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4

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 ...


2

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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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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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 ...


2

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. ...


2

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 ...


2

There is not an unique way to do it. This is one way: I defined my extensions starting with anything but 9, and use 9 to signal that it's a PSTN call. In extensions.conf add something like this: exten => _9X.,1,NoOp(Call to PSTN) exten => _9X.,n,Dial(DAHDI/g0/${EXTEN:1}) Put it on the same context where you put the dialplan for the extensions. ...


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You cannot find the information because you, or possibly some optical character recognition process, have mis-spelled them. They are IAX (the inter-Asterisk exchange protocol, designed for communication between Asterisk systems as the name suggests) and IAX2 (version 2).


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If both side of SIP call use SAME codec, you need format module (format_g729.so for example) but NOT need codec module(codec_g729.so) However if you playback something and recording is not in .g729 you need codec. If you use uncompressed stream(other codec or pstn/e1 card) you need codec. Also in some variants can be situation when one stream downway go ...


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With static public IP address you do need STUN. It wouldn't hurt using it, but you can configure public IP statically: [general] externip=63.182.70.1 localnet=192.168.0.0/255.255.255.0 [peer1] type=peer nat=yes qualify=yes (example from http://www.smartvox.co.uk/astfaq_configbehindnat.htm) You most likely would also need to forward RTP port range on ...


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Simpleast idea - setup openvpn for example with network 10.0.0.0/24 and pool 10.0.0.10-100 After that just reload asterisk and add to sip.conf localnet=10.0.0.0/255.255.255.0 All clients have connect to 10.0.0.1 Please note, you have start asterisk AFTER openvpn


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Use Congestion command pro-sip*CLI> core show application Congestion -= Info about application 'Congestion' =- [Synopsis] Indicate the Congestion condition. [Description] This application will indicate the congestion condition to the calling channel. [Syntax] Congestion([timeout]) [Arguments] timeout If specified, the calling channel will ...


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What you can't do You can't put pcm_mulaw audio or H.263 video in MP4 container. The error you provided by itself is generally not informative, but it is always accompanied by the actual error (at least for ffmpeg, not sure about avconv). Probably something this: Could not find tag for codec h263 (or pcm_mulaw) in stream #0, codec not currently supported ...


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The basic premise of this document is to give any person a way to tell if their ISP is blocking any incoming ports. Attach a PC directly to your service provider's modem. There should not be a router or any other device between the PC and the ISP's modem. Open a web browser (IE) and in the address bar enter www.grc.com. Once at the Gibson Research ...


1

It depends a bit on what you want to achieve how you go about doing things. You can hook up hardware (like an old analogue phone) and use an asterisk module to send commands to it which then effectively sms from a phone. Some mobile companies offer an analogue line dial in service with an anologue modem (remember those?!) hanging off the back of them ...


1

I did setup RasPBX on my Raspberry Pi and I really like it. I didn't try to setup fax on it, but I saw option in documentation. Forwarding should work fine too, I forward all phone calls from G voice to my Russian cell phone thru another (Multifon-Megafone) sip provider.


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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 ...


1

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 ...


1

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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I've just made changes in /etc/odbc.ini. For MySQL Driver Connect: Changed: UserName to UID and it works. Below is the configuration details: [MYSQL] Description = MySQL connection Driver = MySQL Database = <Database Name> Server = <Host Name> UID = <Your-DB Username> Password = <Your-password> Port = 3306


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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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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. ...


1

How the shell finds your programs When you type foo in the command-line of your shell, it looks foo up in a set of paths defined by your shell's PATH environment variable (which you can inspect by running echo $PATH). So my guess is simpe: the asterisk package you had before reinstalling had the asterisk binary located in a place listed in your user's ...


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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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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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