similar to: About minimum requirements to install PSTN GW+SIP Client

Dear all, I would appreciate it if you could teach me "Asterisk SIP NAT Config". I'm trying to capture SIP Register with externip that should set in contact header at External SIP Server as shown below, but I haven't seen it. I need your help. My experiment environment is as follows.

Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to

Hello, I'm really new on Asterisk. Is it possible to use a telephone machine connected to a modem as an asterisk voice input output device? I do not need PSTN connection. The scheme i'm thinking about is; user -> phone -> modem -> asterisk -> ip -> vice versa. If it is possible can a user dial another asterisk user via the phone? I've searched astersik lists but

---------- Forwarded message ---------- From: Christopher Aloi <chris.aloi@gmail.com> Date: Jun 18, 2006 9:52 PM Subject: Re: FW: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts? To: Alexander Lopez <Alex.Lopez@opsys.com> Alexander, Thanks for your reply, may I ask a few questions? - Does the

HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all ! Have anyone a resource / link for documentation to configure Asterisk to act as a SIP 2 PSTN gateway (ISDN PRI) ? Thx. Regads, Andreas. -- "If you want to pray. Go to the sea." ---------------------------------------------------------------- Andreas Czerniak <cognac@amcs.net> PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC

Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a "183 Session progress". Obviously, asterisk thinks that the telephone is not ringing (because it expects a "180 Ringing") and we have no ringback on the pstn side. Putting a

Hello, Ever since I updated my x86_64 system from CentOS 6.5 to 6.6, I have when I open a terminal, four lines of 'return: Command not found' diplayed. I'm at a loss as to why this is, and was wondering if there is a command I can use to find out what is causing it, and hopefully fix this issue. Brian Bernard

Having set up the Windows clients and Samba (version 2.2.5-10) with a 'tmp' share the share cannot be accessed from the Windows clients. On the Windows clients, after attempting to map a network drive to the Samba share using Windows Explorer the message 'No service is operating at the destination network endpoint on the remote system' is diplayed. I believe I have checked all

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

Hi, I'm using asterisk version 1.8.3.3. In earlier versions I used queues, but with the new version the queuing mechanism doesn't work If I look in the CLI at I see that the queue-member is invalid: Members: DADHI/g3/0655871460 (Invalid) has taken no calls yet The queues.conf looks like this: [general] persistentmembers = yes monitor-type = MixMonitor [test] musicclass

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to place calls on hold using Asterisk Manager Actions? Amaury -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I''m trying to configure QOS , but I''m don''t have success. My files: #/etc/shorewall/tcdevices #INTERFACE IN-BANDWITH OUT-BANDWIDTH eth0 256kbit 256kbit eth1 256kbit 256kbit eth2 256kbit 256kbit #/etc/shorewall/tcclasses #INTERFACE MARK RATE CEIL PRIORITY OPTIONS eth1

Hi! Is it possible to display a 3d array as an RGB image? For example, given the following array: > matriz3d , , 1 [,1] [,2] [,3] [,4] [,5] [1,] 170 174 173 172 161 [2,] 171 178 174 166 149 [3,] 168 174 173 166 156 [4,] 171 170 173 166 164 [5,] 167 170 170 171 169 , , 2 [,1] [,2] [,3] [,4] [,5] [1,] 138 131 128 128 125 [2,] 138 127 129 122 134

Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register:

Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

Hi Guys: I am very new in Asterisk Queue, so may be i'm doing wrong somewhere. I have Asterisk 1.8.3.3 and Dahdi 2.4.1.2. I defined some agent's on Asterisk Queue, and the problem is that the agent is allways on UNKNOWN status, so Asterisk can dial to the agent even if the agent is allready busy. No matter if the agent is dynamic, realtime or static. I tried with sip channels and there

Hello list i have configured astersik 1.4 with sip i have a question when i put in dial plan.conf exten => _0678922645.,1,Set(CALLERID(number)=520460587) exten => _0678922645 .,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => _0678922645 .,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)) exten => _067892264*5*,2,Hangup() i can not call my