search for: sipgate

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" <si...

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, th...

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 fromdom...

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.1...

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,...

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;...

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 I want to be able to do some SipGate to SipGate calls. As I said I can dial out on SipGate no issues, but I cannot get my *@home box to receive S...

...to port 51000-55999 to device 192.168.3.99 (same ports) other direction is totally open. I observed that all sip calls are closed exactly after 32s. call is disconnected on calling side as well... seems to be a timeout issue. here i have some debug logs. I see lot of requests from asterisk to sipgate.de, which are not answered. but communication is going fine in both directions (otherwise registration would not be possible?): <--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 ---> INVITE sip:2636146e0 at 80.142.13.32:55060 SIP/2.0 Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG...

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...

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TO...

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...

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...

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 made a call, Ast...

Hi all, I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall related, but I'm unsure. A registration to Sipgate is established successfully: <Registration/ServerURI..............................> <Auth..........> <Status.......> ========================================================================================== pjsip_sipgate/sip:sipgate.de:5060 pjsip_sipgate Registered C...

Hi people, My asterisk wont register with any sip providers, I have tried three different but they all end up with: Sep 26 17:36:36 NOTICE[114696]: chan_sip.c:4035 sip_reg_timeout: Registration for 'whatever@provider.tld' timed out, trying again There is no firewall and my server has a public IP. Could this be a Asterisk problem? -Fredrik vK

...LAN, behind a D-Link router/firewall connected to a cable modem. The 82.x.x.x is the IP for my cable modem. 192.168.0.101 is my router (internal IP). Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK19f25c03;rport=63817;received=82.x.x.x From: <sip:1437594@sipgate.co.uk>;tag=as4bc4badc To: <sip:1437594@sipgate.co.uk>;tag=019efdbfb68dd123f382ae5ce73ea92d.d391 Call-ID: 41b71efb79e2a9e37545e146515f007c@127.0.0.1 CSeq: 110 REGISTER WWW-Authenticate: Digest realm="sipgate.co.uk", nonce="42f4e13cd7f2b087ca91bbf35f3fb85814fc18e1", stal...

...g=100 mohinterpret=default mohsuggest=default callerid = asreceived immediate=no overlapdial=yes facilityenable=yes callprogress=yes group=1 context=isdn1 channel => 1-2 EOF /etc/asterisk/extensions.conf: [default] exten => _X.,1,NoOp(${EXTEN}) [isdn1] exten => _X.,1,Dial(SIP/${EXTEN}@sipgate,30,trg) exten => _X.,n,Hangup [from-sipgate] ;Skype issues exten => _X.,1,GotoIf($["${CALLERID(num)}" != "anonymous"]?notanonymous) exten => _X.,n,NoOp(Changing Caller ID number from ${CALLERID(num)} to 9999999999}) exten => _X.,n,Set(CALLERID(num)=9999999999) ext...

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

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 got...

hello ! My problem is: Astriks should create a connection to other members using a german Sip provider (www.sipgate.de). there are no problems with connections to: o Sip- Accounts o national phone numbers o mobile phone numbers but connections to international phone numbers DO NOT WORK (see the attached protokoll). The connection to international phone numbers does work when I directly use a VOIP hardware...