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3

Executing a shell of asterisk on an incoming call just doesn't feel right to me. It probably works ok but shouldn't the status of a peer be already known to asterisk? I use the function SIPPEER() with which you can request the status of a peer. If the first 3 characters (of OK (44 ms)) is OK then you can call the peer. All other situations you can forward ...


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At this point in asterisk, you're trying to build a dynamic lib. Since this dynamic lib can be loaded at any memory location, everything needs to be relocatable. The -fPIC flag means Position Independent Code, the code is made to be independent of load location - loaded anywhere. As part of this asterisk dynamic lib build, it's trying to bring in the Lua ...


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Here is one guide : Configuring and Using SIP Domains in Asterisk. Please try everything that looks pertinent and comment on it in your post, so we can know what you did or did not


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Modern asterisk works same with pci card or without. Thoose recomendation valid for linux kernel <2.6.18, that is not actual in modern distro.


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There is full documentation about ports availible. It is not clear why you think it not availble. Asterisk use port depend of technology/channel type used For sip protocol asterisk use port described in sip.conf(default 5060) and for rtp data use ports described in rtp.conf(default 10000-20000). Usually sip use only udp port. TCP-sip can also use tcp port. ...


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The configuration of the spans on your card depends primarily on what sort of lines your provider has given you so it is very hard for anyone to look at this configuration and know whether or not it is correct in your situation. However, generally, the alarm state of the span (GREEN, RED, UNCONFIGURED, YELLOW) depends solely on the configuration in ...


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Sure this described everywhere. If you not want read, try this: yum update -y yum install kernel-devel -y reboot


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You can buy an analog card like this one from Digium: http://www.digium.com/en/products/telephony-cards/analog/4-port (I assume you have a standard analog line. You can also buy ISDN cards). Install the card on your PC then you can use the PSTN AND your SIP provider as required. you can setup a dialling plan to cater for both. eg dial 9 prefix for ...


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Set it up like: ... domain=test.com domain=anothertest.com domainsasrealm=yes ; Use domains list as realms ; You can serve multiple Realms specifying several ; 'domain=...' directives (see below). ; In this case Realm will be based on request 'From'/'To' header ; and should match one of domain names. ; Otherwise default 'realm=...' will be ...


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Ideally you want to know the scope of the phone network but as your network admin is on leave it's time for some trial and error 1st IP 10.22.0.253 subnetmask 255.255.255.0 This will allow you to access the 10.22.0.* range 2nd IP 10.22.0.253 subnetmask 255.255.0.0 This will allow you to access the 10.22.*.* range Test on both to see if ...


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Yes, that is posible. Require above average level in asterisk internal work understanding. Plan is following: Do dialplan for intercom your phones(need support from phones/some headers) Test it from your extension, ensure intercome is ok. Create script to do new call for asterisk with one side Local/intercom_extension@intercom_context, other side ...


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No, the CPU and memory usage aren't limiting you; the permissible number of open files for the asterisk user is. This can be adjusted by setting ulimit/sysctl settings. The HOW will depend on your Linux distribution.


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There are quite a number of different options. Here are a few examples: Flash Operator Panel 2 Noojee Receptionist iSymphony


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I personally use Elastix. It has a widely support forum with a great newbie corner as they call it found here We personally also run it on a VM and found that 2 Gigs of ram with 1vcpu was sufficient to get it going. I would advise downloading and installing the 2.3.0 stable version as this has been tested for a few months now with no faults. The graphical ...


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As @arheops mentioned, you can use DEVICE_STATE() to get the status of a specific device, if it's 1:1 mapping (1 account on 1 device). You can use the following: ${DEVICE_STATE(${CHANNEL(channeltype)}/${CHANNEL(peername)})} Another possibility is EXTENSION_STATE() function in case you have 1:many mapping (one extension with more than one device). Both ...


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When you run asterisk on ec2 you need ensure you use realtime kernel with 1000hz timer. http://voxilla.com/2009/02/13/asterisk-amazon-ec2/ About density: For m1.small max lines usualy 20-30 For c1.medium it is near 80-100(safe value 80) For other - to be calculated. NOTE: number of lines depend of every single instance( general load on physical ...


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You can setup your mailer(postfix or sendmail) to send using different port and setup mailer on other (external) system to listen on other port. Asterisk email can use external script for email delivery, script can do whatever you want. /etc/asterisk/voicemail.conf externnotify=/usr/bin/myapp OR mailcmd=/usr/bin/myscript


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You need configure nat and firewall http://www.voip-info.org/wiki/view/Asterisk+sip+nat


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The problem is that the firewall is blocked the RTP (and RTCP) traffic. The session is established with SIP, hence you can do calls and receive calls but You dont listen, because the media traffic is blocked. Then, You should open the /etc/asterisk/rtp.conf file and edit the follow lines with the ports that You want use to RTP (and RTCP) protocol. ...


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That depends how you're going to provide connectivity to your Asterisk box. If you plan to use your single analog phone line then you will be limited to its capabilities. For example, you can't do caller ID on the Asterisk box if your telco doesn't provide it. If you're willing to get a SIP trunk (from a provider such as voip.ms or sipstation.com) you ...


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You need find voip phone with changable ringtone. Simplest variant is cisco 7940/7960 used phone(30-50usd each) After that you need have in all places where you put phone ethernet (local network) Need have router, server for asterisk For connect landline you need have fxo sip addapter, for gsm you need gsm sip gate or usb dongle supported by chan_dongle. ...


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It turns out that Database named as asterisk was needed along with asteriskcdrdb but wasn't mentioned in the tutorial. Creating the database and all tables in it fixed the issue. Cause: Some SQL scripts in the updates enforce usage of asterisk database, even if you override the database in initial settings. To resolve previous errors (which, BTW, also ...



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