similar to: Anyone else lose iconnecthere service in recent CVS?

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 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

I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.

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 am trying to use iconnecthere to make outbound calls. I am behind a linksys router. I keep getting this error 481 "Call Leg/Transaction Does Not Exist". Does anyone have any prior experience with this problem. Any leads will be much appreciated. Attached are the conf files and logs #SIP.CONF ; SIP Configuration for Asterisk [general] port = 5060 ; Port

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 =>

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.

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,

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

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

I've been using * CVS code from May of this year and was able to connect to iConnectHere and receive calls with * being behind NAT. Now that I've upgraded to 1.0 RC1, this no longer works. I've tried setting nat=yes in places, externip, et al with no success .. even though the code I was running from back then worked without that. Any suggestions? BTW, I've gotten DTMF from

I want to test outgoing VoIP Cheap, and they have a $5.95/month plan that would be perfect. I want it to interface through asterisk though, and I heard some providers are blocking asterisk. I just want to make sure that it still works before I sign up. Also, if you have any experience with them, let me know. Thanks, -Todd

Hi All, I can only seem to get iconnecthere working with incoming calls intermittently. One minute it seems to work, and the next it doesn't. I am not aware of anything being changed in the config files. Outgoing calls work ok all the time. The Asterisk box is behind NAT so that does complicate things slightly. However, the Iconnecthere PCPhone client software works perfectly for