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Is there a proper and accepted way to go about setting an E164 compliant caller ID (ANI) ? Currently, we're using just the Set(CALLERID(num)=XX) where XX is some E164 compliant number like 3539146632431 or some such. Is there another way we should be doing that or is that proper? N.

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

Hi, I've seen some hardphones or Softswitchs now support this sip.instance feature : http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt I don't really see any convincing use of this draft but I would be curious to share thoughts on it. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

...s > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20080925/461ed71d/attachment-0001.pgp > > ------------------------------ > > Message: 13 > Date: Thu, 25 Sep 2008 14:56:04 -0400 > From: SIP <sip at arcdiv.com> > Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <48DBDEC4.1000203 at arcdiv.com> > Content-Type: text/plain; charset=ISO-885...

We've just launched the beta of a free service which is, really, still only JUST out of the alpha stages. http://www.voipmagnet.com The basic idea is this: it's an opt-in directory focused on VoIP contact info (with elements of social networking and privacy control). Again, the service is very rough, but we'd like input from the VoIP community. There are a good many things that are

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk

We've been using RNK Telecom for terminations for our SIP service, but their billing is questionable, they've been in breach of contract multiple times, and when I brought it to their attention, Russ Man, our 'friendly' account manager told me, and I quote "If you are having that much of a problem..please find another carrier." With customer service like that, I've

Is it possible to record or playback a video file in Asterisk? N.

Look, ma... spam! We dun never seen that 'n before. N. Sunkara RaviPrakash wrote: > > Hi, > > Imagine a billion Indians together. > > Already 3 million Indians have chosen Indyarocks.com to bring India > together. > > I am already part of it and dont be surprised if you find most of your > other friends too :). Also you can send Unlimited Free SMS to your