warning: mdbox /home/cpmailuser/mail/cpanel.net/aaron.stone/storage: rebuilding indexes -- Jan 2 13:15:18 mx1.cpanel.net dovecot: imap(a.lateef at cpanel.net): Connection closed (LIST finished 0.121 secs ago) in=56, out=1805, bytes=56/1805 Jan 2 13:15:18 mx1.cpanel.net dovecot: lmtp(aaron.stone at cpanel.net): Warning: fscking index file /home/cpmailuser/mail/cpanel.net/aaron.stone/storage/dovecot.map.index Jan 2 13:15:18 mx1.cpanel.net...

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

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

Hi friends, How i can enable and disable RBT in asterisk for SIP users. We have linksys IP Phones but its give ring to the caller before ringing the called phone. -- Thank You, Code Lover

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 =>

Dear All, I am trying to recieve call from inbound proxy then route to internal peer (localhost) and then route to outgoing sip proxy but it failing with subject error. Can any one please correct me what i am doing wrong in below config. SIP.conf [Inbound] type=peer context=introuting host=184.107.XXX.XXX disallow=all allow=all [astinside] type=peer context=introutingB host=localhost