similar to: Cant get Asterisk server talk with IAX

Hello, I have just set up an asterisk box (Debian unstable) and I would like to test it with a H.323 application (gnomemeeting). When I call the demo voice menu, I can't hear any sound. asterisk says that the soundfile is played: -- Executing BackGround("H323/ip$212.9.189.172:30005/29597", "demo-congrats") in new stack -- Playing 'demo-congrats' (language

I'm going to try and lay out all the relevant information I have here in this one post. I can provide more info if necessary. ISSUE 1: Office A routinely looses connection to Office B. When typing IAX2 Show Peers, it will show as Unreachable. I issue IAX2 Reload and it will work again for 1-3 days (haven't narrowed the time down yet). My theory is that the DSL at Office2 is changing

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

In my IAX config file I have: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 notransfer=yes allanrobertson- 209.23.224.97 (D) 255.255.255.255

Hello Everyone, I am trying to change our /etc/sudoers (using visudo) to allow 2 commands to be run as root without a password, but it isn't working. Here is the part of the sudoers file that is in question. # User alias specification User_Alias FULLACCESS = doug, scott # members of the FULLACCESS User_Alias may run chown and chmod without a password FULLACCESS ALL = (root) NOPASSWD:

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

Hi, I've tried to upgrade my asterisk to 1.2.3 again after disastrous bug incident yesterday but when I called and the phone was picked up, there was simply busy tone... Weird, is this another bug in asterisk 1.2.3? Currently, I rollback again to asterisk 1.0.10...:( Is there any configuration change issue in 1.2.3 cause I've just used my configuration that worked in asterisk1.2.2 ?

Hi, I have two Asterisks identically installed on two computers. One of them is directly connected to the Internet, the other one through a NAT router (Netgear MR314). On the one behind the router I have an X100P card installed for PSTN connections. In the local LAN of each PBX they are several hardware IP phones (Cisco 7960 and 7940 with SIP images, firmware image P0S3-04-4-00.bin). I have

hello i finally have success with winbind!!!! :) thanks everybody so, i use this acl's in squid external_acl_typqe NT_global/-group %LOGIN /usr/local/squid/libexec/wb_group acl FullAccess external NT_global_group "domain group" httpd allow FullAccess httpd deny all with this configuration all is DENID by squid. what???????? it's possible to write acl's specifics for

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk; here below the debug output 192.168.0.75: first 192.168.0.75: 1 192.168.0.75: 10 192.168.0.75: 102

Hi, I'm running samba-3.0.14a_1,1 under FreeBSD 5.4-STABLE. I migrated all my data from samba 2 to samba 3, and I'm having a problem with some users. For example, user john can login perfectly to the network. It?s ok. But user smith not, him get the error ?The network name cannot be found?. But all configurations of john and smith are the same. I found this on Samba Official Hanbook:

Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple

I posted earlier in the day about needed help with IAX trunking. I did some more reading and made some more changes. Here is what I have thus far: Iax.conf on one server [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [vvfarm] type=friend username=colo secret=testpassword auth=plaintext host=64.194.211.170 context=iax-incoming

Hi all. Thought you would all be interested to know that i, like many others, am experiencing issues with Samba and Photoshop 7.0. The problem is specific to Photoshop 7.0 when saving files to a Samba share via a W2k/XP OS. Users experience a disk full error when saving from Photoshop but have no issues copying/moving etc files to/from the share outside of PS. Well, after some testing, the

Hello, I am experiencing weird problem in Asterisk 16.2, possibly a bug. Same thing works fine in Asterisk 11. Here is the situation: I have 2 extensions on 2 phones. 4 extensions in total. phone 1: 8882 8382 phone 2: 8884 8384 And I have 2 SIP trunks for outgoing calls. I want to call via SIP1 when called via 8882 or 8884, and SIP2 when called via 8382 or 8384. And one last detail. SIP1

Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird

Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. I do not have a good knowledge of Asterisk, I'm not very familiar with its configuration and I've a

Hi, I was just wondering if anyone could help me with this weird problem. I am trying to install an Arabic program using wine. In order to do so, I run: LANG=ar_SA.utf-8 wine install.exe The title in the title bar comes out correctly, however, whatever content the program might have comes out blank, nothing. I just could not figure out why it came out right in the title bar, but nothing comes out

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1