similar to: multi call iconenct?

Hi, I was able to connect asterisk to iconnect's service. It took me almost two hours, but it's because I was having NAT trouble. I finally discovered that you can set the iconnect host to natrealy.deltathree.com to make it work. (for those of you who, like me, don't have the time to search the archive I'll provide a working sample in a minute) My problem was sound

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

I am going to have to find a fix for this problem or I'm going to have to quit using iconnect. About one call in 10 or so, iconnect's gateway gives me an error (console output appended below). So upon receiving the error, which as a 4XX error means, "Fatal," asterisk gives up and drops the call. But not iconnect!! The phone at the other end starts ringing, and rings

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

Hi all. Can anyone comment why shouldn't we use FC 3 for an * production system? I'm not looking to start a distro war, but we just found out that redhat 9 (and FC 1) don't support SATA drives, and apparently FC 3 does. We are only familiar with red hat and are in a point in time that switching distros is not available. The guy installing the system is already on location. Yes, I

Hi all, I'm working on a setup for a small office. I'd like to use SIP/iconnecthere most of the time, because they're cheap. But they only allow a single call. When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse instead: exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN} Well,

Symptoms: when calling my iconnect phone number (13033913323 in my bogus example below) from my cell phone, I can see that the call makes it to my asterisk server, and my phones even ring once as * passes the call through during the "180 Ringing" period. However, it seems that iconnecthere.com cannot see my "100 Trying" and "180 Ringing" messages, as they

Hi, I just got a SPA-2000 in and was finally able to complete my asterisk setup. I'm making my outgoing calls through iconnecthere from the asterisk server however I'm running into a problem when placing calls. I can connect fine but when the person (or answering machine) picks up I hear them talk for a about half a second then there is a half a second pause or muted period and then the

This may be slight off topic, but perhaps it has relevance: My iconnecthere account no longer works for "inbound" calls through NAT using the standard configuration that they provide on their website. I have sent them a message, but I believe it will be flushed down the toilet by the first-tier support people. When I call my iconnect number, it goes directly to voicemail. There

How do you suggest doing that? How can I convert wav files to gsm files? thanks Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: shoval@softov.co.il Mobile: 972-55-229220 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031102/f84d7805/attachment.htm

Is it possible to use * as a VOIP gateway? Can I connect asterisk to one of the trunks on my current PBX and on the other side of the world connect another * to the trunk of another regular PBX - is it possible to transfer calls from here to there? I guess I'll need one port FXO card for each asterisk, but I can't figure how to configure the thing. I know I'll need to

Hi, Anybody have an idea how to integrate * with a Panasonic td-1232? We one at the main office, and are installing * in a branch office. We'd like to be able to make calls from * extensions to Panasonic extensions and the other way around. Making outgoing calls from extensions one one side to lines on the other would be nice too. I can put another * machine at the main office, but what is

Hi all. Can someone recommend a good UPS for using with an * machine that provides some linux tested software to do managed shutdown in case of power loss? Thanks. Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

I have several Linux machines some running on really old hardware and some on brand new, some run old distros (RedHat 6) and some new (FC3 or CentOS). All of them experienced power failure more then once, none of them has failed to load after a reboot. BUT, Asterisk is running your PBX. Your PBX isn't your proxy server, it isn't your web server, mail server, firewall, or whatever

Hi, Where can I find a complete * manager api guide, the one one wiki is missing informations like the monitor function for example, Thnx Serge

Hi, We've got IAX softphones, GrandStream VOIP phones and zaptel connected analog phones. Caller id, internally, works just fine (as long as I use numeric only callerids) for IAX and grandstream. Is there a way to have the analog phones' LCD display show the caller id? These are plain old regular analog phone, that if I had callerid from my telco would show on the screen. thanks

Hi all I am trying to figuure out how to get iconnecthere incoming calls to work outbound works fine but incoming goes nowhere but to my iconnecthere vocemail if I do a sip show registry it shows up as regg'ed nnn=is my iconnect here number xxx is my secret Thank you Jeremy [general] qualify=no register=NNNNNNNNNNN:XXXX@iconnecthere/NNNNNNNNN context=default bind = 0.0.0.0 port=5060

Hi thanks to everybody who responded to my earlier post. I have looked at all the material and links provided and tried everything in there, but it simply won't work for me. My SIP phones register with Asterisk, but they cannot be called (everybody is busy at this time) nor can they call anything (error code 4, whatever that means) not even internal (yes I did give them appropriate

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.