similar to: Outgoing calls via Sipgate

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules..... and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever.... but the problem is now I can't get past the boot up and the

What happens if you take out the Zaptel I/F's? If it boots, you can correct whatever you did then replace them. hth -----Original Message----- From: Chris Earle (CBL) [mailto:cearle@cbltech.ca] Sent: Tuesday, February 28, 2006 7:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot boot machine up after working on zaptel.... Hi all, hard for me to explain this, but

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --

Hi, have some problem with incoming calls from sipgate. This was working in 1.4 but in 1.6 I get a 401 Unauthorized :-(. Sipgate has mentioned that I have to change the type to friend, but it is already friend, so what's wrong? Kind regards, Michael Here is the sip.conf: [sipgate_out] type=friend nat=yes username=1234567 fromuser=1234567 fromdomain=sipgate.de secret=secret host=sipgate.de

Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I

Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of "8|." to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk

Hi all, I have two sipgate accounts (numbers), if I have both accounts register only one will work for incoming calls (which is all i'm interested in). However if I disable either account the other account will work perfectly. Am I missing something obvious? Thanks in advance, Ray. Excerpts from sip.conf - [general] 8<---- SNIP! ---->8 Register => 1212121:aaaaaaaa at

Hi, I have a few accounts with sipgate.co.uk to get some different DiD numbers. However, when an incoming call comes in, it seems to pick the wrong peer from sip.conf which sends the call into the wrong context and it fails because there is no extension in that context to match the register. Using the config's below, if I dial the DiD on account 2222222, it works fine - picks peer 2222222

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is "we don't support SIPBroker"... So whats the easiest way to support SIP <> SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell

I could call a regular phone through sipgate. Now i can not: Failed to authenticate on INVITE to '"xyz" <sip:4xxxxxx@sipgate.net>;tag=as4ddd4a6f' A call from outside to my sip-phone through sipgate is OK. Can anyone verify ? Is it a sipgate problem ? greetings nicolas

hi anybody running with german SIPGATE? my configuration don't works :-( regards thorsten@gehrig.de

Hi, I have installed Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100xxxx. I configured my Asterisk server with 0207100xxxx. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I

Hi all I use sipgate and FWD but seem not to get it going. I do not have NAT on the asterisk box (static ip). The asterisk box has 2 network interfaces. One internal and one external. Now when I make an call to a FWD or SipGate number all I get is -- Executing NoOp("SIP/113-6d2e", "") in new stack -- Executing Goto("SIP/113-6d2e",

Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've

I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to recognize digits pressed on a keypad coming in from a Sipgate trunk. There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. Locally, if I call from an

Hi i have a trunk up and running with Asterisk and Sipgate.de and i can make call out but no call in but the Enddevice Status on the Sipgate Webpage says offline. Maybe somebody had the same problem in the past and can give me some hints ? Regards MArkus

Hi all, I have been running Asterisk for years (CVS-HEAD on 2005-08-24) with no problems save a failed harddrive. I have decided to build a new box and have Asterisk 1.6.2.2 playing nicely with mISDN after lots of changes to dialplan syntax etc. I am struggling with SIP trunks to sipgate.co.uk and dualtalk.com. Does anyone have a working examples? When I make an outgoing call I get... [Feb 5

I noticed that I'm no longer able to receive calls from PSTN to my SipGate DID number. I changed the sip.conf and extension.conf as per http://www.voip-info.org/tiki-index.php?page=Sipgate but the problem remains... However, I can receive calls from another sipgate user. The problem is only affecting calls from the PSTN (DID). Anyone with the same problem?