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I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line

Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '7ad7ce9a09aebcce6e1c3e551fa4d401@192.168.1.1' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '06d1853a435e01fb05c3233d7aecba85@192.168.1.1' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on

Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

http://www.pcmag.com/article2/0,1759,1812887,00.asp Specifically, his assertion that ISP's would sniff traffic and block, say, the SIP port. You could play wack-a-mole with port numbers, no? Also a community based, Freenet style of encryption implementation for "free" VoIP traffic would address this issue. I raise this to the list because I'm sure there's a grain of

I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give.

Afternoon A little off-topic...but... Does any1 know why recorded call with "IAX2" in the filename are not displayed within ARI? LittleJohn's website isn't a helpful place for ARI (o; cheers rick

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

...ation that I have found is that I have to setup my rxgain to get a 14800 value with ztmonitor but that is not working. Any others suggestions ? Regards Ken -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dan Littlejohn Sent: 15 ao?t 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo calibration with ztmonitor and a test linefrom a telco On 8/15/05, Ken Dresdell <kdresdell@globetrotter.net> wrote: > Hello everyone, > > Does anyone have experie...

...late or updating to the appropriate ari.po page and returning it to me would be very helpful. German Greek Spanish French Hebrew Hungarian Italian Portuguese Swedish If you would like to translate ARI into another language, I would be happy to support it. Loaded into AMP CVS and also here: www.littlejohnconsulting.com?q=ari If you have a chance, take a look. Comments and suggestions are welcome. Dan 512.791.0137 www.littlejohnconsulting.com

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

Am trying to monitor and record an in-process phone call using a remote computer and the Asterisk Manager API. Have something that is working, but the call recording volume is to low to be usable. dialplan exten => 8159,1,ZapBarge(Zap/1) remote application with Asterisk Manager API $telnet->print("Action: Originate\nChannel: Local/8159\@default\nApplication: ChanSpy\nData:

Seems like other postings tend to think that saving recordings as files and not as blobs in the database are a more reliable way to go. Opinions on this? Looking at supporting it for ARI and judging interest. Dan 512.791.0137 www.littlejohnconsulting.com