similar to: g.729 codec -- open source?

Hello, I patched asterisk cvs head sources with http://juraj.bednar.sk/work/software/asterisk/messaging/ and presnce patch without success. asterisk send "405 method not allowed" to sender. I use polycom ip300. Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger

Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there

Hello, I see, that Rails is quite english-centric. I am developing some webs, that are not primarily in English. I have a few questions: - besides turning of plurals, what should I take care? How to use utf-8 for all data and converting it from local charsets to utf-8? - how do I make my page multilingual (i.e. adding english support later)? Is there something like gettext support? Is

Hello, has anyone seen or created a Debian Sarge package for 1.2beta1? I saw some for Sid, but I prefer not upgrading glibc right now, as this is a production server (Asterisk on it will be for testing). Thanks, Juraj.

Hello, I am migrating one server to dovecot. The only problem is, that users have logins with @domain as part of their user name. I want to use pam auth (for other reasons, if only for dovecot, I would use mysql, but I need the same password db to be used for other services, like samba). Is there a way to allow this type of login? Just cut everything beginning with @. I can change the

Hello, I'm looking for a way to have G.729 codec working on Solaris/x86. Binary codec from Digium is not compiled for Solaris/x86 (only sparc). Are there any alternative (free or commercial) G.729 implementations, which would work? I saw something from Intel and got it to compile on Linux, but it was only for evaluation purposes, so we upgraded to commercial codec from Digium. I

Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client "subscribes" to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome.

Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. Regards, Sahil Gupta VoiceValley

Hello, I would like to ask, if there's a way to transfer a call from some external program? I would like to build something like Asterisk Flash Operator Panel, with the ability to transfer a call using drag and drop. So I would like to connect to asterisk command line interface and transfer one side of a call to someone else. Is this possible somehow? Thank you, Juraj.

Hi, We seem to be having some teething issues with a new Hylafax - happy to pay someone to complete the installation. Please contact offlist. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499

Hi, I'd like to know how I can playback a pre-recorded message to a user using our system without answering the call. I want to do the above in the scenario where the user dials a number and the number has been dialled incorrectly. Regards, Sahil Gupta VoiceValley

Hi, I have a Gateway running in "TE" (terminal equipment mode as "slave" that I need to connect to my asterisk server using a TE100P card. Can anybody give a quick run up of how to run the TE100P's in Network Termination mode to have this working sucessfully? Cheers! Regards, Sahil Gupta VoiceValley

Hello, I was again asked to try to add support for presence (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions: a.) are there any, at least partial projects, patches, anything, that at least partly implements presence and/or IM to chan_sip? I don't care about presence on other channels, I have one SIP client per user. I've read this list's archive several times and

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

I have created shell script for MRTG statistics of droped/rejected packets: ftp://slovakia.shorewall.net/mirror/shorewall/mrtg/ http://slovakia.shorewall.net/pub/shorewall/mrtg/ rsync://slovakia.shorewall.net/shorewall/mrtg/ example: http://slovakia.shorewall.net/pub/shorewall/mrtg/example/ It is not based on /var/log/messages (syslog), but iptables counter. A lot of packets are droped/rejected

Hi, When I ran wine for the first time, font metrics information is built. It doesn't take a lot of time, but when you are forwarding X session over half a world, this can be pretty slow and annoying. Well, I know that there is probably no way how to avoid this, but I think there is a way how to avoid this when you reinstall wine. So, question is, where is stored font metrics information

Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sip----ASTERISK----h323-----GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could

Hi! I would like to use the realtime extension of Asterisk and got the latest asterisk-addons from CVS. Upon compiling things, I got a couple of error messages from app_addon_mysql... is it me, or are the files in the CVS broken? Thanks, Christoph

Hi I'm looking for a company who can provide me an Internet connection between africa and Europe. Plesa If someone can give me some contact name or company dont hesitate to send me a mail at lmavericks@gmail.com Best regards

Hi List I am very new to Asterisk but have been alloted a job to replace my traditional PBX with it. Kindly provide me some useful info (PDF's etc) to setup Asterisk with FXO and FXS both. I have to cater some 60 users with 10 simultaneous calls. Regards _________________________________________________________________ Biography of Shah Rukh. His profile, awards, films.