asterisk

问题 Does asterisk support instant messages? I have tried to configure asterisk for IM (from this example), but when I'm trying to send IM to another sip account asterisk returns warning: WARNING[20128]: chan_sip.c:16379 receive_message: Received message to sip:test3@192.168.21.153 from sip:test1@192.168.21.153; tag=d9fdcc28313946c5a2a6ae1eae997c62, dropped it... Content-Type:text/plain Message: hahaha Additionally the SIP client returns message "method not allowed". This is my sip.conf file:

问题 Im trying to setup voip exchange using asterisk ans CSipSimple as client, fol are the detials Server Side: Generate certificates for server and two clients Place the server cert in /etc/asterisk/keys/ sip.conf: [general] context=local allowguest=no alwaysauthreject=yes allow=gsm allow=ulaw allow=alaw directmedia=yes allowoverlap=no bindport=5061 tlsdontverifyserver=yes tlsenable=yes tlsbindaddr=192.168.0.119 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt

问题 Is it possible to restrict number of extensions created in asterisk using sip.conf when asterisk is using realtime tables. 回答1: Asterisk not create extensions. It read extension from sip.conf, iax2.conf, realtime database etc That files usually created by some web or control system(for example freepbx). You should add restriction in your app/system which do add of extension. There are no special limit on extensions number. Only limit like that is limit of concurrent channel count(controlled

问题 I am setting up a new Asterisk system at a hosted Asterisk provider using 1.4.38, moving from a self-hosted version 1.2. Whereas before I could do: [incoming] exten => _1NXXNXXXXXX,1,DoSomething Now, it appears the incoming context will only get properly called if it is: [incoming] exten => s,1,DoSomething How do I determine what number was dialed in this scenario? We have dozens of numbers and I need to be able to know which was called to route correctly... Thanks, Ben 回答1: You can use a

问题 I m using Asterisk and I want to know if the User-Agents could register with SIP using TLS authentciation algorithm SHA-1. Does the Asterisk support SIP authentication using TLS authentciation algorithm SHA-1? 回答1: Asterisk 1.8 support only md5-based auth sha1 supported by FreeSwitch. Asterisk 11 seams support sha-1 if (!strcasecmp(hash, "sha-1")) { dtls->set_fingerprint(instance, AST_RTP_DTLS_HASH_SHA1, value); } else { 来源： https://stackoverflow.com/questions/23157257/does-the-asterisk

问题 I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is jitsi-----> asterisk server-----> analog PBX ----> landline phone I configured this scenario as follow in chan_dahdi.conf file ; General options [channels] usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;FXO Modules group=2 echocancel=yes signalling=fxs_ks context=Incoming channel=1-20 After loading

问题 I'm trying to get the status of an Asterisk Server using a python socket but nothing happens. Here is my code: import socket s = socket.socket(socket.AF_INET, socket.SOCK_STREAM) HOST = '192.168.1.105' PORT = 5038 s.connect((HOST, PORT)) params = """Action: login Events: off Username: admin Secret: mypass Action: status Action: Logoff """ s.send(params) data = s.recv(1024) print data + '\n' s.close() I just get a message saying Asterisk Version and nothing more. I hope somebody could help me

问题 I am trying to run a shell script using asterisk AGI. I have used the tutorial mentioned here http://www.shiffman.net/p5/asterisk/ My extensions.conf is as follows [default] include => clicall [clicall] exten => _X,1,Goto(s,1); exten => _X.,1,Goto(s,1); exten => s,1,Answer(); exten => s,n,EAGI(runEAGI.sh); The script I am trying to run (runEAGI.sh) is as follows #!/bin/bash java /home/sphata001/Downloads/EAGI/JEAGIClient $$ The permissions have been set as 755 and the script is placed in /var

问题 I want to set direct peer to peer media setup in asterisk I used directrtpsetup=yes Also I want to achieve it without re-Invite.So I use this parameter.In sip.conf its written that it works without re-Invite,But its not working for me.Any one please help me how to solve it. I want to bypass asterisk for media. 回答1: Please note, directrtpsetup not work with nat nor if sdp offer DIFFERENT for both peers. Also require both directmedia and directrtpsetup directives and asterisk version above 11.*

问题 does anyone know the steps required to debug the agi secript (a2billing.php) of asterisk . I am able to remote debug php cli if i call it form the linux console. but if it runs from within asterisk , it does not work. any idea how to debug it ? 回答1: To turn debug mode of a2billing you have to go config, and in agi-confX put verbose ->DEBUG(see comments) After that you can check a2billign debug file(specified in /etc/a2billng.conf) or do from console asterisk -r agi set debug on core set