I want to make breakout calls (extension to phone/landline number) in the UAE (since VoIP is banned here) and currently, I have a PBX server with Asterisk 1.8 installed with Wildcard TDM400P FXO VoIP card.
I am now upgrading the system and have upgraded to Asterisk 18.22.
I attempted to obtain VoIP cards like the Sangoma PCI-e AEX410, but was unable to do so due to outdated technology.
Can someone tell me if there is a workaround that would completely replace the VoIP card and allow me to accomplish my goal?
On Thursday 22 August 2024 at 10:39:59, hishamjan via Asterisk Community
wrote:
I have a PBX server with Asterisk 1.8 installed with Wildcard TDM400P FXO
VoIP card.
I would expect you should be able to use that card with current versions of
Asterisk; you need to get the DAHDI driver built, which is mostly dependent on
your kernel version, and then Asterisk can use it.
Did you try this and run into problems?
I attempted to obtain VoIP cards like the Sangoma PCI-e AEX410, but was
unable to do so due to outdated technology.
Please explain the above comment - what was the problem in obtaining FXO
cards?
Antony.
–
+++ Divide By Cucumber Error. Please Reinstall Universe And Reboot +++
Maybe, see this!
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I think the OP is confused about the meaning of VoIP, the card he is referring to is an analogue card, and also he has previously said that it is illegal to use VoIP in the country in question.
On Thursday 22 August 2024 at 11:43:07, david551 via Asterisk Community wrote:
I think the OP is confused about the meaning of VoIP, the card he is
referring to is an analogue card, and also he has previously said that it
is illegal to use VoIP in the country in question.
It is illegal to use VoIP in UAE to bypass the local telecoms provider.
Therefore anyone wanting to use VoIP (internally to an organisation) needs to
have an analogue card to interface to the PSTN to place call outside the
organisation.
So, I think the OP is looking to interface Asterisk with an analogue PSTN
connection.
Antony.
–
If you ask a Yorkshireman whether he knows the German word for “egg”,
don’t be surprised if he just smiles and says “Aye”.
We are using analog voip cards (FXO) to place calls outside the organisation. However, we cant switch the server off and remove the analog card from it as its at a single point of failure; no secondary server to be able to handle load of the main PBX incase of a disaster.
So to answer this,
Please explain the above comment - what was the problem in obtaining FXO
cards?
we did get an analog card (Sangoma PCI-e enabled AEX410) but it came with 4 FXS (S110M) modules instead of FXO (X110M or X101M) and hence, we are not connecting to the PSTN to make calls outside the org. Therefore, we are now thinking of replacing it with;
something that is not hardware dependant, rather allows to connect to with a sip trunk
Digital PRI cards for better sound quality and faster dialing
Now the question is,
I would like to work on either of the two solutions but i dont know how would the PRI cards behave or what their effect would be (if any?).
I need guidance as to what is the most viable solution in my case…
Please feel free to ask me more questions if anything i said is unclear…
On Thursday 22 August 2024 at 14:19:11, hishamjan via Asterisk Community
wrote:
we did get an analog card (Sangoma PCI-e enabled AEX410) but it came with 4
FXS (S110M) modules instead of FXO (X110M or X101M)
You cannot buy the FXO modules?
Therefore, we are now thinking of replacing it with;
something that is not hardware dependant, rather allows to connect to
with a sip trunk
Digital PRI cards for better sound quality and faster dialing
I think you have to start by asking your telephone service provider whether
they support either SIP trunks or PRI ISDN.
Only then do you know whether it’s even worth thinking about which might be a
good solution for you.
In Europe it would generally be impossible to obtain a PRI connection these
days, and the telephone service providers are encouraging anyone who still has
one to switch to SIP, however I have no idea what the situation in UAE is.
There’s no point in you deciding to use something that’s not supported by the
company you have to connect to, though.
Antony.
–
I bought a book on memory techniques, but I’ve forgotten where I put it.
Yes @Pooh , my carrier does support SIP trunks and PRI.
You cannot buy the FXO modules?
Unfortunately, no. Have tried obtaining them from different vendors, but all of them say the same thing - it is an old technology, it is no longer available.
On Thursday 22 August 2024 at 15:51:37, hishamjan via Asterisk Community
wrote:
Yes @Pooh, my carrier does support SIP trunks and PRI.
In that case I would go for SIP without a doubt. It requires no dedicated
hardware (so, reduces costs, and eliminates “the server with the interface
card in it” as being a single point of failure), keeps the technology simple
(you’re only using one signalling protocol throughout the entire system), and
is almost certainly the best-known and most widely-used signalling protocol
with Asterisk, therefore you’ll get the best assistance with any questions
which come up.
Antony.
–
“The future is already here. It’s just not evenly distributed yet.”