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<font class="title">[[VoIP_PSTN_Gateway|VoIP - PSTN gateway]]</font><br/>
...class="basetext" width="10%"> Connect the existing PBX to <br/>alternative VoIP providers.</font>
8 KB (1,086 words) - 09:08, 31 July 2018
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The '''VoIP Server''' is the piece of software that the client connects to it to make o
...erver can control all the types of communication that could take part in a VoIP environment as we will see below.
5 KB (862 words) - 13:42, 21 March 2013
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The '''VoIP Client''' can be one of the following:
Yate can be used as a VoIP Client.
3 KB (473 words) - 08:02, 3 April 2013
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...rrently focused on [http://www.fcc.gov/voip/ Voice over Internet Protocol (VoIP)] and [http://www.webopedia.com/TERM/P/PSTN.html PSTN], its power lies in i
* [[VoIP_Server|VoIP Server]]
3 KB (376 words) - 12:42, 4 November 2013
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Yate has the functionality of a VoIP - PSTN gateway. It's main task is to provide signaling interworking and to
Yate can connect an existing PBX to alternative VoIP providers providing cost savings for enterprises. It can also be used by pr
10 KB (1,440 words) - 16:11, 31 October 2013
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...ne who has a analog or digital telephone line, no matter if we are talking about a local, long distance, mobile or international call.
... disconnect it from the present network and making a new contract with the VoIP service provider or to keep them both.
9 KB (1,569 words) - 13:20, 20 March 2013
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...ng as SIP. Often H323 is associated only with the call signaling part of a VoIP transmission. Actually it’s a bit more complicated then this.
...endpoints that support H323 and the endpoints that use other protocols for VoIP communication including interfacing to PSTN. Typically a H323 gateway is fo
4 KB (562 words) - 15:04, 19 November 2012
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...direct just to mention a few. The implementation of SIP it is done using a VoIP SIP driver based on [[YASS]] library.
4 KB (564 words) - 14:00, 13 May 2013
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...r controller is used to overcome some of the problems that may appear in a VoIP communication by integrating it between the caller and the called party sig
... problem, even it doesn’t apply every time and for every firewall. It is about the probability that the dynamically opening and closing of ports on incomi
2 KB (385 words) - 07:27, 27 May 2013
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Performs routing using the ENUM protocols that maps phone numbers to VoIP (or even other PSTN) resources.
3 KB (433 words) - 09:01, 10 March 2016
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See more info about using regexroute for [[Transcoding|transcoding]].
; that for VoIP protocols is in the format: ip.ad.dr.ess:port
20 KB (3,110 words) - 10:27, 4 November 2013
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...g/wiki/Telephone_Number_Mapping ENUM] protocols that maps phone numbers to VoIP (or even other PSTN) resources.<br> This protocol uses a specific format of
3 KB (500 words) - 10:28, 4 November 2013
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Useful information from various users about routing.
==VoIP to PSTN gateway==
9 KB (1,376 words) - 10:01, 27 October 2017
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...atter of Yate configuration but also of system setup. Hardware devices and VoIP protocols also have specific demands on the system that place different lim
This is a very demanding VoIP protocol because of its complexity and also some implementation issues. You
4 KB (758 words) - 16:06, 31 October 2013
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The implementation of SIP it is done using a VoIP SIP driver based on [[YASS]] library.
More infomations about [[SIP Routing in Yate]].
8 KB (1,261 words) - 08:43, 4 November 2013
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|class="yate-header-left"|'''About'''
|class="yate-header-left"|'''About'''
8 KB (1,083 words) - 11:49, 27 October 2017
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#%# family=voip
echo "graph_category voip\n";
3 KB (332 words) - 16:02, 31 October 2013
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IAX is a VoIP open protocol that uses a single UDP port (usually 4569) for the channel si
... protocol] and in the [http://www.voip-info.org/wiki/view/IAX IAX page for VoIP info].
12 KB (1,986 words) - 15:13, 31 October 2013
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This module is a VoIP IAX2 driver based on ours [[Yiax]] library.
