search for: zeltins

I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb:

I'll have MGCP hardphone that needs to dial pre-defined number as soon as it goes off-hook. So far I'm lost as to how (if at all) this can be implemented in Asterisk. Any pointers? TIA, Peter

I'm trying to set up * server behind NAT. The box is set up as DMZ in my DSL router, i.e. all incoming connections without explicit port mapping are forwarded to *. So far I'm unable to get this setup to work for either IAX or SIP (tried IAXComm & XLite softphones on public IP address). Data seems to come in fine (IAX/SIP debug shows message interaction taking place), but there is no

I'm still a newbie in Asterisk, just yesterday installed it for home use (so I can call home while travelling). Using AVM A1 (BT Speedway) ISDN card. Anyway, I find it very hard to locate supporting information for Asterisk. User's Handbook is still a draft, this mailing list is not easily searchable, and any info out on Usenet is scarce. I spent some 30mins just trying to find out naming

I've checked everything (pwlib + openh323 + asterisk) out of CVS, compiled, and chan_h323 module does not load with "undefined symbol _ZTI19H323AudioCapability". What could be the problem? Peter

--- Peter Zeltins <peter@fintrading.com> wrote: > > > Well, I happen to be one of those very specific cases... ;) and looks > like > will have experiment with it myself. Although I'd hate to re-invent > the > wheel. > > Peter Checking e-mail this morning it looks like we have...

My Asterisk box also does NAT for internal network, and establishes site-to-site VPN tunnel(s). As a result I have several internal interfaces with private addresses on them, and only one public interface. By trial-and-error I've found out that FWD (SIP) won't work unless I disable my VPN tunnels - it would send the internal IP address to FWD's SIP server instead of public one. I

Hello there Yesterday I updated my pwlib, openh323 and Asterisk from CVS. After making "clean opt" in pwlib and openh323 and make "clean install" in Asterisk i get an "Undefined symbol" error when I try to start Asterisk. As far as I can see its when loading the h323 channel driver the error occurs. Do I have to update other things as well, by reading the various

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter

Why wouldn't "SIP show channels" display lag & jitter, it's always 0ms? Is there a "deeper reason" for this or this is just something not implemented yet? Peter

What does this error message mean? WARNING[262160]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 2 frames I've been getting these a lot lately, sound quality seems to have suffered. I'm using I4L driver with Fritz PCI ISDN card. However, even the regular echo test sounds a bit choppy over LAN connection. Happens both on CVS and 0.4.0 versions TIA, Peter

Hi all, I've got my dirty hands on (free!) Vocaltec 4-port FXO/FXS gateway. It is used unit, I managed to configure correct IP settings in it but am somewhat at loss how to integrate it into my existing Asterisk network. I have no H323 gatekeeper, no Vocaltec Network Manager software, and am not familiar with Vocaltec products at all. While H323 support is compiling on my Asterisk box I was

I wonder what would be the easiest way to con Asterisk into logging all activity on ISDN line? Like incoming calls, outgoing etc, even if these calls did not originate/terminate at Asterisk server? I'm using chan_capi if that matters (it should), with Fritz PCI & S-type ISDN connection. TIA, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: