similar to: Question about context from-internal

Hi List: I can not find out an example how to store "include => context name" statement into Realtime static. Please help me on this one. Thanks, Tielin

Hi List: The main concern I have is that multiple accounts of the Asterisk servers could be configured in the phone, any one server of group of registered servers (2-3) down, the phone still works with existing registered servers, or the sip phones automatically register to next server referencing the account settings when current registered Asterisk server is down. Does anybody know some

Hi List: Can someone hlep to provide one or two examples to data entry for sip.conf under the table structure? CREATE TABLE `sip_conf` ( `id` int(11) NOT NULL auto_increment, `cat_metric` int(11) NOT NULL default '0', `var_metric` int(11) NOT NULL default '0', `commented` int(11) NOT NULL default '0', `filename` varchar(128) NOT NULL default '',

Hi All: I'd like to test a pure VoIP call center set up under Asterisk, Can I use existing IP routers to get VoIP traffic from service provider to Asterisk with good quality of voice? In other words, do I have to do any hardware upgrade to make VoIP work in existing enterprise environement, we have 10g Ethernet LAN? Many thanks, Tielin

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

I've installed Asterisk from the Asterisk @home distribution. Ultimately I will be using Asterisk for voicemail for about 150 users. Calls are (mostly) handled by a legacy PBX although we do have a couple of Cisco 1760 routers that connect a remote office. I've setup a SIP trunk that routes calls from Asterisk to the 1760, and that works fine. I've also configured one of the 1760s to

Hi list: Has anyone done to set up two servers in different remote offices through VPN in order to get the VoIP communication? Thanks for your information. Tielin Xu

Hi List: Please someone help me to point out where I can get the idea to configure Asterisk for mysql server running on different Linux. Many thanks, Tielin

Hi All: I'd like to use Realtime Static in terms of the performance concern about dynamic realtime. Assume that I create a table: as following: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20)

Hello, I have problem with interpretation of CDR entries. What happened? ------------- There was: 1. at 09:00:26 we received call from unknown caller (no callerid) to secretary with extension 17 2. secretary answered and after some conversation called to extension 18 to check if she could transfer customer call 3. at 09:02:55 extension 18 answered and call was forwarded 4. call was very long

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

Hi We're using asterisk 1.8.23.1. Our inbound calls are routed into the default context with explicit number matching. If found they are passed on to a distinct context for the number being called using the Goto application. If the call is successful or even if it has no answer, the cdr dcontext field has the correct second context. However, if the call fails or is busy, and even though we

Hi, I'm running *@home 0.5 exactly as it comes off the .iso image and have configured an extension (206) using AMP for a grandstream 102. Have checked sip_additional.conf looks ok, but I can't get the incoming cli on the 102 to read anything other than 't ri', apparently from other posts the phone is trying to display 'asterisk'! I've got a couple of pc's

Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

> Hi list! > > I'd like to save all information about calls (CDR) in a MySQL-Database. > I created the DB and a user for Asterisk on a separate server, then I > configured my cdr_mysql.conf so: > > [global] > hostname=192.168.10.3 > dbname=asterisk > table=cdr > password=MYSECRET > user=asterisk > port=3306 > > and my cdr.conf so: > >

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

Hi a new small question ;=) We have two Asterisk, connected in IAX2. On the first, in dialplan, we have: exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)}) we sent into the IAXVAR "ACCOUNTID" the accountcode. On the second, in dialplan, we have: exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) That's work, the second server get the variable. I

Luca Bertoncello wrote: > Hi list! > > I'd like to save all information about calls (CDR) in a MySQL-Database. > I created the DB and a user for Asterisk on a separate server, then I > configured my cdr_mysql.conf so: > > [global] > hostname=192.168.10.3 > dbname=asterisk > table=cdr > password=MYSECRET > user=asterisk > port=3306 > > and my

Here is the thing. All workstations in the network have the /etc/profile modified and set up to call a profile script in the network that sets up variables that we need here. Here $LD_LIBRARY_PATH and other variables get set, fixed, or modified to support our development process. The weird thing is that while in a TTY the $LD_LIBRARY_PATH is fine and retains its value. But if I log in to the gui