similar to: Broken Pipe error while using UpdateConfig command

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID:

Hi people, I need to update the voicemail.conf from the UpdateConfig Action (AMI). The problem is that I executed: Action: UpdateConfig srcFileName: voicemail.conf dstFileName: voicemail.conf Action-000000:append Cat-000000:test Var-000000:exten Value-000000:>999,test But I don't see the changes in the file. Can anybody tell me if there is something wrong in that code? Thanks,

hello all, I have installed centos 5.5 ( linux text) and I have updated it with # yum install bison bison-devel================?ok # yum install ncurses ncurses-devel==========?ok # yum install zlib zlib-devel===============?ok # yum install openssl openssl-deve=======?ok # yum install gnutls-devel============ ==?ok # yum install gcc gcc-c++============?ok # yum install newt

"In computer software standards and documentation, the term deprecation = is=20 applied to software features that are superseded and should be avoided.=20 Although deprecated features remain in the current version, their use = may=20 raise warning messages recommending alternate practices, and deprecation = may indicate that the feature will be removed in the future. Features = are=20

Hi All; To enable the compilation for the addon that is coming with Asterisk 1.8 when doing compilation for the Asterisk, what should I do? Regards Bilal

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

Hi gang, Since I'm getting no joy from device_Status or SIPPEER in 1.4.26-rc1, I thought I would do an AGI to read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet;

Hello List, A few days ago I installed ViciDial on a server, and while looking to the default 'extensions.conf' file, I saw this line: exten => _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT) Can someone point me out to the Asterisk documentation part where explains how to use server IP's as extension number? I could not see it in the ATFOT2 book, and I would

Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hi everyone, Installed asterisk from yum repository but I think H.323 is not supported as I tried commands like this and they don't work: - *h.323 debug*: Enable chan_h323 debug - *h.323 gk cycle*: Manually re-register with the Gatekeper - *h.323 hangup*: Manually try to hang up a call - *h.323 no debug*: Disable chan_h323 debug - *h.323 no trace*: Disable H.323 Stack Tracing

Hello Incoming calls from the FXO trigger an AGI script which simply NOOP data sent by Asterisk through stdin. The first two NOOP work fine, but after this, Asterisk isn't happy: ============= extensions.conf ... [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/test.lua) exten => s,n,Wait(5) exten => s,n,Hangup =============

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",

Dear, I am getting this in CLI on release candidate version of Asterisk. Any ideas, or points where to look? -- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi [Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe -- <SIP/916-fc001968>AGI Script rad-auth.agi completed, returning 0 Best regards, Josip

Hello, I am looking for a dialer program, free or not, that allows me to perform scheduled calls, generate reports and let me upload sound files. Is there something that fits these features?. If there is not any product like I mentioned before I am interested to build this kind of software but I need ideas to make it useful for technical and non-technical people. I don't want to spend my

Hi all, I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI, with command 'sip show channels' , I see two channels in use with callID and other infos detailed; also 'sip show inuse' give me same result (in terms of channels usage): Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message xx.xx.xx.79 209

Dear All, First, I am not so much experienced in Asterisk. I need asterisk to route the call to soft switch when the caller is not in its extensions list. And also when routing to soft switch, a number 4327 has to be added in the caller's number and then routed. I think its not so hard in asterisk. Please help me. Regards, Pratik -------------- next part -------------- An HTML attachment

We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 in 3 times the call at the final landline is silent and we see "RTP Read too short" scrolling on the console log. Where do we

Hi list, I've been beating my head for about 3 days on this one. I have Asterisk 1.4.41 installed using openh323. As long as I'm inside my firewall, everything is hunky-dory. When I move to server on another subnet, I'm still able to connect, but no longer have sound. Any good pointers or suggestions? Thanks Danny Nicholas

Hi I have followed https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29, to my surprise there is only one config file by the name zapata.conf under /etc/asterisk/ There are no other config files. Any thing i am missing ? Please suggest/guide. Regards, Kaushal