Mainly asterisk voice quality is amazing with latest codec’s and it’s a free open source framework sponsored by Digium. Let’s see how we can install and configure Asterisk PBX 13.x on CentOS. Here we have built the Asterisk with SRTP to accept the encryption connection so the communications between the server and phones are secured and encrypted. I tried more but i am unable to install codec g729 on asterisk server. The uname -i return x8664 the model name: Intel(R) Xeon(R) CPU E3-1271 v3 @ 3.60GHz Asterisk version 13.1.
Inter-Asterisk eXchange (IAX) configuration file is divided in contexts. The context general contains general settings for the IAX protocol, like on which port Asterisk will listen, to use jitterbuffer, which audio codecs are allowed and which are disallowed, etc. Every other context is consider for user account configuration.
But what is context? Context is the group of context name and all of the following after that name lines (the end is the begining of new context): Example: start-myfirstcontext option1=something.
End- mysecondcontext option1=something. 2.1 context general. Example: codecpriority=host Codecpriority options: caller: Consider the callers preferred order ahead of the host's. Host: Consider the host's preferred order ahead of the caller's. Disabled: Disable the consideration of codec preference alltogether.
(this is the original behaviour before preferences were added) reqonly: Same as disabled, only do not consider capabilities if the requested format is not available the call will only be accepted if the requested format is available. Rtcachefriends yes no Cache realtime friends by adding them to the internal list just like friends added from the config file only on a as-needed basis.
All possible optoins that can be set for individual user: type user peer friendTo set the type of the user: friend (allow user to make calls and to be called), peer (user can be only called) or user (user can call only). Context Sets the incoming context fot his user. Auth plaintext md5 rsaIAX supports three methods for authentication. Plaintext - is the least secure. Md5 - it uses md5 algorithm to confirm the authentication.
RSA - is the most secure one. It uses public/private encryption key, that can be generated by astgenkey application (public key must be manually transferred to the server need put in /var/lib/asterisk/keys/.pub, the private keys are placed in /var/lib/asterisk/keys/.key) secret This is the authentication password for that user. Disallow all Same as the option in context general. Allow all Same as the option in context general. Setvar = We can set some variable. Dbsecret / The authentication password can be stored also in the Asterisk database ( astdb). Callerid Specify the Caller ID string that will be used for this user.
Inkeys The public keys used to decrypt authentication for the incoming client requests. Outkeys The private key used to encrypt the outgoing requests for this user.
Permit Permit IP address/network for incoming calls. Deny Deny IP address/network for incoming calls. Host dynamicYou can set static IP which will be associated with this account or to use dynamic one ( dynamic). Mask Subnet mask for the host. Defaultip IP address to be used before registration. Accountcode Billing account code.
Qualify yes no Check this user for availability. The is in milliseconds.
Mailbox @Voicemail box for this account. Trunk yes noIf set to yes,it will be used IAX2 trunking for this context. Notransfer yes noTo disable IAX native transfer, set this option to no.
Peercontext Default context to request for calls to peer. Regexten After registration what extensions to be added. Jitterbuffer yes no We can turn on/off jitterbuffer individually for every user. Sample configuration. general bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no tos=lowdelay autokill=yes guest type=user context=test callerid='Guest IAX User' anatoliy callerid=Anatoliy K.
Username=anatoliy secret=anatoliy type=friend host=dynamic context=test disallow=all allow=ulaw allow=alaw allow=gsm ivan callerid=Ivan T. Username=ivan;secret=ivan dbsecret=Password/ivaniax type=friend host=dynamic context=test iax-test type=peer username=iax-test host=dynamic trunk=yes context=test extensions.conf. User Comments Ghina (q65ux1azj9f at outlook dot com) 18 December 2015 10:03:17 Kyle Korver. He plays for the Bulls now. I may or may not have a full size card board cut out of him in my room. He might not be the best player but I sure do like to look at him.
