similar to: Unsupported Media error from iConnectHere

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, 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 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

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 am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [root@asterisk sath]# cat g723.1 - Executing SetCallerID("SIP/-08122ae0", "1001") in new stack --

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

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 All, Outbound Iconnecthere calls work without any problem but Inbound calls are very intermittent. It seemed to work for a week or so but over the past week 99% of inbound calls are dropped to ICH voicemail. Would moving the Asterisk box to a public IP resolve the problem or is it just an ICH/Asterisk problem? I am registering against natrelay.deltathree.com. asterisk -vvvc shows an

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

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

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

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

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

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

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 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 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,

I have an asterisk Redhat 9 box running 4 hardphone extensions. Inter-extension calls are crystal clear. However when I dial out through my iconnect account I get a lot of jitter. At first I thought it was my nat gateway but after I programmed on of the hardphones (budge tone 100) for direct dial to iconnect I have clear voice transmission. I have no way of explaining this. My asterisk sip.conf

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

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