similar to: Inbound not working with iconnect

Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040808/ecc99c4a/attachment.htm

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

Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse,

Asterisk Users, I'm looking for U.S. providers that will provide access to the PSTN and allow me to easily use my Asterisk box with their services. I would prefer a provider that supports number portability, so that I can park my existing cell number on their network and later move it again, but I'm open to doing some funky stuff with call forwarding if I have to do that. Can anyone

This appears to have been the magic bullet for me. Thank you very much. So, the bottom line is that there is a bug that ends up making inbound calls use type=peer rather than type=user. Correct? > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paul Cheng > Sent: Tuesday, August 10, 2004 8:35

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 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 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

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

Hello all, I have a 7960 registered to asterisk. I am trying to use iconnect as my sip provider. When I send an invite to delta-three, I get the normal INVITE - 407 - INVITE exchange. The problem is, asterisk is sending the second invite using the 'dialed number' from the 7960 as the username, and not my 'username' configured in sip.conf. I believe that digest authentication

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

Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323

Hello, Recently I am seeing this message on my asterisk console received from Iconnect. Apr 30 11:37:21 NOTICE[1125329600]: chan_sip.c:5648 handle_request: Unknown SIP command 'NOTIFY' from '213.137.73.41' It is prety annoying as it appears once every four seconds. I've seen similar posts in the archives which points me to NAT keep alives being send by the remote end. I am

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

Residential Long Distance. One of my biggest pushes towards a VoIP provider was cheap long distance. Now in the U.S. at least with SBC they now have a plan for Unlimited Long Distance. The price is 30.00 a month if you do not have a couple of required features on the line already like Caller ID and a feature like 3 Way or Call Waiting etc. in which case that lowers the price to 20.00 a month.

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:

I'm looking to set up multiple inbound Voicepulse Connect lines and have Asterisk route them direct to different IVR or Voicemail based on the inbound number that is called. Unfortunately, I just can't see how one would go about identifying the number that is being called. Has anyone been able to do something like this with Voicepulse? I appreciate any assistance. Phil

My goal is to have only 1 primary phone number that can seamlessly "pool" multiple VoicePulse accounts. Let's say I have 3 accounts with VoicePulse Connect 212-555-1000 (primary) 212-555-1001 212-555-1002 When I receive inbound calls on 212-555-1000, I want to "forward" or "roll over" the connection to 212-555-1001 and 212-555-1002 so that the 212-555-1000

Hello- I'm obviously doing something wrong here in trying to get an inbound DID to work with voicepulse. I have an outbound context set-up for those calls in iax.conf, and the appropriate register in- statement. within extensions.conf I am doing something like this: exten => 212xxxxxxx,1,Dial(SIP/admin,t) (where admin is the phone i am looking to forward to from sip.conf). i'm

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be