Khanh (khanhvctl at yahoo dot com) 05 January 2008 11:39:48 I have configured following your instruction but my two asterisks haven't connected together and it released error: 'autocongestion is due to low responsibility'. Could you give me solution for this situation. Thank you so much. Nabeel (nabeel at convergence dot pk) 04 December 2007 08:39:29 what kind of this warning show explain plzzz WARNING611: chaniax2.c:1760 attempttransmit: Max retries exceeded to host 192.168.200.123 on IAX2/104-30 (type = 6, subclass = 11, ts=3138185, seqno=76) Shiju Joseph (letterstack at gmail dot com) 18 September 2007 16:42:18 Hi, How we can add jitterbuffer to individual iax trunks, can we add it in iaxadditional.conf? Do we need to give command 'iax2 set jitterbuffer' from asterisk cli? Regards Shiju V.Joseph dhiraj (2dhiraj at gmail dot com) 26 March 2007 06:37:20 I have to public ip and asterisk is running in nat mode on both the ip.these two asterisk is not able to communicate with each other.
I have tried with sip.conf and not used iax.conf. If iax.conf works then please send me configuration example. Amjad ali amjad (amjadse at yahoo dot com) 26 January 2007 00:26:45 asterisk is no doubt a nice pbx and the asteriskguru is really a guru fro nice learners. Amjad ali amjad (amjadse at yahoo dot com) 26 January 2007 00:21:39 asterisk is no doubt a nice pbx and the asteriskguru is really a guru fro nice learners. Arben Myrtaj (info at dimal dot com dot al) 30 December 2006 14:38:06 I am trying to use G723 on asterisk and it says no compatible codecs. Should I buy a license for that? If yes where can I buy it.
Ezio Vernacotola (ezio2 at uptime dot it) 12 November 2006 22:05:11 There is a typo on notransfer option comment. Notransfer yes no As the name suggests to disable IAX native transfer you must set this option to yes ZABIII (zabiiiiii at hotmail dot com) 12 October 2006 01:00:46 plz help me with this. I always have error with IAX2/vitel-outbound-1 is ciruuit-busy. Always when i reboot. Rudra narayan acharya (rudraach at yahoo dot com) 26 September 2006 16:31:32 Hi, I'm experimenting asterisk in a dialup environment.But I get lot of voice breaks while there is a conversation between two remote extentions.Is there any way to reduce that.As jitter buffer is one of the solution but it only works with iax trunk. Thanks rudra zhiyong (zhiyong dot m at gmail dot com) 13 September 2006 14:59:01 I use IAX2 trunk between two asterisk server.
At a few calls (less than 30) enviornment, both caller and callee can hear each other clearly. But when calls reach 45 or above, the quality of sounds is bad. I wonder if a IAX2 Trunk should limit concurrent calls? I use ILBC codec in the trunk. Btw: I use asterisk-1.2.12, zaptel-1.2.8 Elmer (elmer at diavox dot net) 06 September 2006 14:52:40 Hi, I am new to asterisk and linux. I installed asterisk and already configured SIP extensions and IAX.
I made some calls already from a hard phone to E-lite and vice versa. But all of a sudden, I am receiving the error below. I could not call any more from the hard phone. Below is the error message. I really need to resolve it urgently.
NOTICE3537: chaniax2.c:6522 socketread: Rejected connect attempt from 192.3.33.74, who was trying to reach 'TBD@' effendi (effhal at freebsd dot or dot id) 25 July 2006 08:44:37 is that any possibility for iax use as trunk protocol between asterisk server? But from asterisk server to the client still use sip as a signalling protocol. Thanks for the answer Wolfgang Alper (wolfgang dot alper at altrust dot de) 29 April 2006 10:06:27 Mike, IAX2 is perfect to connect several asterisk servers. The configuration is quite easy. All it needs is a peer and user account on each system. The advantage i see in comparison to SIP is, that IAX2 is really firewall friendly.
Just open one UDP port and you are set. Mike Keizer (mickelkeizer at bluewin dot ch) 04 April 2006 12:59:40 I'm wondering why I don't seem to find a good example where IAX connects several asterisk servers including a dial plan. Isn't IAX supposed for that? It's called Inter Asterisk Exchange isn't it?
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Besides that why would I want to use IAX for a telephone? There is SIP, MGCP, Skinny you name it. Do we realy need another (none standard) protocol for telephones? Eric De Ron (eric dot deron at skynet dot be) 18 August 2005 09:58:41 Type=user/peer/friend It seems that a channel od type 'user' cannot be a host='dynamic'. The registration fails. If the definition of a user is 'may only call', I wonder why a user cannot register Thanks for an answer.