How To's
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* [[Configure Mediant M2UA]] | * [[Configure Mediant M2UA]] | ||
The [http://www.audiocodes.com/products/mediant-2000 Mediant 2000] and [http://www.audiocodes.com/products/mediant-3000 Mediant 3000] series from [http://www.audiocodes.com/ Audiocodes] support standard [http://en.wikipedia.org/wiki/SIGTRAN SIGTRAN] M2UA backhaul for SS7 links. | The [http://www.audiocodes.com/products/mediant-2000 Mediant 2000] and [http://www.audiocodes.com/products/mediant-3000 Mediant 3000] series from [http://www.audiocodes.com/ Audiocodes] support standard [http://en.wikipedia.org/wiki/SIGTRAN SIGTRAN] M2UA backhaul for SS7 links. | ||
| + | * [[Configure SCCP and GTT]] | ||
| + | The [http://en.wikipedia.org/wiki/Signalling_Connection_Control_Part SCCP] protocol provides additional signaling commonly used by mobile networks. In most cases the [http://en.wikipedia.org/wiki/Global_Title_Translation Global Title Translation] needs also to be configured. | ||
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Revision as of 10:23, 26 October 2017
Yate has multiple routing modules and signalling modules. Here you can find some how to's for configuring Yate using different modules.
Contents |
Routing
The most popular scenarios for routing:
| How To's | |
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Example of register users in Regfile module. Example of redirecting calls to another user. Examples of how to route calls from javascript module. Example of how to register users from a database. Round robin routing examples. |
Examples on how to convert SIP headers into SIP parameters. Example on how to use copyparams from routing in CDR. Setting up an IVR. Useful information from various users about routing. |
Yate configuration as Server and / or Client
Various configuration for Yate to act as a Server and as a Client using different protocols.
| How To's | |
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Yate can be used as IAX Server and YateClient as IAX Client. Yate used as a H323 Gatekeeper and YateClient as a H323 Client. Configuring Yate as a SIP server. Configuring Yate Server to use Jingle and YateClient to use a Jabber client. How to set up Yate as a PBX depending on the features you need. Configuring Yate Server to be a H323 to SIP Signalling Proxy. Use Yate server to handle all your Google Voice calls. |
How to use YateClient with SIPsocial How to connect to FreeWorldDialUp How to connect to InPhoneX How to connect to SAPO How to connect to IAXtel How to enable SIP MESSAGE request in yate5 to allow chat messages between registered users. Yate provides a protection mechanism against several types of SIP flood attacks. |
Call detail records
Below are the modules and some tips you can use when writing call logs. You use this modules to obtain billing information.
| How To's | |
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How to write call logs to a database Some things to take into account when wanting to bill from a database How to write call logs to a file |
Example of how to add custom parameters in CDR For basic setups configure yate to write single cdr entry per call, instead of entry for each call leg Example on how to start CDR Ring Timer on call.progress message. |
Monitoring and debugging Yate
Some examples on how to monitor and enable debugging in Yate and the modules involved in this actions.
| How To's | |
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Overview on how to monitor yate using SNMP and rmanager. Yate provides Debugging info on console. Debug an external module in Telnet |
Yate offers the msgsniff module to allow the investigation of messages at runtime. Yate can be monitored using Munin. |
Miscellaneous
| How To's | |
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Tips on what you should pay attention to when running Yate in VMWare. This example allows chat and short files transfer between Twinkle clients using SIP MESSAGE Request Method. |
Notes on various protocols when you wish to handle many simultaneous calls. How to modify call release cause codes. |
VoIP to PSTN gateway
| How To's | |
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Example on how to use Yate as VoIP-PSTN gateway Guide to install Sangoma cards under Linux. |
Guide to install Sangoma cards under Windows. Guide to use Yate with ISDN BRI. |
SS7 Setups
| How To's | |
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The MTP2 links are the classic way of interconnecting SS7 over TDM. This SCTP/IP protocol is used as network transport by all forms of SIGTRAN to replace classic TDM links. This is a SIGTRAN protocol and it was designed for remote processing of SS7 messages above MTP3 protocol, mostly ISUP and SCCP. The voice channels on Cisco Universal Gateways (AS 53xx and 54xx series) can be controlled using the Media Gateway Control Protocol. This usually complements the Signalling Link Transport so both signaling and voice can be provided on the same set of trunks. ISUP is normally configured in telephony exchanges to allow them to use SS7 to control calls over fixed TDM circuits. Voice circuits on Mediant gateways can be controlled on MGCP. Specific instructions for configuring Yate and Mediant are provided. |
This SIGTRAN protocol is used to replace classic TDM links in a SS7 network with a cheaper and higher capacity alternative. This is a SIGTRAN protocol allowing to access a remote MTP2 implementation on a Signaling Gateway. The configuration of MTP3 in Yate describes the type and Point Codes of each attached linkset. Cisco Signalling Link Transport is a backhauling protocol that transports remotely MTP2 links on a Cisco Access Server. The transport protocol is Reliable UDP (RUDP). The Mediant 2000 and Mediant 3000 series from Audiocodes support standard SIGTRAN M2UA backhaul for SS7 links. The SCCP protocol provides additional signaling commonly used by mobile networks. In most cases the Global Title Translation needs also to be configured. |
Troubleshooting
| Issue | |
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What to do if you suspect Yate is not connected to the database You can use tcmpdump to follow VoIP traffic |
How to solve 'Headers already sent' warning for YateAdmin Various issues: STUN, kdoc, YateClient on Ubuntu 11.10 |
See also
