search for: waddington

When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk --> network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this

I have read the guides on using Messenger to connect via SIP. I just cant get it to connect, even inside the LAN. I enter <local ip address>:5036, it trys to sign in, but times out and says Service Unavailable. Do I need anything extra in my configs for Messenger to work? Have * admins managed to get this to work? Any help welcome. Thanks

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

When I saw the update for Cisco Phone RTP issue I thought I would try it. Unfortunately chan_capi wont compile on this update. Can anyone recommend a good * release for Capi, Bri ISDN and Cisco 7940's SIP 6.3. Or will CHAN_CAPI also be updated ? Running Eicon Diva Bri Cards. Error: chan_capi.c:1187: too many arguments to function 'ast_dsp_process'

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesn't

Hi. I am having the following warning when using music on hold. It works from X-Lite to Grandstream. I get a lot of errors and warnings. 1.Warning, flexibel rate not heavily tested! 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Thanks for any help. Full Output below: Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486

Asterisk Monitor seems to be working fine. Though the problem I am having is the two files (in & out) muxing. I added ,m to the string, yet the call records two files still, and I get the resulting error, at the bottom. monitor executing ( nice -n 19 soxmix /var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:4 8:23-in.gsm

Hi All, i dont what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS

I am trying to have the ring types different for internal and external incoming calls. I have followed the guide on the wiki, the SetVar executes, in extensions.conf I have it as s,1, Yet it doesn't work? When the phone rings, the ring type is the one I chose on the phone, it rings same tone for both when I test. Using Asterisk Stable. Anyone got this working and can

Hi all! I'm newbie, so here goes my situation: I have succefully compiled the cvs version as shown in asterisk website in some linux distros: Debian (2.4.22), Conectiva, Fedora Core 1 and in all of them, * starts and consumes all the cpu (on top). Does anybody know this issue? Thanks! Testa

...Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tracy R Reed Sent: 16 April 2004 19:20 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7940 no audio On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly: > When we receive or make a call to the outside - they can hear us, but we > cant hear them. I have had this problem several times and so far no resolution. However for me it has always been with IAX. I have been told that IAX is supposed to be NAT-safe but that does not seem t...

...(Thorsten Neumann) > 8. Re: Asterisk Process Stop After few hours (Michael Manousos) > 9. Re: software phones for Asterisk - is there a list? > (Tomasz Chmielewski) > 10. Re: Advantage of IAX2 to SIP? (Rich Adamson) > 11. Asterisk Call Monitor and soxmix error (Craig Waddington) > 12. RE: Avoided deadlock (Brian West) > 13. RE: CallerID on X100P in South Africa (Doug Reid - Stormcorp) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 1 Dec 2004 08:23:49 -0600 > From: "Roger Hanson&q...

Dear All, I have a working asterisk installation in the UK on BRI point-to-point. I am using Redhat8 with one Eicon Diva Server 2.0 card with chan_capi-0.3.5 and Asterisk 1.0.1. I have got to the stage where I can make and receive calls over ISDN. My question: How do I group the 2 B channels so that when one channel is in use, the other channel is availble to receive[make] an

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details

...x11drv: Remove some no longer needed wine 16bit headers. user32: Remove some no longer needed wine 16bit headers. Remove more unneeded wine 16bit header usage. Tijl Coosemans (2): loader: Introduce FreeBSD loader. libwine: Use GDT entry for %fs segment on FreeBSD. Trent Waddington (1): server: Handle existing timer replacement when no window handle specified. Vincent Povirk (3): shell32: Add SHPathPrepareForWrite and related constants. shell32: Add tests for SHPathPrepareForWrite. shell32: Implement SHPathPrepareForWrite. Vitaliy Margolen (5):...

Kudos to the CAPI developers. I would like to have multiple MSN's on my ISDN Bri lines. I see all the cool features but cannot find any examples or guides to build from. Currently running Diva Eicon Cards with CAPI from http://www.junghanns.net <http://www.junghanns.net/> I would like to route calls to sip phones via msn. Set up callgroups etc. Can anyone share

On say 2 out of 10 calls, when on a call, the Audio at our end will drop for about 5 seconds, we can hear them, they can't hear us. It doesn't happen every call, random, which is making it very hard to trouble shoot, I am guessing it has something to do with RTP stream? Nothing has changed this end, yet this has just started happening. Seems to happen at about 2-3mins into a

I have a problem with incoming calls being recorded after a supervised transfer. Call comes in, receptionist answers, caller put on hold, Asterisk Monitor is recording, caller is on Hold, Callee picks up the call, Asterisk Monitor Stops. All recorded calls are named CallerID to Exten. Receptionist sees the incoming PSTN callerID, yet when we get a transfer from the receptionist, we