similar to: sip hangup detection problem

Hi ml, this is my problem: I have an Asterisk on remote site (my office) and two x-lite at home behind a ful cone nat. Both my ua can register, I can place and receive calls from both the phones and I can hear voice, so I don't think I have nat problem but when when i place a call if the called party hangup, calling party doesn't receive the signal and it stays connected. I also

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Hello, This morning I just started to have this problem calling from a SIP phone to a regular phone, using one of the 4 BRI cards (passive) I've in my * box. It calls regularly, but somewhere after 8-10 secs, it random hangups, or it hangups immediately after a hold, and so. I've looked into /var/log/asterisk/messages, and this is the output corresponding to the hangup: ##### Jul 30

Hello folx! I just started to play with *. I first installed it this past weekend on my Solaris 9 ultra 5 test box. Now I'm attempting to put it on a freshly built Linux box a mere few hours old. I've installed asterisk 1.2.7.1, libpri-1.2.2, zaptel 1.2.5 and the latest asterisk-sounds 1.2.1. This is running (or installed) on my Slackware 10.2 box running kernel 2.4.31. I'm testing

Hi all, I'm testing Linksys SPA922 phone and I have strange issue. when call is finished on the phone I see "CallEnded" and normal silence for cca. 5 seconds and then I get fast busy for cca. 20 sec. So, this isn't automatic hangup as on other phones I have tried (Cisco 7940, grandstream, XLite,... ) and I have to manually hangup handset to finish a call. Is this normal behavior

Hi ml, I have a problem related to call parking. When on my X-Lite try to parking a call dialing #700 I don't obtain anything. I can only ear dtmf tones during conversation but not other happens. I also read in some post that only pressing # should place call in hold state but this doesn't happen on my system. Can someone help me? Thanks. Marco

We have setup an asterisk box to let everybody call into the university internal network, but I get unexpected hangups when doing an outbound call from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into the call. ----------the dial and the problem----------- -- Executing Dial("SIP/57966-a19d", "Modem/g1:96121||rt|") in new stack --

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

I have a problem with one way audio on Sip and I guess it may be a NAT issue, in the example below 204 is rung by 208 (xlite external) I dial perfectly but when I get to the answering of the Asterisk, I can hear audio from the Asterisk but cannot get audio to the Asterisk, ie If I ring the voice mail , Asterisk answers and then cannot hear my password... I have put the Ports Forward

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

Hello, I've got the following configuration: 2 X101Ps Asterisk built with BUSYDETECT_MARTIN busydetect=yes busycount=10 callprogress=yes signalling = fxs_ks With this setup, the best I can do is get voicemail with 17 to 19 seconds of silence tacked on at the end. Ideally, I'd like at most 2-5 seconds. Has anyone had any success with this? It seems that hangups are indeed detected,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi!. I am having problems with getting asterisk to detect when someone hangs up. I have a TDM400P with one FXO module connected to my telco, and also a FXS-module connected to my phone. The FXS-module detects hangups just fine, but I can't get the FXO to detect them. I am pretty sure i have disconnect supervision on my phoneline since when I

My telco doesn't provide Disconnect Supervision or Polarity Change. So I figured I have to detect hangups with busydetect=yes in zapata.conf. I tested it. When the telco sends a busy tone * detects it and hangsup. So far so good. The problem is the telco doesn't always send a busy after remote hangup. Most of the time it sends a congestion tone. I am guessing these tones from what I

Hi, Although zonedata.c contains ITU E.180 recommendations for Turkey, we are still experiencing unrecognized hangups from Turk Telekom PSTN lines when callers hangup. Turk Telekom does *not* provide supervised disconnects on analog PSTN, and the tone we receive we when caller hangs up is similar to busy, with three short beeps, followed by one long beep, which keeps repeating. We've

First of all, I'd like to send a big "thank you" to all the folks who have helped me get this far. Now on to the next problem. Here's my current network setup: The Big I ---+--- FreeBSD FW --- * (10.0.0.253) ---- PC (10.0.0.1) | +--- Laptop (public IP) natd is set up with the following rules: redirect_port udp 10.0.0.253:10000-20000 10000-20000

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ============ ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here:

Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered