- This topic has 11 replies, 1 voice, and was last updated 11 years, 1 month ago by MikeM to Dirceu Ciupka.
4th January 2008 at 17:52 #31649SipGuest
i want to configure a quintum dx for sip calls, i’ve read a sip document in quintum site and i did all the suggested parameters, but i can’t make a call, i’m asking if you have any idea of how to configure dx for sip, and how can i konw that the quintum is registred in the sip proxy
thank you for your help
Med7th January 2008 at 07:39 #31650TheLostPacketGuest
What are you registering your DX unit to? Are you the admin of this software or are you trying to register to a third party service? You will need to configure the SIP parameters, t1/e1 parameters and properly set the dialplan. If you would like specific help on your setup you can email me at.
Thanks3rd April 2008 at 08:34 #31651ddddGuest
how to use quintum a800 with sip configuration to make 2 port of pbx on quintum,,,,?
thanks you for your question..3rd April 2008 at 23:31 #31652MikeM to ddddGuest
If you are talking about the older generation 1 A800, you need to understand a couple of things;
1. Quintum no longer makes nor supports this model.
2. There is a “BETA” sip software on their web site for this model, but like the A800, it is not supported. Worse yet, because it is beta, it was never fully tested. It was actually created in 2004 so it is 4 years old with no updates since or after.
You can try to use this, but I have seen about 50% success rate with this software and the unit. If your requirement is SIP, I suggest you purchase the newer quintum, AX800 that fully supports SIP.
email@example.com April 2008 at 04:26 #31653ddddGuest
hallo, tahnks for your answer….
my the software on the quintum A800 is have updated with :
Product Name: Tenor Analog A800 Multipath Switch – 8 ports (Rev. B)
Gatekeeper Status: Mini
GK Calls Allowed: 8
Feature Bit Status: -PS/+RB/-ER
Languages allowed: 1
Serial Number: A002-0082A9
Ethernet Address: 00-30-E1-00-82-A9
IP Address: 192.168.68.3
Subnet Mask: 255.255.255.0
Default Gateway: 192.168.68.1
System Software Version: P5-2-1(LEC) (1678285/0xFF74)
Boot Software Version: P4-1-3 (180592/0xE814)
Database Version: 2.08 09-13-2000 (278376)
that is dekripsion config of my quintum. so how to make this quintum work in the port SIP.
thanks for your answer…
please answer Now.
thanks you9th April 2008 at 11:58 #31654MikeM to DDDDGuest
“Please answer Now” a little impatient it seems.
In any case, if you would like me to provide you the full config, but again, there is no guarantee that this this will work, you may send me a message direct to my email of
MikeM26th June 2009 at 15:14 #31655SeyiGuest
i want to configure Tenor dx for h323 config, i want it to work with AVAYA S8720 with both equipment should be on the same setup.how do i do it.
thanks you all.29th June 2009 at 14:38 #31656SardarGuest
i want to configure a quintum dx for sip calls, i’m asking if you have any idea of how to configure dx for sip?
thank you for your help4th July 2009 at 14:05 #31657MikeM to SardarGuest
that is pretty general question. There are many ways to configure quintum for use with SIP. If it is orig, you will need to configure the quintum to connect through a sip server (you must provide). If it is for term, then you do not need to do anything different than you would for H323.
If you need further assistance, please contact me directly at
firstname.lastname@example.org October 2009 at 14:01 #31658Abrar AslamGuest
I want to change the configuration of Quintum Tenor 960 from H323 to SIP. Please explain me what setting do I need to change.8th April 2010 at 19:49 #31659Dirceu CiupkaGuest
I am tryng to setup a Quintum DX 4060.
One side is a SIP TRUNK provided by a Asterisk PBX, and other side a PSTN 2 Mb MFC-R2.
I can do orig calls, but dont incoming calls.
Can you help me with any information?
Dirceu Ciupka12th April 2010 at 16:49 #31660MikeM to Dirceu CiupkaGuest
There is a document on quintum’s web site that may help on this. Just search for SIP on there web site. The basic is that you need to set the quintum to register to your sip server and then your sip server will control the routing.