similar to: iConnect Here PCPhone application and Asterisk

Hi, I wish to connect several ATA186 Phones to each other, to iconnecthere and to the PSTN using asterisk. Please tell the appropriate settings for firewall (ports to open etc.) sip.conf and extensions.conf(part relevant to iconnect). Also I would be glad to get a working example of your ATA186 configuration. I tried searching the mailing lists and several sites but did not find an answer.

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

silly me, I thought the inbound would be the hard part. how little I knew... can someone please give me any insight into why outbound is not working, in fact why trying to enable outbound fouls up everything? I'm using asterisk, most recent from cvs, I'm behind a nat, and I'm trying to use iconnecthere.com for outbound and inbound. Inbound is working fine, no problems. But for

This appears to have been the magic bullet for me. Thank you very much. So, the bottom line is that there is a bug that ends up making inbound calls use type=peer rather than type=user. Correct? > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paul Cheng > Sent: Tuesday, August 10, 2004 8:35

Folks, I have to admit that I MAY have changed something (at someone's advice) on a previous CVS head (May 28), but I'm not sure. I think that it had to do with changing "digest realm," but that may be a different issue. At any rate, I had both incoming and outgoing with iConnectHere. Now, I made exactly ONE change: I upgraded to the CVS head dated 7/30. I

Hello, I've subscribbed to "IConnect". I use it eclusively for outbound calling. I like the rates they charge but people I call complain about the audio quality. They say it sounds like I'm using a "cheap mic." or they complain about echo. The sound is very clean at my end. I'm using a Bundgtone phone with meadi routed through Asterisk to IConnect. It's

Hello, Recently I am seeing this message on my asterisk console received from Iconnect. Apr 30 11:37:21 NOTICE[1125329600]: chan_sip.c:5648 handle_request: Unknown SIP command 'NOTIFY' from '213.137.73.41' It is prety annoying as it appears once every four seconds. I've seen similar posts in the archives which points me to NAT keep alives being send by the remote end. I am

FYI: I'm getting a VERY strange set of responses from the deltathree/iconnect servers when I'm sending SIP calls to them. Looks like their load balancer has gone off the deep end. I won't include the extremely large debug here, but check out your current sessions with iconnect if you're having connection problems. Sometimes my calls will go through, but I'll get

I don't use my iconnect account much, and so I can't say precisely when outbound calls began to fail. But fail dramatically they do, and I'm not sure how to interpret what Asterisk is doing that their server is choking on. I get these messages in droves, even after I quickly hang up the call: -- Called 12195551212@iconnect Dec 1 02:37:56 WARNING[14865]: chan_sip.c:604

Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a

Hello all, I have a 7960 registered to asterisk. I am trying to use iconnect as my sip provider. When I send an invite to delta-three, I get the normal INVITE - 407 - INVITE exchange. The problem is, asterisk is sending the second invite using the 'dialed number' from the 7960 as the username, and not my 'username' configured in sip.conf. I believe that digest authentication

Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040808/ecc99c4a/attachment.htm

hi i got iconnect to work, works pretty well now except calls sometimes (more often than not) hang up after a couple of minutes.. heres a bit of the debuging Record-Route: <sip:61892142222@213.137.73.178:5060;maddr=213.137.73.176> From: sip:61892142222@natrelay.deltathree.com;tag=3281050172-73809 To: "JUSTIN XLITE" <sip:2001@61.95.68.84>;tag=as09766a78 Call-ID:

Hi, Anyone succeed using InternetPhoneWizard as a console device in Asterisk? I can only place some SIP calls from the iConnect Phone application, but will be great to be able to use it as a Console phone directly on the Asterisk box. Thanks, Dan

Hello All, I have gone thru all the resources I could find on google on asterisk + iconnect and managed to get outgoing calls working. However, I cannot get incoming calls to work at all. With the sip debug on, I can see that something is happening everytime a call is received from iconnecthere, but I get an invalid tone on the caller side. The call never rings anywhere on the asterisk. Would

Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB

Hi, There is someone on this list who has the specification for the IPW driver? I want to provide full support for it in DIAX Phone, but Actiontec does not answer to my mails. There is any windows driver available for the Digium's S100U USB interface? Anyone knows if it can provide full functionality for the connected phone, like callerid, callwaiting callerid, message indication, ring,

Hello my name is Ilija Poznic and I have a problem. My configuration is 1. Digium TDM4000P with one FXS. 2. AVM Fritz ISDN adapter (configured with capi). When I connect to my ISP and then start *. Asterisks is registering me to SIP provider iconnect. After that I can call international call trough VoIP. My problem is that I want to dialout to ISP only when I have a international call.

When I place a call using Iconnecthere or Nikotel as my sip provider, I hear no ringback tone when making a call. I have tried adding the 'r' command to the dial sting with no help. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to get my connection to IConnecthere.com working. I didn't have a register command in sip.conf at first, so I believe that is why it was not working. However, I can't seem to get the register command correct, it just keeps timing out. Below is what I have: register=<username>:<password>@natrelay.deltathree.com I know that there is supposed to be