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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
sip-phone-asterisk–>quintum–>Nortel switch–>handset
Best Regards,
newbie_asteWell 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.
MikeM
mike_voip@hotmail.comHI 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. Thanks
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
ev c
ev quMikeM
mike_voip@hotmail.comHi Mike,
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 much
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 mike_voip@hotmail.com and we can discuss this some
Mike
newbie_aste,
Could you post your configuration files to run asterisk with quintum?
I like to try it…
Thank you.
Hi Mike,
Can You help me I am also trying to connect Trixbox with DX 4120 thru SIP Trunk. How I will do it?
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?
Olaolu,
There are many ways to do this depending on what your full need is. Please contact me directly to discuss your application.
Igor,
I received your email, but it was put in my junk mail by accident and deleted. Please resend.
Thanks
mike_voip@hotmail.com
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