Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: del on October 08, 2009, 06:35:24 PM
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If you had a router that was acting as a DHCP server then installed a SARK PBX would you turn off the router's DHCP and use the SARK box? I can see that using the SARK box would make the setting up of VoIP handsets easier, are there any other advantages? What would you do? Also what's the best, as in reliable and easy to setup trunk card for SARK. Thanks in advance.
Del
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Hi Del
Unless your router can do DHCP66 (most domestic and low end stuff can't) then I would use SARK every time unless I was only planning to roll out a very few phones or phones that don't support provisioning. Of course, with SME server running server-gateway mode, you can have the best of both worlds because you can have eth1 running in the customer LAN, so the customer can do its own DHCP up there and you can have the phones running under eth0 and getting DHCP from SME Server/SARK.
Easy trunk cards.... What kind of trunk? PRI/BRI/Analogue?
Best
S
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Analogue cards, can you elaborate on the Server-Gateway setup? Would you need 2 networks to do this? I'm sorry for being a bit slow but I'm trying to imagine how the phones would get a different IP address if they were plugged into the same network cabling. :???:
Del
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Analohue cards.... Digium and Digium clones are all the same. If you can afford it, get one with inbuilt echo cancellation, it will make life easier. The only others that I know of are Sangoma and we don't currently support them.
...I'm trying to imagine how the phones would get a different IP address if they were plugged into the same network cabling.
They aren't, you have two network switches; one which eth1 plugs into (on the customers data LAN) and another which the phones and eth0 are connected to. This is just ordinary smeserver server-gateway mode with the customer LAN upstream and the phone LAN downstream, each on different subnets.
Best
S
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So with two networks I would need to run another network point to each desk for the phones, yes? On the Digium card, do you have a part/item or model number of the one with built in echo cancellation?
Thanks,
Del
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Seems like the TDM400P will do, I need to connect 4 trunk lines.
Del
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TDM 4xfXO with EC is designated TDM404EF in UK. This may not be the same in the USA.
So with two networks I would need to run another network point to each desk for the phones, yes?
Yes, although we have a few customers who do the same thing using a single switch and VLANs. However, you'll need VLAN capable phones (Snom, Aastra, Cisco, Polycom) if you want to run that way.
Best
S
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Analohue cards.... Digium and Digium clones are all the same. If you can afford it, get one with inbuilt echo cancellation, it will make life easier. The only others that I know of are Sangoma and we don't currently support them.
In what way will it make it easier? The TDM400P doesn't have it but you can get module for it but it costs $235.00! The TDM404EF is the same price as the TDM400P with the echo cancellation module so they both turn out tob nearly $600.00. So will my life be $300.00 easier? :smile: Another make I came across was Rhino http://www.rhinoequipment.com/R8FXXcard.html (http://www.rhinoequipment.com/R8FXXcard.html)
Thanks again,
Del
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So will my life be $300.00 easier?
If you do happen to have lines with long echo, then after a couple of weeks' messing about with tx and rx gains and echotuning to try and alleviate the problem, $300 might not seem so expensive. For that reason, you might wish to buy a card that is at least capable of taking an EC daughterboard just in case you do run into problems.
We've not tried Rhino analogue cards so can't comment
Best
S
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Depending on your provider, the echo can be REALLY annoying for people on the local server end, so the $300 could be money well spent if the lines will be used a fair amount.
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So will my life be $300.00 easier?
If you do happen to have lines with long echo, then after a couple of weeks' messing about with tx and rx gains and echotuning to try and alleviate the problem, $300 might not seem so expensive. For that reason, you might wish to buy a card that is at least capable of taking an EC daughterboard just in case you do run into problems.
We've not tried Rhino analogue cards so can't comment
Best
S
OK you've convinced me, I'm going to order a TDM404EF and be done with it :P
What is the most stable version of SARK? Is it available on your ISO? Finally, will the TDM404EF need Zaptel or DAHDI?
Thanks,
Del
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OK you've convinced me, I'm going to order a TDM404EF and be done with it :P
What is the most stable version of SARK? Is it available on your ISO? Finally, will the TDM404EF need Zaptel or DAHDI?
Thanks,
Del
I've managed to get a TDM404EF card now and I've installed it. SAIL seems to have found it OK, I've got 4 trunks and DAHDI group2 in the trunks panel :smile: The telephone company are installing my lines on Friday and I was wondering what else I need to do before I plug in the lines to the TDM404EF? This is the first time I will have used analogue lines on my PBX so any advice is welcome and very much appreciated.
Thanks,
Del
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Don't forget to set up a default route so your handsets can dial out.
