similar to: sip trunk to Cisco router

On Tue, 2005-05-31 at 13:38 -0400, Jared Mauch wrote: > On Tue, May 31, 2005 at 12:30:07PM -0500, John Lange wrote: > > I am not a Cisco person; so the question is, is it possible to have one > > of the following: > > > > 1) Have the Cisco authenticate (register) as a SIP client to the > > Asterisk server. This allows me to place the Cisco in its own context. >

Folks, I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via standard wic-t1 card. The NEC needs to call two different asterisk servers with 4 digits. I have two way calling working with the one * box, but the other is perplexing me. Here's the layout * <--> Cisco 2811(192.168.13.1) <--> 1.54 point to point <- Cisco 3725(192.168.8.1)<-> NEC 2400. The

Hello, I'm having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends '@CUBE' onto the end of the dialled number, as per the following examples; Asterisk log; app_dial.c: Called SIP/test/0429123456 at CUBE chan_sip.c: Got SIP response 500 "Internal Server

Maybe someone could elaborate on my first question again. If the ip address changes while a REGISTER period, the ip address of the peer isn't been updated. How can asterisk update the ip address of the peer? > Am 31.03.2015 um 12:36 schrieb Daniel Heckl <daniel.heckl at gmail.com>: > > Hello Sebastian, > > I had already seen this list of the hosts, but it is not

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

David, I should also note; 246 is my extension, it has IP 172.22.3.238. 172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway. The trunk is called ?testing? at the moment. The route that selects this trunk uses a 9 prefix. This system is in semi-production, so there might be fluff in the log from other active calls. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten

Asterisk: 1.4.17 with sip realtime Openser 1.3.x Hi, I had this setup working fine until I try putting OpenSER in the picture as a proxy. Unauthenticated calls go to a PRI based app via a ZAP channel, calls to sip users get send to them etc. Now with a proxy in the picture asterisk asks the incoming calls for authentication "407 Proxy Authentication Required". It seems that the

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

Scott, thank you four your reply. I had already though about both options, but the problem is, that after an ip change AND a new registration the ip address of the peer is not updated automatically. INVITES are answered with 401. Only after a sip reload the peer works again. That can't be normal... Daniel > Am 31.03.2015 um 22:45 schrieb Scott Griepentrog <sgriepentrog at

On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: > Hello > > I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom > Germany. We have sometimes problems with incoming and outgoing calls. > I hope I can explain it understandable. Hello Daniel, I'll find myself in the same situation a few weeks from now :-) > > For example, Asterisk sends a

Hi Can anybody help me on this ? I am using Asterisknow 1.5.0-Beta(Freepbx) I am having a problem getting the sip trunks to register. It makes no different which provider one is using. Trunk name: callcentric Peer Details: context=from-pstn fromdomain=callcentric.com fromuser=1777xxxxxxx host=callcentric.com insecure=very secret=pasword type=peer username=1777xxxxxxx Register String:

I haven't been successful in getting this to work. The issue looks to be that Asterisk is wanting peer authentication for the invite request as it sends back 401 Unauthorized. I am using FreePBX 2.7 and have tested both Asterisk 1.6.1.18 and 1.6.2.9. My trunk settings are type=peer transport=tcp qualify=yes insecure=port,invite host=10.10.1.31 context=from-internal Here is snippets of the

Hello Sebastian, I had already seen this list of the hosts, but it is not active. All servers with which my Asterisk has been communicated are not listed. A port scan, to eventually update the list, found hundreds of servers provided in the address range 217.0.0.0/13 with open port 5060, some were even not found. I think there must be another solution. If I change insecure to

Im sure this has been thrown around this list 1,000 times, and Im sure its been around the net too.. But I have done everything, and cannot seem to get inward calls to be processed on my asterisk box.. First, Let me tell you what works: 1) Softphones (ZoIPer using IAX2 Protocol) Can make calls behind a Natted Firewall to the FXS Port, and it rings, and calls work full duplex. 2) Soyo IP Phone

Hello I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain it understandable. For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de <http://tel.t-online.de/>), the message is answered with OK and the peer is registered. Usually INVITES comes now from this ip address. All

Yes indeed. Do you have the dialplan, eg from /etc/asterisk/extensions.conf? Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 172.22.4.12. On 18 August 2015 at 16:21, Brendan Ord <bord at staff.onthenet.com.au> wrote: > Starting to make sense when I saw this line: > > > > [2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785 >

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install

We are using a Cisco router with a T1 card plugged into a PRI provided by a local telco (Allstream). This Cisco accepts calls and sends them to a couple of servers running Asterisk depending on which number was dialled. But there is a problem. When a call comes in to the Cisco from the PSTN it sends it to the Asterisk server something like this: FROM: 204XXXXXXX@<CISCO IP> TO: