similar to: Error SIP/2.0 488 Not acceptable here

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

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

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 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

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' --

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

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 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

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.

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

On Thu, 2016-08-11 at 20:41 +0200, Benoit wrote: > Thanks Andrea, >  > Yes I confirm to you that I have enabled libvirt as well. >  > I don't see any error right now but I have to investigate a little bit more. >  > My guests are in autostart in virsh so everything is fine on this, the  > only issue I got is in case of shutdown. >  > The strange thing is that

Let me try to be more specific. The x y coordinates are different because of NAs in the dataset. In this analysis, a set of hat values (a measure of influence in regression) is given for each observation. On the basis of the regression that was run to get these hat values, the sample size was 1164 (one removed due to NA). The length of the data set is 1165. If I remove the NA from the

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

http://bugzilla.mindrot.org/show_bug.cgi?id=1164 Summary: scp with spaces username no longer works Product: Portable OpenSSH Version: 4.3p2 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: P2 Component: scp AssignedTo: bitbucket at mindrot.org ReportedBy: dtucker at

https://bugzilla.mindrot.org/show_bug.cgi?id=1164 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Blocks| |1930 --- Comment #4 from Damien Miller <djm at mindrot.org> 2011-09-06 10:34:22 EST --- Retarget unresolved

https://bugzilla.netfilter.org/show_bug.cgi?id=1164 Bug ID: 1164 Summary: FTP NAT fails in a specific scenario after upgrade to kernel 4.7+ Product: netfilter/iptables Version: unspecified Hardware: x86_64 OS: Fedora Status: NEW Severity: normal Priority: P5

Hi, While running the sample application controls/control.rbw I encountered the following error: controls.rbw:1064:in `onAbout'': undefined method `free'' for #<Wx::BusyCursor:0x2 827db0> (NoMethodError) from controls.rbw:1039:in `initialize'' from controls.rbw:1039:in `call'' from controls.rbw:1164:in `main_loop''

https://bugzilla.mindrot.org/show_bug.cgi?id=2443 Bug ID: 2443 Summary: Bugs intended to be fixed for OpenSSH 7.1 Product: Portable OpenSSH Version: -current Hardware: Other OS: Linux Status: NEW Keywords: meta Severity: enhancement Priority: P5 Component: Miscellaneous