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Don't forget to set up a default route so your handsets can dial out.
I've just be looking on the docs wiki and the example shows the dahdi group being chosen rather than each individual line, I was wondering if this will cause problems as it is a 4 line card but I will only being using 3 of the ports? I am thinking (I know, I shouldn't) that if all 3 lines are in use maybe Asterisk would try and use the unused line. Any thoughts?
Thanks,
Del
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AFAIK you can set up three identical dial plans for the individual ports instead a single plan for the DAHDI group.
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Hi Dave,
What I was going to do was create a route and choose line 1 for the Primary Path, line 2 for the Secondary Path and line three for the Tertiary Path, that way it should (I believe) use line 1 if available then line 2 if line 1 is in use and line 3 if line 1 and line 2 are in use. What do you think?
Del
AFAIK you can set up three identical dial plans for the individual ports instead a single plan for the DAHDI group.
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Sounds exactly right. It's been a while since I did one of these so please excuse my latenight semi-ignorance... 8-)
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Sounds exactly right. It's been a while since I did one of these so please excuse my latenight semi-ignorance... 8-)
I know the feeling :smile: My problem is I don't do this often enough, mainly due to the fact that SME/SARK just runs and therefore I don't mess unless I have to :P This is how I have my current server set up, except I am using VoIP lines at the moment, this will be my first attempt at analogue lines and there as been so many changes since my last install (Asterisk 1.2 and sail-2.2.1-617) Thanks for your input.
Del
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Hi Dave,
What I was going to do was create a route and choose line 1 for the Primary Path, line 2 for the Secondary Path and line three for the Tertiary Path, that way it should (I believe) use line 1 if available then line 2 if line 1 is in use and line 3 if line 1 and line 2 are in use. What do you think?
Del
OK, I've 3 trunks into my TDM404EF card, added 4 Snom 300 handsets and 1 X-lite ext. to SAIL and all internal calls work OK, I can dial in to all 3 trunks but I can't make outgoing calls on the trunks, it rings a couple of times and then AT&T says that I need to put a 0 or a 1 plus the area code to dial long distance, even if it's a local call. I have set up a route as follows:
Active Yes
Prmary Path DAHD2-1
Secondary Path DAHDI3-1
Tertiary Path DAHDI1-1
Quartenary Path None
Route Dial Plans _011X. _1NXXXXXXXXX _NXXXXXXXXX _NXXXXXX
Altwernate (Blank)
Authorization required? No
Description all_calls
Cluster default
I went to the Asterisk CLi and dialed out and it actually worked, then I tried again and it doesn't work anymore
Call that worked:
Verbosity was 0 and is now 3
-- Executing [13212351867@internal:1] AGI("SIP/5003-086c9d18", "selintra|OutCluster|13212351867") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script selintra completed, returning 0
-- Executing [13212351867@default:1] AGI("SIP/5003-086c9d18", "selintra|OutRoute|all_calls") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Dial) Options: (DAHDI/2/13212351867||)
-- Called 2/13212351867
-- DAHDI/2-1 answered SIP/5003-086c9d18
-- Executing [h@default:1] Hangup("SIP/5003-086c9d18", "") in new stack
== Spawn h extension (default, h, 1) exited non-zero on 'SIP/5003-086c9d18'
-- Hungup 'DAHDI/2-1'
== Spawn extension (default, 13212351867, 1) exited non-zero on 'SIP/5003-086c 9d18'
Call that failed
Verbosity is at least 3
-- Executing [13212351867@internal:1] AGI("SIP/5003-086d0e10", "selintra|OutCluster|13212351867") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script selintra completed, returning 0
-- Executing [13212351867@default:1] AGI("SIP/5003-086d0e10", "selintra|OutRoute|all_calls") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Dial) Options: (DAHDI/2/13212351867||)
-- Called 2/13212351867
-- DAHDI/2-1 answered SIP/5003-086d0e10
-- Executing [h@default:1] Hangup("SIP/5003-086d0e10", "") in new stack
== Spawn h extension (default, h, 1) exited non-zero on 'SIP/5003-086d0e10'
-- Hungup 'DAHDI/2-1'
== Spawn extension (default, 13212351867, 1) exited non-zero on 'SIP/5003-086d0e10'
Can anyone see anything wrong with it?
Thanks,
Del
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I have been told by the Digium dealer that I need to add www to the dialing prefix to ad a short pause but I can't find where to add it, they use FreePBX and I can see where it goes there so can any one point me in the right direction?
Thanks,
Del
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And you want a delay because....
?
S
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Apparently the extensions are dialing before AT&T's line is available and therefore it's missing the first digit dialed :???: