similar to: SIP INVITEs borked with iconnecthere

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 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 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 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 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

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

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

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

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 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

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 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! 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

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 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. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

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

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

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 =>