similar to: Asterisk@home 1.0 + Sipgate UK/SIP Provider

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 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

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

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

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, 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

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 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 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

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",

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

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 Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott

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

Hello all, With some help from people in #asterisk on freenode, I've managed to get incoming SIP calls working. Outgoing calls however are however a different matter. My whole working (incoming calls only) SIPgate configuration can be found here. [1] When I uncommon what's in there, nothing works. There doesn't appear to be any useful error being logged , even when debug is

Hi, Now that I have Asterisk up and running I would like to find a good SIP gateway in the UK. I have looked at sipgate.co.uk and they look pretty reasonable. I am looking for peoples recommendations. Apologies if this is the incorrect forum for this type of request. Regards, -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint: