similar to: SIP/2.0 404 Not Found

Hi all, I am running an Asterisk server behind a NAT. I want to forward the calls from PSTN to a SIP phone (no nat and also an asterisk). I set the externip and localnet in sip.conf already. I opened the ports in my firewall. (I changed SIP port from 5060 to 5065 and limited the rtp port to 12000-13000) However, I just can't call out. I've always received SIP/2.0 404 Not Found. My

hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns "SIP/2.0 404 Not Found" any ideas ?

We are attempting to configure SIP trunking between asterisk 1.2.22 and a Mitel 3300 CX box. The Mitel machine will gateway to the PSTN for us. I found this earlier post about doing this from July: http://lists.digium.com/pipermail/asterisk-users/2007-July/191630.html Unfortunately the promised configs never came ;(. We're having the exact reverse problem: we can register with the Mitel

The hardware I currently have is: TDM400P with 3 FXO ports, and 1 FXS port 4 Cisco 7960 Phones (only 1 is currently configured for testing purposes) Asterisk on slack 10 I can dial out just fine via the Cisco phone, but when I try to dail in I get the following output when I load asterisk up in debug mode. -- Got SIP response 404 "Not Found" back from

Hi All, I have installed Asterisk Successfully and configure a out bound trunk for another SIP server so that if I'll dial 777123 from an asterisk-registered-phone then it will dial to the phone extension(123)-registered in the third party server. But my problem is that the reverse is not happening, that is I am not able to call from Third party SIP server to Asterisk

The configuration for kphone in sip.conf: [177204] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend ;regexten=1234 ; When they register, create extension 1234 ;username=xlite1 ;callerid="Jane Smith" <5678> host=dynamic ;nat=yes ;

I have a Sipura SPA2000 2 line SIP FXS box with line 1 on port 5060 and line 2 on 5061. The SPA2000 is on IP address 192.168.17.6, and the asterisk box is on 102.168.17.2. Both SPA2000 ports(5060 and 5061) share the same IP address. Every minute I repeatedly get the following output: SIP Debugging Enabled 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.17.6 SIP/2.0 Via:

Hi, I am trying to send "404 Not found" reply, without any luck with the following: exten => 555,1,Playback(you-dialed-wrong-number,noanswer) exten => 555,n,Playback(check-number-dial-again,noanswer) exten => 555,n,Congestion() However the above results in "500 Service Unavailable" being send out. What would be the correct application/function to generate "404

I have several SIP devices (Sipuras) that are working fine with *, except for one annoying little problem. Occassionally, after being registered for some period of time, the Sipura returns a 404 Not Found to (I assume) an INVITE request. Of course, this makes the extension appear busy. When this happens, I check the Sipura and it is thinks it is still registered and I check * and it shows

I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server was working. Doesn't fix it. I reset one of

With the help of one of the providers we terminate on, I've found the source of the problem of getting busy even when the called isn't really busy in the absence of ANI codes in sip headers generated by asterisk. If I put a NoOp(${CALLINGANI2}) in the dialplan before the dial I can see it holds the value '0', but seems that value won't find the way to the sip header. Is

Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer.

Using the Xten X-Lite client (free) I am able to connect to a local asterisk@home server and when trying to connect to the remote server (a mirror of the local) I am unable to connect. The first server is a local lan, the remote is using microsofts pptp vpn client to connect. Looking at the diagnostics from X-Lite I see this: <begin log> SEND TIME: 341549001 SEND >>

Hello All, I've been trying for some time to get Asterisk to register with a remote SIP gateway. I?ve recently managed to configure an SJ Phone to work with W2000 so know the configuration parameters work correctly. Asterisk doesn't authenticate properly and I notice that the authentication request appears different to SJPhone's. Do any tools exist to enable me to check these

We currently have an Asterisk server set-up, serving a handful of sip-phones and sipuras, and connecting to the outside world via one FXO and various SIP and IAX providers. In order to conserve bandwidth and have a fall-back in case our internet connection becomes unavailable, we're looking at putting * on a hosted server and funnel the calls to our office, such that: DID DID DID

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

Hello, I am doing some testing with my brother. We both have asterisk running with a Cisco 7960 locally and it works great. Using SIP between the asterisk boxes works great also. If I use IAX to call his remote extension, it fails because the remote asterisk server tries to use GSM to talk to the 7960. I end up going to his voicemail, which works fine. If he calls the same way it has the

> Asterisk Project Security Advisory - ASA-2007-010 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP

> Asterisk Project Security Advisory - ASA-2007-010 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP