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> -----Original Message----- > From: Robert Webb [mailto:rwebb@ropeguru.com] > Sent: Saturday, March 05, 2005 5:24 PM > To: 'Asterisk Users Mailing List - Non-Commercial > Discussion'; 'leandro_tenorio' > Subject: RE: [Asterisk-Users] IAX2 (Variables) > > > > > -----Original Message----- > > From: asterisk-users-bo...

I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp 0 788 0.0.0.0:5060 0.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A

Hello all, For some reason I am not showing registration in SIP. Can anyone give me an idea what can cause this? asterisk1*CLI> sip show registry Host Username Refresh State -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/bd3a7577/attachment.htm

Hmmm.... Can't place calls... Can't access website... Neither of the 3 nameservers answer anything... Anyone heard/know something to explain all this?

Hi all, With the ability of an easy install using Asterisk@Home, I have decided to give it a try. It is my understanding though, that one cannot add zap fxs ports as extensions using AMP. Is there anyone using Asterisk@Home and have added any extensions as zap fxs channels? Would be interested in how you accomplished this. Robert

Sorry, this is off topic to asterisk itself, but is about the list server. I had a power failure lastnight at home, where my email server resides, and my network was down for about 20 minutes, that was after 45 minutes of uptime on UPS. Since power was restored, around 9:45 PM EST on 2/16, I have not received a single post from the users, biz, or dev lists. Normally when this has happened

Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 & 942's. Just curious about call quality, programability, and functionality with asterisk. I have read through the literature, but would like some real world feedback. Thanks

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

Anyone having trouble parking calls? I haven't tried it in a while, but it seems to have stopped working. If I dial 700, I get a invalid extension. I have "include => parkedcalls" in the correct context, and I can dial 701, which tells me no call is parked there. Any ideas? Parking.conf is stock.

Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. >From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley

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Is it me... or is it voipjet? This week I've been trying various providers, just can't seem to get voipjet to work. I signed up with voipjet but so far can't get it to work inbound or out bound. I always get 'all circuits busy'. May 12 22:27:05 VERBOSE[2442]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/101-ad89[0;37;40m",

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

Does the MWI feature work with IAX2? I have read where it should but cannot get the indicator to work on any of the IAX softphones that I have tried which have this feature. I even did an IAX debug and did not see where and indication was sent to the phone when it registered. IAX2 registration session: *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ

This is getting VERY annoying. Is there anyone in here that has access to the list administration to delete the user below??? >This is an automatically generated Delivery Status Notification. > >Delivery to the following recipients failed. > > asterisk-users@telappliant.com > > > >This is an automatically generated Delivery Status Notification. >

I was just browsing Digium's web site and noticed they are taking orders for the new IAXy. Has anyone purchased and tested one of these yet?? I have thought about buying one for testing, but want to make sure it isn't going to be a flop like the last one. Robert

I sent the below message out last Friday when the list seemed to be having issues. Never got any responses and not sure if it just no one knows or if it did not get through. Please don't flog me too bad for reposting... :-) ------------------------------------------------------------ Hi all, I am having some audio quality issues with a provider under sip. The issue I am having is

Me again... I have service with a company that does not allow for a BYOD plan. They will not give out credential or server info either. Is it possible to run the FXS port of the ATA to an FXO port in *? The service I have is throug Broadvox Direct using the Mediatrix 2102. I have tried this using loop start and kewl start. The * box sees the incoming ring, picks up and starts my dial plan. But

Ok, With all that has been going on with the list today I may be sticking my head out of my gopher hole and find a 12 gauge at point blank. But I am going to take the chance.... I have just started using * about a month ago. I have a small unit setup at home running all my PSTN and VoIP lines. I am using external SIP and IAX soft phones from work and everything is running. I want to do more,