similar to: rtp.conf and Asterisk as a sip agent/client

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 TOS bits 184 == Using SIP RTP CoS mark 5 ==

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

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 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",

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

On 11/06/19 13:29, Rowland penny via samba wrote: > On 11/06/2019 13:13, Sebastian Arcus via samba wrote: >> >> On 11/06/19 11:49, Rowland penny via samba wrote: >>> On 11/06/2019 11:38, Sebastian Arcus via samba wrote: >>>> >>>> On 11/06/19 11:07, Rowland penny via samba wrote: >>>>> On 11/06/2019 10:34, Sebastian Arcus via samba

On 21/08/17 13:39, Robert Wolf wrote: > > On Mon, 21 Aug 2017, Sebastian Arcus wrote: > >> >> On 21/08/17 10:37, Gedalya wrote: >>> On 08/21/2017 07:28 AM, voytek at sbt.net.au wrote: >>>> is there a 'preferred way'? should I tell users to use 143 over 993 ? or >>>> 993 over 143? or? >>> There is no concrete answer. There

Hi, when i include a sip configuration from another file in my sip.conf using #include /etc/asterisk/sip-sipgate.conf everything seems to be working. The peer is listed when i execute "sip show peers" and Status is "OK". But the peer is not listed using "sip show registry". I need to place the "register => ..." in the sip.conf to make it work. Is this

Hey, we're experiencing a weird problem with Asterisk 1.8.13.1 (1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via a PBX (sipgate.de) work perfectly fine, almost 100% of the time. However, calls that are routed to sipgate.de, which then routes the call back to our Asterisk instance are "silent" most of the time. What I mean with that is that even though RTP traffic

On 02/12/19 15:44, Rowland penny via samba wrote: > On 02/12/2019 15:32, Sebastian Arcus via samba wrote: >> >> On 02/12/19 15:10, Rowland penny via samba wrote: >> >> Thank you for the quick reply. I should have mentioned that these DC's >> are at at different sites. At each site there is only one Linux server >> - hence why the DC is also the file

On 02/12/19 16:53, Rowland penny via samba wrote: > On 02/12/2019 16:24, Sebastian Arcus via samba wrote: >> </snip> > >>> You should have 'vfs objects = dfs_samba4 acl_xattr recycle' >> >> Thank you very much for this - now it is working. This lack of >> permissions inheritance issue has been plaguing me for months - it is >> very

I've just bought an OpenVOX B200p ISDN card - and if I remember correctly from last time I used one of these - it is possible to use either DAHDI or mISDN with it. Are there any factors to consider when choosing which software to use? Is one better than the other - or does one have features which are not present in the other? I will be using it for a simple PBX, with 2 ISDN channels as

I can see in the docs that a domain rename is not recommended/supported by Samba for an already provisioned domain. However, what I can't work out is if this is not possible during the classicupgrade step either? Does this make any difference, or would it present the same difficulties as renaming an already provisioned Samba AD? It case what I'm asking is not quite clear - I have a

Hi all, we are working on a A to B basic Call scenario with early media. On that scenario we get a call from a PJSIP endpoint and we place a new call using ARI. On the created channel we receive a 183 Session progress where we have an announcement regarding e.g. the cost of the call (it's important for us to have this announcement to inform our customers about the costs). Using asterisk

On 11/06/19 11:49, Rowland penny via samba wrote: > On 11/06/2019 11:38, Sebastian Arcus via samba wrote: >> >> On 11/06/19 11:07, Rowland penny via samba wrote: >>> On 11/06/2019 10:34, Sebastian Arcus via samba wrote: >>>> I've just upgraded a Samba AD server to 4.10.2 a few weeks ago from >>>> 4.x (I'm afraid I'm not sure the exact

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' --

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= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

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