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Unread 10-10-2008, 12:55 AM   #4
emoci
 
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Meantime another glitch – calling from my cell to Sipbroker PSTN – hear the answer, can call my alias, it connects. Now calling from Packet8 phone to Sipbroker PSTN – hear the answer, dialing the alias – hear busy signal. At that my ATA does not receive any SIP request. Tried to call other numbers via Sipbroker PSTN – same story – can call from cell, but can not call from Packet8. Any explanation?

---oleg
Ok,

1. New users posts are automatically moderated when they post links, hence why it took a while to see your post

2. One of the main reasons to bring VoXalot into play is to avoid forwarding from one line to the next

3. I'm gonna try explaining a bit of the background on SipBroker, but keep in mind I am not 100% clear on it either.

A good chunk of the PSTN access numbers have been set to use re-invites. As a method this ensures more users can be accomodated.

It however uses a method where SipBroker stands in the middle sending out messages to both ends of a call on how to reach each other, but does not get involved in the actual audio stream, so the two ends have a direct connection to each other for audio.

That said when the calling party is using a VoIP line it is possible that it has compatibility issues with Re-Invites and hence your trouble...

When you place a call via SipBroker the common path is:

PSTN Caller>> Provider of PSTN Number (eg. CallCentric) >>SipBroker Server >> Network You're Calling>> Called User's Device

Re-invites make it possible that despite that long path audio is actually just:
PSTN Caller (which is assigned a temporary adress by the SipBroker Server)>>Called User's Device

Now in your situation the path would look like:

Packet8>> Packet8 Termination Provider>> PSTN>>Provider of PSTN Number (eg. CallCentric) >>SipBroker Server >> Network You're Calling>> Called User's Device

Re-Invites will attempt to reduce that to:
Packet8>>Called User's Device
for audio purposes.

If Packet8 or the Packet8 termination provider do not support Re-Invites, you may run into trouble.

Unfortunately the SipBroker PSTN access numbers were meant to be used from the PSTN. It is true they do work fine when called with most VoIP lines as well but incompatibilites can lead to issues...

4. Alternatively it could also be simply a DTMF issue where the device/softphone you have with Packet8 is not sending digit tones properly

One thing I would try for now, is to add a # to the end of the dialed number (So *010123456#).

If is it the case I described in (3) there isn't much that we can do. Question though: Does Packet8 allow direct SIP URI calls? (that could a solution)
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