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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

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

Try these guys: http://easyspeedy.com/ Haven't tried them, but when I was looking into a while back they responded quickly. -- Bjorn -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Sahil Gupta Sent: Wednesday, May 03, 2006 1:47 PM To: asterisk-users@lists.digium.com Cc: asterisk-biz@lists.digium.com Subject:

Hi, We are terminating around 60 channels on one of our Asterisk boxes, which the client sends in H323 mode. Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk The problem we face is that at random intervals the H323 process (as part of Asterisk) dies and can no longer accept new calls whilst Asterisk is still running happily. We have to then kill asterisk and start it

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

Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar.

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.

Hi sahil: I think the reason why you cannot get page shared is due to the gref you got. Gref is responsible for a page allocated from domU, in my understanding it should not be 0, that is a gref 0 can not be shared, that''s why I skip gref 0 to be nominated. The gref is nominated to Xen and later used to find a corrspond MFN, so it shall not always be the same.

Hi, We seem to be having an interesting issue with Asterisk whereby, it keeps routing calls coming in to the 'default' context.... regardless of what changes occur to h323.conf. <SNIP> [POP-A] type=user host=1.2.3.4 context=international </SNIP> == Starting H323/ip$1.2.3.4:12914/16313 at default,12126599878,1 failed so falling back to exten 's' == Starting

Hi, I am currently interconnecting to a PRI using a Cisco AS5350. I'd like to be able to dial specific numbers out by a specific isdn channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out via isdn channel one from the Cisco AS5350. If somebody would be able to guide on this, it would be appreciated. Regards, Sahil Gupta VoiceValley

Hi, I'm looking for someone to do time-to-time mantainence on some of our machines going up in New York. The person *MUST* be stationed in New York. Areas of expertise required: - Proficiency in Linux: Slackware, Fedora - Proficiency on Cisco Routers If anybody is interested, please contact me off-list. Regards, Sahil Gupta VoiceValley

Considering that I had a failure with compiling the latest version of asterisk because of cdr_mysql, I am going to assume that I need to have a copy of the mysql headers on the system in order to compile cdr_mysql.so. Does that sound correct? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi,I am looking for a reliable provider that can provide 3 dedicated linux servers asap. Unfortunately, the provider I have used for YEARS has become way too slack in recent times and we have to move on. Cheers, Sahil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081114/e06232f4/attachment.htm

I am running version 1.1.1 with mail_location: maildir:~/Maildir. This is working great as all our users have UNIX accounts with nologin shells. New domains (and their users) are about to come online and we would like to migrate to a setup with virtual mailboxes/users. From the wiki and comments within dovecot.conf, I see it is possible to do this piecemeal so both local and virtual users

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into