- This topic has 11 replies, 1 voice, and was last updated 9 years, 7 months ago by Elias.
9th October 2005 at 18:06 #30251PeterGuest
I have a tenor AX, and i’ll have to configure it to forward all incoming voip calls to a logger.
The tenor has 24 voip channels but i can only configure primary or secondary accounts. I’m missing something here. It accepts multiple calls on the SIP account As a PBX?
I just cannot call the tenor on any voip account numbers.
How can I configure the sip account that i will use to make a call to the tenor? will this call be sent to the FXO/FXS ports? Basically all 24 voip channels should be mapped to the RJ11 panel. How can I make a route betw. them?
Tenor registers itself on the proxy but calls are busy.
h323-disconnect-cause = “12”
Sometimes 11. any reference?
Anyone knows what i’m missing here?
it is registered in the proxy as sip:voipno@tenorip
:ALR:3:Border Element connection lost:0:0:0:0:SUN OCT 09 13:22:47 2005
:RPT:4:Gatekeeper status (Gatekeeper(TENOR IP) register):0:0:0:0:SUN OCT 09 13:22:47 2005
RPT:2: (ossEncode failed MsgType::Reason:
, in file ras/RasMessage.cpp, line 181 ):0:0:0:0:SUN OCT 09 13:22:17 2005
sip proxy logs
User-Name = “9990301500”
Calling-Station-Id = “9990301500”
Called-Station-Id = “9990763160”
h323-call-origin = “answer”
h323-call-type = “VoIP”
h323-disconnect-cause = “12”10th October 2005 at 23:09 #30252Wilson BoyrieGuest
You could make up to 24 diferent setups, with diferent treatment for each setup.
You will need to group the ports into one or several groups, adn them apply the treatment to the group.
There is not “one step” system, where you make a stament like “channel one goes to place a”
Is more like “port trought 4 belongs to group x”
And group “x” is related to treatment “y”.
Iy makes sense after you are familiar with the boxes.
Wilson Boyrie11th October 2005 at 15:26 #30253peterGuest
Yes, I just realized how it works. I got more familiar, but not enough yet.
Now what i need to do is to catch incoming numbers, and route calls based on that number to specific interfaces. If I’m using the FXS ports what method do i need to route calls? Do I need to create hunting, hopoff, bypass?
If routing is based on caller id (is that possible?) i’ll need to create 24 “rules” one for each. Am I right?
Peter11th October 2005 at 23:06 #30254WIlson BoyrieGuest
If I read your question right, you have a AX, that means a analog gateway, with analog ports.
You could create one phone number for each “port”,or FXS interface.
That way, when calls come in from the internet, or the VOIP side, you could send the calls to the proper port.
Lets say that your country code is 989. Your area code is 777, and numbers will be 555-1000 trougth 555-1024.
In theory, you could create 24 groups, with one channel each, and asign a diferent number to each group.
The calls will be routed to the diferent ports based on the number received from the VOIP side (“called number”).I do not know of any way to route the calls based on the caller I.D. on Quintum (“calling number”).
To associate phone numbers with FXS ports follow this steps:
1)Using the GUI, go to the tab “Phone FXS,Line FXO configuration”
2)Create a “associated channel group” . On the check marks, allow only one port, like port 1.
3)Repeat for all the other ports, until each port or FXS channel is asociated to a channel group.
Now, you have phisical interfaces asociated with channel groups.
Each channel group have two more asociations on the bottom.
One asociation is the “signaling group”
You could use the “cas signaling group” for all of them, since all the interfaces (FXS) are alike.
Next step: Create “line circuit routing groups”
Create one “line circuit routing group” for each number or port that you will have.
If you will have 24 ports, the best will be to create one per each port. I never have done so many, but I think that it will work.
On each “line circuit routing group”, there is a tab called “bypass/hunting”.That is where all the realtions are tied together.
Now create the phone numbers under “hunt LDN directories tab”
One entry per port, like everything else.
Go back to “line circuit routing groups”, and now you could asociate the “hunt LDN directories” into each “circuit routing group”
IF everything worked , you could do the following test:
type from the telnet prompt “cmd gkep”, and it will look like this:
Call Signal : 192.168.2.10:1720
Ras : 192.168.2.10:20000
DN : 740021 Public Ldn Priority(2)
DN : 740022 Public Ldn Priority(2)
DN : 740023 Public Ldn Priority(2)
DN : 740024 Public Ldn Priority(2)
That is a indication that the phone numbers are registering to the GK on the quintum.
You will not receive any calls into the FXS ports until this step works.
If you are absolutelly confused with the procedure, I got lightheaded even typing it.
Post your e-mail and I will contact you if you need a hand.
Wilson Boyrie.12th October 2005 at 01:30 #30255peterGuest
My email is email@example.com
Thnaks a lot.
Peter13th October 2005 at 06:34 #30256SaadGuest
Can i make a local call between two telephones connected on the same tenor? both on FXS port, or FXS to FXO? and can I connect directly a phone on fxo port? whit jus the adopters come with the tenor?
Thanks13th October 2005 at 20:01 #30257MikeMGuest
You can make calls between phones connected to the FXS port of a Tenor using the intercom feature. You may not connect phones to the FXO port of a Tenor as the FXO port can only connect to a phone line from phone company or from station side port of a PBX.
Basically, the FXS port of a Tenor gives dialtone, like a phone needs to work. The FXO port of a Tenor receives dialtone.
Mike16th October 2005 at 14:41 #30258peterGuest
I would like to be sure that calls are switched to fxo. How could I be sure of that? I’m checking ‘ev qu’ while ‘ev l3 ch’ is set.
CasManager: [2,0,2,2] Sent message to cas: Disc.
ch |01/01| 2005/10/16|10:52:44:440 |CasManager: [2,0,2,2] Received message from cas: RelComp.
what is [2,0,2,2] exaclty? is that XFS or FXO?
peter16th October 2005 at 18:57 #30259MikeMGuest
It means the following;
2 – Slot number. This is a holdover from the CMS, there really are no slots in the AX, so just ignore this.
0 – again a holdover from the CMS. Ignore this.
2 – Interface type. 1 = FXS, 2 = FXO
2 – Line/port number.
So in this case the call used line 2 of the FXO.
Mike17th October 2005 at 08:32 #30260peterGuest
Thanks a Lot!
peter7th November 2005 at 18:31 #30261AsimGuest
i need little help for the Axm
i want to use this to orignate calls using different sip accounts from some provider on fxs ports accordingly
i have done that for one fxs port but having problem in to get it work on the other lines can some one help me please
i have tried that by creating another sip signalling group and the line circuit group etc…
please let me know if i am missing some thing i will be greatful….
thanks15th April 2011 at 04:40 #30262EliasGuest
I just read your writing.You don’t finishing your writing.Please send me detail config “port trought 4 belongs to group x”.I am waiting for your response