similar to: ser+asterisk problem

hello like if 6000 is the main exchange number. any one dial to 6000 will be asked for pressing his desired extension then he can press his desired extension then his number is diled exten=>6000,1,Background(enterdesiredexten) exten=>6000,2,Wait(2) exten=>2000,1,Dial(SIP/${EXTEN})

hi any one tell me how to make a dialplan my extensions.conf exten => _40XXXXXXXXXXXX,1,Dial(OH323/${EXTEN}) i want to dial to 40XXXXXXXXXXXX number. XXXXXXXXXXXX could be any number like 923335224005 or 92512213248 at the moment when i am trying to dial 40923335224005 asterisk is dialing Executing Dial("OH323/R11429", "OH323/40923335224005") but i want him to dial

i am getting 403 Forbidden message from asterisk when it try to register my user agent. i am basically useing mysql through ODBC. i hvae checked ODBC connecteion with 'ODBC Show' command. ------------------------------------------------------ *CLI> odbc show Name: mysql1 DSN: asteriskdsn Connected: yes *CLI> ------------------------------------------------------ and user is added to

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

i am not able to communicate with ip scheme 172.16.x.x but when i change it to 192.168.x.x it works properly any one help me __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

Hi Charles, I made the change, and it still won't incite a poweroff: ========================================================================== [root at localhost ~]# upsc myups at localhost battery.charge: 76 battery.charge.low: 90 battery.charge.warning: 30 battery.runtime: 1811 battery.temperature: 31.9 battery.type: PbAC battery.voltage: 49.2 battery.voltage.nominal: 48.0 device.mfr:

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

Hi, Microsoft Dynamics CRM 3.0 integration with Asterisk/Trixbox has been included in StarJunction and Star Outlook Dialer. This is in addition to existing support for SugarCRM and Salesforce CRM. It is available at http://www.starutilities.com/staroldialer.htm Thank you for your valuable comments and suggestions. Kamran

i have problem in compiling asterisk-addons 1.0.1 --------------------------------------------------------- [root@kamran asterisk-addons-1.0.1]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -c -o cdr_addon_mysql.o cdr_addon_mysql.c ../asterisk: Not a directory make: *** [cdr_addon_mysql.o] Error 1 --------------------------------------------------------- i want to

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

Hi Everyone, New to the list. Thanks in advance for any assistance you are able to provide. I have a TrippLite SMART2200RM2UN UPS. I have installed and configured NUT as instructed on the website, and am able to monitor the status of the UPS without much problem. The only problem I am seeing is that I cannot get the machine to actually send a Low Battery ( LB ) signal. When I run

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

hello i was searching for solution to problem (sip->h.323). any one from this list asterisk mailing have any idea how to fix it. i am getting error when i try to call from sip to h.323 user i am successfully registering my asterisk box with gnugk. but when i try to call to h.323 openphone on working on GnuGatekeeper, asterisk is not routing it to GnuGk. i am getting the following error. do

hello i am trying to make a callback solution. client will call callback number and call is terminated. now callback server will create a call for that client. actually i have a problem in this process. that server is creating call to client (UA) when previous call is not disconnected yet. UA---------->Asterisk(callbacknumber) callis answered UA<----------Asterisk(callbackserver) call is

HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 62222 asterisk passes this is ser and then again ser passes this no 2222 (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello all, i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to 1.23 requests from SER to asterisk die quietly, no

Hi all I am using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf

I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in