search for: sathyaw

...1.12.2 and you will be OK. There isn't anything wrong with your syntax, it's an OpenH323 issue. Michael. SW wrote: > anyone who can shed some light ? Or oh323 is completely dumped and I should > go to chan_h323 ? > > >>-----Original Message----- >>From: SW [mailto:sathyaw@sbcglobal.net] >>Sent: Thursday, November 27, 2003 8:28 AM >>To: Asterisk-Users@Lists. Digium. Com >>Subject: Help for oh323 >> >> >>Hi Friends, >> >>Hope you would help me out here, I have searched the asterisk >>user list for hours and also re...

Hi friends, I've noticed that first register message sent by * always get rejected by the destination sip server. Then * sends a second registration message ( with Autherization section, and that get accepted by the destination host). Why is this ? Isnt there a way to tell * to send with Autothorization message the first attempt ? Asterisk sends this first 9 headers, 0 lines 11 headers,

Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I make a bridge call (using .call files in outgoing/) I always get 'billsec=12' in the cdr, both mysql and Master file even if the call lasted longer, watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a

Hi, I am using * to function as the voice mail system for Vocal. Since I do not have a context in sip.conf file for each vocal client, I can't set the mailbox=xxxx in sip.conf. How do I get the MWI to a Vocal client ? Cheers Sathya

Greetings, I am progressing well with this great product, the *. SIP to SIP calling, Vocal to *, Voicemail all in the past. Did Iconnect, FWD etc. Also, purchased couple of FXO cards and did zaptel as well. It's time to get to h323 now. Read the mailing list for H323 and OH323 etc. need some help to where to start. Requirement is very simple, SIP calls need to be routed to a third party

Hi Gurus, I we seen references to 'codec pass through feature' in the mailing list. SIP to SIP and SIP to chan_h323 as well. Could someone help me to understand this feature, or point me to some examples etc. Appreciate any pointers here. Thanks a bunch Sathya

Hi folks, Sorry to post bunch of messages today, I am deep into this and hope any help from the experts. I am using the command Authenticate as explained in wi-ki: =================== Authenticate(password[|options]) Requires a user to enter a given password in order to continue execution. If the password begins with the '/' character, it is interpreted as a file which contains a list

Hi, I just applied four new g729 license to my * installation. Registration was successful ============== NOW, PLEASE ANSWER THE FOLLOWING QUESTION: Do you accept the terms of this agreement? yes(y) or no(n)y ...Please wait a few seconds Registration successful! ============== But, Now I cant start *, it comes up with the following error; [codec_g729b.so] => (Annex B (floating point)

Hi Gurus How do I make x100P does not answer incoming calls ? I want * play dead for incoming calls. I do not have any context for incoming calls from x100p, in zapata.conf. Call also get logged into the CDR, that too I do not want. I am using x100p for outgoing calls only. Any help appreciate. Cheers SW

>but if I disable hyperthreading (noht on the kernel line in lilo/grub) Could the same result could be obtained if hyperthreading is disabled from the BIOS ? Supermicro Motherboards (most Phoneix based) have this option Also, would we see same pops and clicks in a dual CPU box (Zeon) then ? SW Message: 1 Date: Wed, 07 Jan 2004 14:32:16 -0500 From: Jeremy McNamara <jj@nufone.net> To:

> Hi Folks, > > I have a situation where my Colo insists on a particular IP setup > for my * server box. They allocate two blocks of IPs to my colo > server. One set as my own (ex 20.20.20.20.4/30 - 4 ips) and the > other as a transit lan (es 10.10.10.0/29). These are all public > IP addresses and there is no NAT involved in. > > So essentially I have to set-up IP

Hi, I've tried almost every where to see whether there is a config parameter to set dtmf on IAX channels. Just like in SIP where we can set dtmfmode as inband, info or rfc. I am experiencing a problem with inbound DTMF where * interoperates a digit in a string as couple or more (intermittently). Like if I dial 665533, asterisk catches it as 6655533. Cheers SW

Hello, Recently I am seeing this message on my asterisk console received from Iconnect. Apr 30 11:37:21 NOTICE[1125329600]: chan_sip.c:5648 handle_request: Unknown SIP command 'NOTIFY' from '213.137.73.41' It is prety annoying as it appears once every four seconds. I've seen similar posts in the archives which points me to NAT keep alives being send by the remote end. I am

Hi, I have a nufone connection (IAX2), works fine. In my iax.conf I do not specify a time interval that * needs to renew registrations with nufone server. However I can see following registration messages on my cli every 90 seconds (approximately) --Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569 --Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569

Hi, No matter what I set in my sip.conf, I always get '3' as amaflags in my mysql cdr. (a) How do I make amaflags correctly set in mysql cdr (b) Seperate note, how can I set amaflags from agi Thanks SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040618/f45c4287/attachment.htm

Hi, I've just initiated a new bounty for the above; http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL Any takers or any contributors please respond to me privately. I do not know exactly how the bounty process works, but I can coordinate on this ? SW -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Really appreciate if someone who got astcc working lists the steps to make it work. I've got it installed and using the gui could get the database created. Would like to know how those two .conf files be populated and some pointers to the important fields in the database. Thanks Sathya -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, a.. uniqueid: Unique Channel Identifier (32 characters) I would like to know how is the unequeId constructed. ? I need a primary key in cdr database and would like to know whether I can make uniqueid the primary key. If asterisk channel does not have any idea of previousely crated ID's and if this id is randomly crated number then there is a chance that unequeid get a duplicate