similar to: Sipgate.co.uk on Asterisk 1.6.2.2

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

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

Dear the following is the asterisk's dbase(Mysql5). if the extension =17171000 asterisk run appdata=222222, but I prefer to run appdata=3333333. let me know how I can run the appdata=33333 best Mani mysql> select * from ext; +----+---------+--------+----------+------+---------------------------------------+ | id | context | exten | priority | app | appdata

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 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

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

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

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 there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

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

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 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?

Hi everybody, I would like to know if it is possible to add a suffixe number to my official sipgate number? For example my official sipgate number is 123456 My local users numbers are 0001 0002 Could it be possible to compose from internet 1234560001 or 1234560002 , then the message could be forwarded to user 0001 or 0002. Actually I can only forward my sipgate number to only one user. So I am

Already Friday, the week went by in a Flash. If you haven't yet registered for a free Sipgate DID, I suggest you go do so. If you are in, or interested in the business, Sipgate has a few tricks up their sleeves and you should be aware of them. Someone from Sipgate will be joining the conference, which begins at 12 Noon EDT (9AM PDT, 10 Mountain, 11 Central, 5PM UK, 6 Central Europe) ZipDX

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

I have an asterisk server and I am trying to set the server up as a redirect server of all my internet SIP phones. My Asterisk server as his own internet IP address. At this moment I can make international calls to a IAX provider but I am now trying to setup a SIP provider as well And I get the following error -- Executing Dial("SIP/t10002-4666",

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

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