similar to: Polycom behind firewall issue

I have a remote installation that connects via IAX from my office pbx. When I call an extension on the remote pbx, after the dial period, the call is terminated. Nothing I do in configuration of that extension seems to matter: -- Executing NoOp("IAX2/netconcepts@nnn.nnn.nnn.nnn:4569-5", ""Dial 710"") in new stack -- Executing

I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?

Hi all, i'm trouble with codec setup on an asterisk machine 1.4.18 and some Thomson ST2030 as extensions. In the users.conf file for internal extension i have: disallow=all allow=g729 allow=alaw allow=ulaw Without any codec installed (i mean with original g729 of asterisk) all go fine, calling from an extension to one other: Peer User/ANR Call ID Seq (Tx/Rx) Format

Please point me in the right direction.... My ISP is giving me an IPv6 prefix, but to get that I have to: The current Speedstream ADSL router will be configured as a bridge. I will have to set up a Linux (read Centos, I hope) router that will connect ethernet to the Speedstream but run PPPoE to his network and get both the IPv4 and IPv6 route delegations. There is no easy way that I know of

Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

Hi, I am using Trxibox 2.6 latest ISO install. Following is the output of : "sip show channels" [trixbox ~]# /usr/sbin/asterisk -rx "sip show channels" Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 212.53.40.40 0218245 6cfb845d050 09011/00000 0x0 (nothing) No 192.168.1.116 (None) YTc4ZmM3NjV 00101/00006 0x0

Dear asterisk users, A few weeks ago I've been attacked by a DOS on REGISTER that I've solved with a fail2ban script. Now, since a few hours, I have my asterisk 1.4.21.2 running at 100% CPU again. I've checked the log and it shows nothing related to failed register or whatever. It just tells me that some of my peers are lagged, even with a verbosity of 10000 I've made a

'morning everybody, Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e

I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

I'm trying to reconfigure my IP300 softkeys.. Currently when on a call, I have to hit more and then transfer.. I'd like make transfer appear on the first screen. Right now there's hold on there.. and hold is kind of redundant, since the IP300 has a "hard" hold button. I tried doing it in the <keys/> section of ipmid.cfg, but it doesn't seem to work.. anyone done

I'm running tinc 1.1pre11 with AutoConnect set to 'yes' and I recently started seeing lots of these messages on my VPN and cannot connect to various hosts from other hosts: (I have obscured the hostnames and vpn name, but otherwise this is a direct paste from syslog) Aug 19 14:51:51 AAA tinc.nnn[2217]: Got REQ_KEY from XXX while we already started a SPTPS session! Aug 19 14:51:54 AAA

Hi Jayson, I forward your request to the right place. Carlos (or some other) will have an answer, or at least some questions for you. -- Arnaud 2008/6/27 Jayson Anderson <jayson.anderson at gmail.com>: > Greetings Arnaud, > > I wrote you some time ago about whether NUT would support the Belkin > F6H550-UPS device in the future. > > You had replied and mentioned to send

Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. In iax.conf, I have something like the following: [General]

Hi, one of the sip-extensions we created always returns busy when someone tries to call the phone. The extension itself can place calls. We're using snom360 phones with the latest firmware. On every one of those phones when we register with the sip-extension, we've experienced the same problem. This is the output from sip show channels: Peer User/ANR Call ID Seq

Hello, I've been trying to get samba set up to authenticate users off a W2003/ADS system and it appears to be working for the most part. However, there is one issue plaguing me and I'm not sure how serious it is. In brief, the Windows SID => Unix GID mapping is failing in odd ways. After getting things set up, the following work: * wbinfo -g (lists all domain groups, ie DOMAIN+user)

Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register => username:password@speex.dyndns.org and I cant get it to work. Maybe someone who already got this to work will help... When dialing my speex extension I see the next output from consol: IAX2 Debugging Enabled *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno:

I've attempted to change my smb password on a remote NT PDS, but it always fails with resolve_name: Attempting lmhosts lookup for name SERVER<0x20> getlmhostsent: lmhost entry: 127.0.0.1 localhost resolve_name: Attempting host lookup for name SERVER<0x20> Connecting to nnn.nnn.nnn.nnn at port 139 error connecting to nnn.nnn.nnn.nnn:139 (Connection refused) unable to connect to

I have a question about software bridge setup (initscripts). If one sets up a bridge network: ifcfg-eth0: DEVICE=eth0 ONBOOT=yes BRIDGE=br0 HWADDR=xx:xx:xx:xx:xx:xx ifcfg-eth1 DEVICE=eth1 ONBOOT=yes BRIDGE=br0 HWADDR=xx:xx:xx:xx:xx:xx ifcfg-br0 DEVICE=br0 TYPE=Bridge BOOTPROTO=static BROADCAST=nnn.nnn.nnn.255 IPADDR=nnn.nnn.nnn.nnn NETMASK=255.255.255.0 NETWORK=nnn.nnn.nnn.0 ONBOOT=yes Deep

Hi, I just upgraded my entire environment from Tinc 1.0.24 (or prior) to the most recent version from git (1.1pre10+), using ED25519. Sometimes, especially after some failed authentication attempts or timed out authentications, I get the following log messages: 2014-08-11 21:51:10 tinc.NNN[3376]: Connection with XXX (1.2.3.4 port 1) activated 2014-08-11 21:51:10 tinc.NNN[3376]: