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What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions?

A while ago SIP transfer via the # key on a call to a cell phone via iconnect was working. I updated to the current CVS tonight and now that functionality is gone. Any ideas as to how to enable it again? Thanks in advance -russ

Can someone else confirm that your phone does not recieve MWIs when using SIP and RealTime? Is this a problem with SIP or with Voicemail? -Matthew

Do you mind explaining why this bug was downgraded from grave (which had a justification) to normal? -- C'est avec les pierres de la loi qu'on a b?ti les prisons, et avec les briques de la religion, les bordels. - Blake, William -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size:

...Sergio Fern?ndez <sergio at wikier.org> Bug #605503 [wnpp] RFP: torrent-episode-downloader -- Torrent Episode Downloader Changed Bug submitter to 'Sergio Fern?ndez <sergio at wikier.org>' from 'Sergio Fern??ndez <sergio at wikier.org>' > submitter 606979 Antoine Beaupr? <anarcat at debian.org> Bug #606979 [redmine] redmine: more secure LDAP authentication Changed Bug submitter to 'Antoine Beaupr? <anarcat at debian.org>' from 'Antoine Beaupr?? <anarcat at debian.org>' > submitter 606982 Antoine Beaupr? <anarcat at debian.or...

Hello all, I'm having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I can't seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. Originally, I had configured Asterisk to run on the NAT side so that those within the office could connect easily, and those outside the office

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using