similar to: sipgate doesn't work with sipgate anymore

hi, let me explain in detail, what i have configured and what is happening now: 1st router w724v (Deutsche Telekom AG): - port forwarding, everything to destination port 51000-55999 to device with ip 192.168.2.50 (interface of 2nd router) 2nd router Bintec RS353j): - configured NAT, everything to port 51000-55999 to device 192.168.3.99 (same ports) other direction is totally open. I

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 (217.10.79.9): icmp_seq=4 ttl=57

Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my

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

Hi everybody, I have strange problem: I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone using Zap Channel) using sipgate to a number in public network. When I'm hanging up before the other person picked up the phone, the line is not closed correctly. The phone keeps on ringing until timeout (of Sipgate I assume) and it even costs my money, if the other person

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.......> ==========================================================================================

ok, solved the firewall issue. A first test call worked fine. Another one now still gets disconnected after 32s. But in FW there are no blocked packets anymore?! And I don't understand why the registration to the same IP and same Port is working, but not later transmission of further SIP packets? that doesn't sound logical to me. What do you think? regards, andre

Hi all, I have a strange issue, with a some kind complicate architecture... A router of our internet provider is in front of another bintec rs353j router, at which my freepbx installation is located. However, NAT etc. seems to work fine. BUT: Something is not working...: When registering my sip-trunk towards my provider (3 different providers, all behave comparable), everything works at first.

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

Hi all, i hope i''m not totally wrong on this list. I setup a NAT router with the help of adsl-setup and shorewall. I''ve got a ppp link to the net and shorewall created the iptables. after a reconnect of the ppp link i get a new ip-address, but as long as the existing kernel udp mappings| which were create by outgoing udp traffic| don''t get timed out, the router

How can I see which codec is in use during conversation. I can see (for example) which codecs are negotiated before SIP connection, but I don't know which is chosen: 12 headers, 12 lines Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 217.10.79.30:15666 Found description format GSM Found description format iLBC

Hi everybody, I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk. It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers. The Optipoint shows "no Server..." (Registrar?) in Display. Sip debug shows no unusual (to me) Messages. Sip show

Hi there, I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with my asterisk server. I already changed the name of the user to "anonymous" since it looks like the phone sends that name. The WiFi Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200 What is it that I am missing? Any help very much appreciated!!! The error message I get is: Jan

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' --

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log