Scott,

Yes, and this is one of the principal reasons people choose Q.SIG.

I've worked on quite a few large voicemail servers, and these tend to do a lot of transfers for follow-me and operator features. Q.SIG support can significantly reduce the number of telephony channels needed, as not only are there zero channels in use rather than two during the transfer, but transferred calls last significantly longer on average than calls to leave or retrieve messages.

You do need to check that the remote end supports this; some older PBXs only support parts of the Q.SIG standard.

Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/


Scott Stingel wrote:
Alistair-

Good writeup! Question regarding Q.SIG: Can it be used to solve the problem of signaling a remote switch to "take a call back" and extend it to another channel instead? This, as you know, is always a challenge when using IVR in a call centre environment, when one wants to extend an IVR call to a live operator without holding up channels in the IVR.

Regards,
Scott Stingel
Emerging Voice Technology, Inc.
www.evtmedia.com


Alistair Cunningham wrote:

Eric,

E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe, which can be used for data, or can be split into 32 64Kbps telephone channels, or a mixture. If used for telephone channels, 30 of these channels can carry one telephone conversation each, and 2 carry signalling and timing information.

T1 is similar to E1. It is used in North America. It is 1.544Mbps, and can carry 24 telephone channels, each of which can carry a telephone conversation (but see below).

There are a number of protocols which can run on top of E1. Some of these are called CAS, Channel Associated Signalling. Examples are FXS loop start and E&M wink start. They provide information such as the number that was called, and what state the call is in. They're limited in what information they can carry, and are slow to set up.

A more modern protocol which overcomes these problems is ISDN. On E1, EuroISDN is the standard. On T1, there are different standards from different providers. DMS100, DMS250, NI1, and NI2 are common examples. ISDN uses one channel (called the D channel) for signalling call information. On E1, this is one of the 2 signalling channels, leaving 30 channels for voice (called B channels). On T1, there aren't any spare signalling channels, so one of the voice channels is used, leaving 23 B channels for voice.

A PRI (Primary Rate ISDN) is simply an E1 or T1 with ISDN on top of it. ISDN gives fast, reliable call setup and hangup detection, and detailed information about the call. In the UK, PRI is also called ISDN30.

An important extension to ISDN is Q.SIG, which provides extra signalling information that is used when connecting PBX systems.

An alternative to PRI is BRI (Basic Rate ISDN), which is a cheaper system for small offices. It has 2 64Kbps B channels for voice, and 1 16Kbps D channel for signalling. It is sold as an alternative to analogue telephone lines. IN the UK, it is also called ISDN2e.

I hope this answers your question! My company offers commercial support and installation services for PRI and Asterisk if you need help for specific scenarios.

This email may form the basis of a future Integrics Tip. See:

http://integrics.com/tips/

Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/


Eric Bishop wrote:

Hi all,

I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit fuzzy on this
topic. I have seen it said the PRi is a protocol that runs on top of
E1. Is this true?
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