similar to: GUI

To All If anyone can shed any light on this it would be greatly appreciated. My phones are unable to enter pins numbers correctly when required by the party they are calling. For example I was given an outside number to attend conference bridge. After the call was connected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if

To All; My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. < Norstar1 >----PRI----< Asterisk-1 >----IP-WAN----< Asterisk-2 >---PRI---< Norstar2> The Norstars are MICS 0x32 4.1

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

To All I am using a SNOM 190 w/Asterisk server. Here is my sip.conf [7501] type=friend context=external username=7501 callerid="Telx 7501" <7501> mailbox=7501@telx.com host=dynamic dtmfmode=rfc2833 My question is this. With above settings in my sip.conf specially "dtmfmode=rfc2833" What should my "DTMF Payload Type:" be set to on my SNOM 190 phone.

Hi all, I just recently began using HTB to try and manage bandwidth for my network. This is the script I''m using: /sbin/iptables -t mangle -A FORWARD -o eth1 -s ! 192.168.244.2 -j MARK --set-mark 53 tc qdisc add dev eth1 root handle 1: htb default 20 tc class add dev eth1 parent 1: classid 1:1 htb rate 100mbit burst 131072k quantum 59000 tc class add dev eth1 parent 1:1 classid 1:10

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

Hello - sorry for my bad english. I will make a list of all sound files on asterisk and i'll record then on professional studio. the french prompts from sineapps sounds bad... sorry for her... tell me if their is many peoples want it ! thank's. en francais: dites moi si ca vaut le coup que j'investissexr dans l'enregistrement des messages en francais. La voix sera la "voix

To Members, I currently looking for someone who can configure Cisco IAD's Ie.. IAD 2431 And yes we are more than willing to pay for the service. If interested please drop me an email mdm at openaccessinc.com<mailto:mdm at openaccessinc.com> Michael DiMartino | Director of IT | Open Access, Inc. 115 Bi County Blvd | Farmingdale, NY 11735 631.227.1034| 631.694.6730 FAX |631.988.6060

First things first. Thanks to all samba developers and this great community. Some background info: I have a Samba 4.1.9 AD domain controller alone at the moment running on FreeBSD 9.2 with aprox. 50 clients, and a TS. The clients are mixed, mostly 7 and XP SP2. Due to a hard drive failure I ended up with a messed up db and a lot of problems, almost all of them including frequent panics in the

Hi folks, My stack of trusty FreeBSD servers always seems to be growing, and it's getting to the point where the daily and security output mail is too much to make good use of. I'm looking for suggestions for log monitoring and aggregation tools, especially from a monitoring-for-security perspective. If I had to imagine an ideal system, it would be a central server that securely

Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks.

Hi, Does the ''tc'' tool show dequeue statistics? If so what switches does one use to see this. If not which is the best way to see these statistics? Thanks.

Hi all, I'm new to Asterisk and I was wondering if the following setup can work. If it can how would I go about setting it up: Phone------PBX------Asterisk Server------Cisco Router | | WAN

I'm trying to record a Meetme conference to disk, but the Monitor application doesn't seem to play nicely with Meetme. In extensions.conf, I have this: exten => 1000,1,Answer exten => 1000,2,Monitor exten => 1000,3,Meetme This starts up the monitoring OK, and it records the prompts that Meetme gives, but as soon as the user enters the conference, the -out WAV file stops

Hi, Could somebody help me this SIP trasport? I'm receiving SIP "invite" with CLI of calling party from the SIP gateway, aster that my IVR has to answer the call. sip.conf: ========= [general] port = 5060 bindaddr = 0.0.0.0 context = incomingsip videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=g729

If you ring into * and leave voicemail It does not reset the line Any ideas would be appreciated Regards Mick

Is there a decent log viewer for colorized log files? less -r for me doesn''t work well, as it sometimes clears the screen. I''d like: * Either a console type less viewer which just removes the colors * Or, even better, a console one which handles the colors without clearing the screen * Or, best of all, a GUI viewer - maybe even with color I use both Windows and Linux.

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike