similar to: iconnecthere 480 error: is there a workaround?

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, 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

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

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 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

Is there a service like iconnect that does allow dialing out more then one concurrent connection? Asterisk works great with iConnectHere, but they only allow one call at a time. I don't want to setup an account for each concurrent call, because it will make iConnect an expensive service, and besides, all of our calls combined doesn't reach 1000 minutes per month. Any ideas?

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 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

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

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

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

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, 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

I can't get the callerid feature to work when being passed through iconnecthere. Is it even possible to specify your own callerid using iconnecthere? -sip.conf- ... [iconnect] type=peer username=xxxxxxxx password=xxxx callerid="Jerky McJerkface" <(555) 867 5309> host=213.137.73.178 -extensions.conf- .... exten=>_1NXXNXXXXXX,1,SetCallerId,4168675309

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

I can't talk through iConnectHere. The connection gets made but as soon as any sound is transmitted the call ends and the Asterisk console shows an "Unsupported Media" error as follow: Got SIP response 415 "Unsupported Media" back from 213.137.73.147 My only allowed codecs are alaw and ulaw. My sip.conf looks like: [iconnect] type=friend secret=xxxx username=yyyyyyy

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