similar to: error cant write to function ODBC_DEVICES

Hi All, I'm having problems getting results from a PHP file. This is what the CLI is showing. -- Executing [111 at internal:1] AGI("Console/dsp", "GoTalk.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php [Dec 14 11:57:25] ERROR[20260]: utils.c:1019 ast_carefulwrite: write() returned error: Broken pipe If I run the PHP file from

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get iax to work. I've opened 4569 in the EC2 Security Group. I'm using the zoiper client. Using tcpdump I can see the zoiper packets coming in on 4569, but nothing shows on the asterisk cli. Frame 33: 79 bytes on wire (632 bits), 79 bytes captured (632 bits) on interface 0 0000 12 31 3b 12 40 84 fe ff ff ff

Hi I have configured IAX2 realtime in Asterisk 1.6.2.13. when I cannect a client to realtime extension, always the state of extension is "UNKNOW" like: * Name : marco Secret : <Set> Context : phones Parking lot : Mailbox : 2345 at default Dynamic : Yes Callnum limit: 0 Calltoken req: Auto Trunk : No Encryption : No

Hi, I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my new one with v. 16.10.0 (B). The trunk seems to be up, and the calls are initiated, eg. an extension from A can dial an extension in B which rings. However, as soon as the extension in B answers, the call is terminated. This is what I see in the console of B: -- Called PJSIP/4053 -- PJSIP/4053-00000002 is ringing

Hi! > How to do this ?? > To proceed with your answer on PICKUPMARK, where do I put this ??? Look at the example for Asterisk 1.4 on this page: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup Philipp

Hi, I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not understand database functionality on asterisk fully. The most suspected area is func_odbc. I already googled but not luck. Your guide is warmly welcomed *Error messages when I make call and leave message.* -- <SIP/1ffa9-00000007> Playing 'auth-thankyou.g722' (language 'en') [2014-05-28

Hello, we are migrating an Asterisk install from Trixbox/Asterisk 1.2 to Elastix 0.92 with asterisk-1.4.17 on CentOS 5. I need to migrate an funtion that consults a remote sybase database, using ODBC and freetds. On the new server I am able to connect to the database using isql without problems. When I try to connect from asterisk logs show: pbx.c: Function ODBC_SQL not registered Indeed I

I don't see a way to do this currently (unless I am missing something) but I would like to be able to specify, that in order for a user to login, they need to use at least 1 public key from 2 separate key sources.? Specifically this would be when using "AuthenticationMethods publickey,publickey".? Right now requiring 2 public keys for authentication will allow 2 public keys from