search for: pushor

Hi Friends, Something I'd like to shed some light on if possible - how is it that a single ISDN conversation only uses 64K for bidirectional communication (using ulaw, correct?), but on several occasions now have seen references to ulaw voip conversations using 64K per side of the conversation, plus packet overhead (http://www.zytrax.com/tech/protocols/voip_rates.htm - seems to be down

Hi All, I have a new asterisk setup running at home and am very happy with it. One thing that I am trying to do is to take various actions in the dialplan *if* a particular phone is registered/authenticated/connected. For example, if someone dials *me* and is shows that I am connected via my softphone, to try it instead of my deskphone (and possibly notifiy the user in advance that it is

I have been reading a number of the past threads about G.729 licensing., about how the registration keys are linked to the network configurations, limited number of registrations etc, etc. Is there no reason why the decoding can't be done in with some Asterisk compatible hardware, so that once the adapter is bought, all licensing issues go away. In that way the owner could fiddle with the

...cific >than "Re: Contents of Asterisk-Users digest..." > > >Today's Topics: > > 1. RE: How to use multiple VOIP provider trunks (Damon Estep) > 2. RE: Asterisk on a dialup connection? (Kerry Garrison) > 3. Re: How to use multiple VOIP provider trunks (Tim Pushor) > 4. Re: Comedian Voicemail Issues (Matias G.) > 5. RE: How to use multiple VOIP provider trunks (Damon Estep) > 6. How to park/transfer a call received from a Queue? > (Wessel de Roode) > 7. pass caller ID to another application or machine. (Richard Reina) > 8. RE...

...isk ended with exit status 1 (Federico > Alves) > 2. Re: Re: teliax [Was: LiveVoip is Bankrupt] > (Rich Adamson) > 3. RE: Polycom & VPN trouble (gw@adcomcorp.com) > 4. Re: Native MoH patch for 1.0.8? (Juan Jose > Comellas) > 5. Re: Polycom & VPN trouble (Tim Pushor) > 6. Re: Re: teliax [Was: LiveVoip is Bankrupt] > (r00t) > 7. Newbie Confusion on Call Forward and > DBput/DBdel (Jeffrey Starin) > 8. Eicon equipment, BRI Server or PRI? > (gw@adcomcorp.com) > 9. Re: Level 3 SIP <--> asterisk (Max Clark) > 10. Re: Aster...

Hi All, I am trying to figure out how Asterisk determines which [user] an incoming IAX connection is for? Is it based on their source address? (I think possible) Is it based on their credentials (unlikely, I think, since we can set insecure=very) Also, for a [peer] section - when is the host= resolved? Is name resolution attempted every time the channel is opened? Thanks! Tim (Oh and sorry

Hi All, Does anyone know of a decent itsp that can provide a Scottsdale, Arizona DID, preferably with no 'plan' but just minutes used? Thanks, Tim

Maybe you didn't intend this for the list? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tim Pushor Sent: Thursday, August 18, 2005 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Set voicemail maximum length by context Hey, can you ask mom if she would watch the kids overnight one night when Heather gets back? Thats what I'd like to do for...

How embarassing. This was not meant for the list. My apologies.. Tim -------- Original Message -------- Subject: Re: [Asterisk-Users] Set voicemail maximum length by context Date: Thu, 18 Aug 2005 13:17:15 -0600 From: Tim Pushor <timp@crossthread.com> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> References: <015801c5a40b$d07262c0$8464a8c0@kc.visioninc...

Hi All, I am a remote office that is connected to my office via openvpn on UDP. Voip has always worked well (after discovering g729). Initially I used a softphone, then an analog set on a sipura 2000, then a polycom IP500 (I still LOVE this phone). At that point, I started noticing that the polycom doesn't ring a lot of the time. Since I was desperate for a phone, I didn't upgrade

Hi All, I am having trouble with receiving calls from FWD via IAX. I know this isn't a FWD support forum, but I suspect the problem is my asterisk setup. The problem is that I can dial out to fwd subscribers, even myself but they can't dial me using my FWD number. I don't know much about IAX, but it would seem to me like a registration problem, but I get no errors or warnings in

I have been able to setup three different providers successfully, but only one at a time. I would like to have all active in a fail over configuration so that one failing would not be noticed by the users. I know it's probably easy to configure but I have not been able to find out how. Can anyone give me an example? Chris Mason

How will this fare? I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP. My setup in the US is working already with a broadband cable connection. I am thinking that dialup may not work because of the bandwidth required

...(Goolsby, Daniel S (Daniel)) 13. Comments on Areski Calling Card Solution plz (Arnd Vehling) 14. IAX register confusion (David Cook) 15. Transcoding problems (Martin Sutherland) 16. Re: Asterisk@home not accepting IAX calls from outside (Mark Phillips) 17. Codecs and bandwidth (Tim Pushor) 18. RE: Teliax to VoIPJet (Wiley Siler) 19. Re: long pause on dialing.. (Randy Williams) 20. RE: swissvoice (Florian Overkamp) 21. Re: long pause on dialing.. (Giorgio Incantalupo) 22. Re: Memory leak in asterisk CVS (Erik Espinoza) 23. Re: SoftPhones: Bad, or just bad QoS? (Time Bandi...

I am tired of nat tricks, and would really like to run ser on a system that straddles the internal and external network, and send all outbound sip traffic to it (it would also rtp proxy). This would also give the huge benefit of actually being able to implement SIP reinvites some of the time, even though the * server is behind a nat. I know there's no outbound proxy support in chan_sip