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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.

This module can be used either to test your Yate setup or to use Yate as a VoIP client.

...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:

...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

This module is a VoIP SIP driver based on [[YASS]] library.

...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.

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.-->

...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

* [[VoIP PSTN Gateway]]

...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.

...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===

===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

* [[VoIP PSTN Gateway ]]

|Nokia N80|| cvs 08.02.2007|| SIP || OK || diana@voip.null.ro |Nokia N81|| 2.0 release|| SIP || OK² || paulc@voip.null.ro

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]

...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)

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Go to [http://net.sapo.pt/extras/voip/ SAPO website] for more information about SAPO. domain=voip.sapo.pt