similar to: Does IConnectHere still work with asterisk?

They say on their website that they allow you to use your own device provided you give them the MAC address. Has anyone tried using * with it? Looks like they have quite a few rate centers and also phone support... Their website is horrible though... Just wondering, it'd be good to get user experiences from different providers other than IconnectHere and BroadVoice... -Chris

I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com

I am having problems getting the latest CVS right now. A cvs checkout asterisk -t gets to this part and sits forever: S-> server_register(fpm-world-mix.mp3, 1.1, , , , , ) S-> Register(fpm-world-mix.mp3, 1.1, , , ) Anyone know how I can just skip the file? Travis Conway EFS, Inc. Information Technology Desk:?? (334) 215-6551 Mobile: (334) 391-4450 mailto:travis@homeoffice.quikpawn.com

Hi all, This is my first post here - I started with Asterisk a few days ago and have "fallen in love" - fantastic product. I've only got softphones connected at the moment - I'll probably order the FXO/FXS cards in about a month (and then think about getting some hardware SIP phones). Our current phone system is quite a few years old and isn't growing with us (when a single

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

Is it possible to send the CallerId to IconnectHere with Asterisk when making outbound calls? I read somewhere that it doesn't work. I have set up everything to send the correct CallerId info to IconnectHere but I get a "442-887-926267" caller id. In [globals] ICONNECT1=1713...(my number) MYNAME=My Name I set up the Caller Id in the dialing macro: [macro-iconnecthere] exten =>

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

For the past several days I can no longer use iconnecthere with asterisk. It is broken in BOTH directions; I can neither make nor receive calls. On outbound calls I get an immediate error: -- Got SIP response 400 "Bad or Missing To" back from 213.137.73.140 On incoming calls, the call switches through OK, and for a few seconds I get audio in both directions, although much

Hi, Anyone having problems receiving DTMF on incoming iConnectHere lines? They disappeared for us sometime in the last 12 hours... And, yes, we've restarted * and rebooted our * machine. Michael Swan Neon Software, Inc.

I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my "plan" with them. Now all my calls get intercepted immediately, "We're sorry, but your account is temporarily unavailable." Incoming calls work just fine. I contacted their so-called

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,

Trying to update to the latest cvs, but Asterisk complained that zaptel was too old. Updating zaptel gives me this during the make. Any ideas, the searches and Wiki gives me no hints. cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB

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

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

I wanted to sign up for the pay as you go plan from iconnect anyway, and see they have the Cisco ATA for $99 and the Grandstream phone for $39.00 Anyone know if they ship these devices "locked"? I know iconnect seems pretty friendly about letting any sip device connect. What sucks is there is no way to contact this company if you're not a subscriber.. Zip, notta.. No email

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the

I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an <mailto:Asterisk@home> Asterisk@home installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant

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

Trying to figure out what's going on, CVS ident CVS-04/20/03-01:34:54. I get frequent errors such as this one, which showed up on the CLI interface within a couple of seconds of a cold start: WARNING[114696]: File chan_sip.c, Line 393 (retrans_pkt): Maximum retries exceeded on call 73015f757661435d247414b104964554@192.168.1.10 for seqno 102 (Request) All calls to iconnecthere terminate

I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped working. I made no changes to my conf files or anything else (other than the upgrade). Now I get a "Unsupported Media" error even though I'm still using ulaw and alaw. It stopped working with my softphone, my Grandstreams, my Snom, etc. Searched the lists for a recent discussion of this but