- This topic has 10 replies, 1 voice, and was last updated 10 years, 6 months ago by MikeM to Igor.
19th December 2006 at 14:35 #31251newbie_asteGuest
Hi, I have a quitum tenor dx2030 and asterisk at our office. I have able to setup the T1/E1 of the quintum dx2030. Calls are being passed to our switch(Nortel). When I dial the the handset number i was able to hear a ringback on the sip-phone that is registered to asterisk. But my problem is that I cannot hear some ringing on the handset. Below is my voip call flow. Please help on this one, I really needs some help. By the way I’m using a R2 signalling from the Philippines. Thanks
newbie_aste19th December 2006 at 15:07 #31252MikeM to Newbie_asteGuest
Well the first problem is that you are using R2. That is one of the worst signaling protocols to have been developed and must have been created by some sadistic Bulgarian (Sorry TSG). There are about 12 to 18 different settings that need to be spot on accurate to get working. Quintum does have some country specific R2 settings and I think there is one for the Philippines, but there is no guarantee that this will work. You should review the document and ask the local E1 provider to check on the settings for all of this and verify.
firstname.lastname@example.org December 2006 at 07:35 #31253newbie_asteGuest
HI MikeM, We already have a working quintum here at our office. I can’t see anything on the quintum cli when I issue the ev qu command. But when I tried issuing the lastcall command, I have seen the call details such as the number that I called and the ANI. Are there any docs in setting up an the E1/CAS of the quintum?How can I debug the quintum? Thank you very much MikeM. This is really a big help. Please help to configure this one. Thanks20th December 2006 at 14:05 #31254MikeM to Newbie_asteGuest
Because there are so many ways to setup the E1 there is not any one document that will provide all the information. As I said, when it comes to R2, Quintum does have some documents on their web site and they even have one for R2 in the Philippines that worked at one time in the past, but there is no guarantee that this will still work especially if the telco changed anything. You can find this document under the support pages and search for R2.
As for the event log, you need to enable the events that you want to see, such as;
ev l3 ch cas
email@example.com December 2006 at 18:27 #31255newbie_asteGuest
First of all I wanted to thank you for the inputs. I have successfully setup the tenordx using sip(asterisk). I was able to pass the call from my asterisk to my TDM/Pstn network. I wanted to ask how am I going to transfer the calls from my TDM/PSTN network back to my asterisk? What would be the dialplan it would follow? Will I add the numbers from my hosted asterisk number in the “hopoff Directory”?Or will I add it on the static?
Please help me. Thank you very much29th December 2006 at 22:49 #31256MikeM to newbie_asteGuest
There are many different ways that you can go from PSTN to IP and it depends on how you want to do this. You also need to have the tenor setup to register to a sip server/proxy to originate using sip.
Hop-off is only for call that go from IP to PSTN. When you are dealing with SIP, as long as you have the sip server and proxy setup in the tenor, then you need only concern yourself with the tcrg area and how you want to do inbound pstn to IP.
If you need more help, you can contact me directly at firstname.lastname@example.org and we can discuss this some
Mike11th February 2007 at 05:09 #31257jysolarGuest
Could you post your configuration files to run asterisk with quintum?
I like to try it…
Thank you.16th November 2009 at 09:07 #31258StasgharGuest
Can You help me I am also trying to connect Trixbox with DX 4120 thru SIP Trunk. How I will do it?18th August 2010 at 20:25 #31259olaoluGuest
Can You help me I am also trying to connect latest Trixbox with DX 4120 and Dx60 in three different locations thru SIP Trunk. How I will do it?24th August 2010 at 20:55 #31260MikeM to olaoluGuest
There are many ways to do this depending on what your full need is. Please contact me directly to discuss your application.30th August 2010 at 12:35 #31261MikeM to IgorGuest
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