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Sorry for re-posting this message - I am trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor. Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency. i.e a sentence which should finish in 4 secs finishes in much lesser time. Where can be the problem? any configuration issue? Thanks in advance. -------------- next

Hi Friends, I have successfully implemented Intercom, Voicemail and International dialing using Asterisk. Now I want to connect my PSTN Lines to Asterisk server. I have 3 PSTN number (lines) to connect to Asterisk. For this, I want to use Sipura SPA-2100. Is my decession is correct or not? Is there any disadvantages with this Sipura SPA-2100? Please tell me. Thank you. Regards, Chandramouli

...is not handled correctly in asterisk if you use any gateway other then asterisk. IE: you use a cisco or TNT as your gateway to/from the PSTN via SIP and asterisk to talk to the 2500 type phones. -larry > Message: 1 > Date: Thu, 16 Mar 2006 12:36:45 -0800 > From: Martin Joseph <ast@stillnewt.org> > Subject: [Asterisk-Users] RFC 2833 and SIP? DTMF? What am I not > getting? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <5ebf9b207fd5e5f2fc8dab598d4f24f6@stillnewt.org> > Cont...

...e VoIP wifi page (404 error) months ago and is the same today. Cisco.... well chan_sccp seems to have not been updated in 19 months and how feature-packed is chan_skinny? Has anyone encountered a WiFi phone that is reliable besides SpectraLink? Regards, Andrew On 10/25/06, Martin Joseph <ast@stillnewt.org> wrote: > > On 2006-10-25 15:00:52 -0700, "Andrew Joakimsen" < joakimsen@gmail.com> > said: > > > > > > > Also the Nokia E60 and E61 are hybird GSM/WiFi phones, when you have > WiFi > > coverage your calls will go over that technology an...

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

..., March 03, 2006 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hardware Requirements for 1M minutes Sorry, I saw that right after I posted. It is per month. And almost all during business hours. regards, David On 3/3/06, Martin Joseph <ast@stillnewt.org> wrote: > > On Mar 3, 2006, at 9:49 AM, David Thomas wrote: > > > I'm doing an install for a client with the following requirements. > > > > - 1 Million minutes of outbound calling > > Per what? > > _______________________________________________ &g...

...Subject: Re: [Asterisk-Users] SIP trunk problem Marty, But with the same 128 bit upstream circuit, directly connecting the SJPhone the Stun server and using ulaw, everything is perfect. The problem comes when i am putting Asterisk in the picture. On 3/25/06, Martin Joseph <ast@stillnewt.org> wrote: On Mar 24, 2006, at 1:19 PM, George Vagenas wrote: > Hi all, > > I have the following problem, working with a SIP provider, if i setup > my SJPhone to register directly to their STUN server and working over > a 384/128 ADSL i have a really good qual...

Probably just me being dumb, but I am trying to update my asterisk to the latest (1.2.5) When I go to my /usr/src/asterisk I type: make upgrade make install This seems to be doing it's thing, but when I type show version from the console (after a restart) I still get: Asterisk SVN-branch-1.2-r7231 built by root @ notdeadyet-imac.local on a Power Macintosh running Darwin on 2006-03-04

Converted a strictly VOIP system in NYC to NEC IPK TDM system... will have 25 Polycom 501's for sale. Best offer, offlist only please. R

I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?). I am currently running SVN-branch-1.4-r46716. Any ideas on why this might be, or how to figure out how to fix it?

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

Hi, I have a spiffy new gateway which seems quite promising. It's the Audiocodes MP114 FXS_FXO (2 of each). I have got it configured and working reasonably well, but have a couple of issues. 1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway after 10 seconds. This isn't asterisk saying it's quiet for 10 seconds, it's the gateway deciding it's time to go

Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

I have a strange problem with echo. My setup includes a Grandstream HT-488 which is both an FXO and a FXS. I noticed last evening that if I called the FXS through my asterisk box from my cell, the resulting connection was fine for me at the cell end, but produced dramatic and conversation ruining echo at the FXS of the HT-488. Today I realized that if I call out from the same FXS to my