* http://www.voip-info.org/wiki/view/Asterisk+bandwidth+iax2
16 KB (2,483 words) - 09:12, 4 November 2013
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VoIP H.323 driver requiring the [http://yate.null.ro/tarballs/openh323/ OpenH323
VoIP SIP driver based on our own [[YASS]] (Yet Another Sip Stack) library
12 KB (1,662 words) - 13:25, 1 October 2019
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The alsachan module can be used either to test your Yate or to use Yate as a VoIP client with [http://www.alsa-project.org/ ALSA] drivers.
885 B (146 words) - 09:56, 4 November 2013
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This module can be used either to test your Yate setup or to use Yate as a VoIP client.
489 B (90 words) - 09:56, 4 November 2013
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...g any other telephony capable channels like wpchan or zapchan. Faxing over VoIP using the G.711 A-law or mu-law codecs is possible but unreliable. You can
The '''chan.hangup''' message will carry the information about the fax sent or received:
6 KB (1,040 words) - 10:09, 4 November 2013
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...ncy tone makes the signal sensitive to disruptions of the order of typical VoIP packetization (20 or 30ms) while the high frequency tone tries to catch pro
...tamps of the received data are also counted. These can be detected only on VoIP calls since in PSTN the timestamp information is not present (the network i
2 KB (376 words) - 07:16, 23 May 2014
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This module is a VoIP SIP driver based on [[YASS]] library.
25 KB (3,811 words) - 10:09, 6 July 2018
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...delivering industry-grade, robust modular platforms for IP-based clustered VoIP systems.
...egrated approach and problem-solving solution in the turmoiled and diverse VoIP environment.
2 KB (343 words) - 15:13, 30 October 2012
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Based on Voice over Internet Protocol (VoIP) and PSTN, it can easily be extended. It supports SIP, H.323, IAX, MGCP, Ji
... Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.-->
5 KB (888 words) - 13:05, 10 March 2015
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...e see [http://www.google.com/search?q=reporting+bugs what Google can find] about it.
* Details about the hardware you are running Yate on - type, speed and number of processors
5 KB (943 words) - 09:36, 9 July 2013
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* [[VoIP PSTN Gateway]]
4 KB (590 words) - 16:13, 31 October 2013
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...ection holds general information about VoIP, various protocols and typical VoIP applications and setups.
...ne who has a analog or digital telephone line, no matter if we are talking about a local, long distance, mobile or international call.
19 KB (3,169 words) - 11:20, 16 September 2013
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...ernational services. Some elements of PSTN have already been replaced with VoIP, which is cheaper and offers a lot of new services. In this chapter we will
===About telephone exchanges===
5 KB (774 words) - 15:04, 21 August 2013
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===About the E1/T1 carrier systems===
Here are some technical data about E1. The 2048Mbit/s speed is achieved using E1 carriers. This is possible by
6 KB (1,091 words) - 15:03, 21 August 2013
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* [[VoIP PSTN Gateway ]]
724 B (91 words) - 08:34, 4 November 2013
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|Nokia N80|| cvs 08.02.2007|| SIP || OK || diana@voip.null.ro
|Nokia N81|| 2.0 release|| SIP || OK<sup>2</sup> || paulc@voip.null.ro
9 KB (1,256 words) - 07:01, 25 September 2015
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Detailed documentation about SLT is available directly from Cisco:
* [http://www.cisco.com/univercd/cc/td/doc/product/access/sc/rel9/soln/voip20/impl/impdesn.htm|Voice Networks: Design Fundamentals]
10 KB (1,328 words) - 09:09, 26 October 2017
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...ines in use on source and destination both 1), but the call was routed via Voipstunt. Else you may have nasty surprises with your next bill ;-)
;Route to the German Alice hotline, using voipstunt as provider (as set by line)
6 KB (833 words) - 15:07, 31 October 2013
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[mailto:yate-subscribe@voip.null.ro Subscribe] - yate-subscribe@voip.null.ro - to the Yate mailing list.<br>
[mailto:yate-unsubscribe@voip.null.ro Unsubscribe] - yate-unsubscribe@voip.null.ro - from the Yate mailing list.
3 KB (425 words) - 08:50, 13 February 2013
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Thank you for agreeing to moderate the yate@voip.null.ro
<yate-subscribe-john=host.domain@voip.null.ro>
3 KB (469 words) - 08:59, 13 February 2013
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Go to [http://net.sapo.pt/extras/voip/ SAPO website] for more information about SAPO.
domain=voip.sapo.pt
1 KB (168 words) - 15:37, 31 October 2013
