1 vicidial and 2 asterisk

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1 vicidial and 2 asterisk

Postby chill_master » Thu Sep 24, 2009 1:14 pm

Is it possible to setup 1 vicidial using 2 asterisk server? and so, can anyone point me to a resource where I can see the details of how it is done?


Thank you. :D
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Postby gardo » Fri Sep 25, 2009 11:14 am

What do you mean 1 Vicidial and 2 Asterisk servers? Do you mean a multi-server setup?
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Postby chill_master » Fri Sep 25, 2009 11:35 am

yes I think so.

like

machine A:
vicidial
asteriskA

machine B:
asteriskB


vicidial in machine A utilizes also asteriskB in machine B.
I wanted to to do this for less bandwidth consumption, where I wanted like 5 agents will be using the asteriskA passing thru ISP A and 5 agents using asteriskB passing thru ISP B.

so I don't need to pay for a big bandwidth in one ISP.
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Postby chill_master » Mon Sep 28, 2009 12:28 pm

anybody?
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Postby okli » Mon Sep 28, 2009 5:16 pm

Will you be using two different VoIP providers?
If it's still a single one- can you register 2 IPs to one account?

For 5 (10) agents how much is your bandwidth consumption? Can't you just use a codec with better compression- gsm or g729 if not doing so already?

If it's 2 providers you could just put another NIC and route all packets to the IP of the second provider through the second NIC, also change dialplan as in here:
http://www.vicidial.org/VICIDIALforum/v ... 4775#34775
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Postby chill_master » Mon Sep 28, 2009 5:48 pm

I will be using two different VOIP, later on I will let go of the other VOIP and just use one.

I'm using g711. I've read reviews of g729 that the quality is not really good. I was opting to use iLBC but it's hard to find a VOIP provider that implements it with a good review.

for 10 agents = 836.06 Kbps

how can you make asterisk route the call to a specific NIC?
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Postby williamconley » Mon Sep 28, 2009 8:52 pm

Yes, clustering is a fairly standard practice. Have you purchased a manager's manual yet? For this, that is your best bet (next to paying someone to do it for you).

Under those circumstances, you CAN both use the same provider, unless you have some other need to use two different providers.

If you have two providers, you can have one vicidial box use two NICs via IP tables

OR you can even attempt one NIC with two gateways using IP tables. In both cases, the reason that it will work is that the routing can be directed simply by the target IP address.
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Postby okli » Tue Sep 29, 2009 2:05 am

chill_master wrote:I will be using two different VOIP, later on I will let go of the other VOIP and just use one.
Not quite sure how you could use 2 NICs or 2 asterisks with one account to one provider.
If opting for one provider, you may need another account with it and another asterisk box used as trunk, or in load balancing setup with number of trunks split between the two.
There could be better solutions perhaps, can't think of any easy right now.

chill_master wrote:I'm using g711. I've read reviews of g729 that the quality is not really good. I was opting to use iLBC but it's hard to find a VOIP provider that implements it with a good review.
Try GSM, try g729 too, and YOU hear how it sounds, then conclude.

chill_master wrote:how can you make asterisk route the call to a specific NIC?
You make 2 accounts to 2 providers in asterisk. Next you route the packets to the appropriate interface.

For example in our Debian box we have this:
/etc/network/interfaces :
Code: Select all
iface eth1 inet static
       pre-up /usr/sbin/ethtool -s eth1 speed 100 duplex full autoneg off
       address 10.17.17.4
       netmask 255.255.255.0
       network 10.17.17.0
       broadcast 10.17.17.255
       up route add -net 10.17.17.0 netmask 255.255.255.0 dev eth1
       down route del -net 10.17.17.0 netmask 255.255.255.0 dev eth1
Similar for eth0.
Figure out what's the CentOS/RedHat equivalent.
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Postby williamconley » Tue Sep 29, 2009 10:04 am

okli wrote:
chill_master wrote:I will be using two different VOIP, later on I will let go of the other VOIP and just use one.
Not quite sure how you could use 2 NICs or 2 asterisks with one account to one provider.
If opting for one provider, you may need another account with it and another asterisk box used as trunk, or in load balancing setup with number of trunks split between the two.
There could be better solutions perhaps, can't think of any easy right now.
On the contrary, we use MANY asterisk boxes with one "account" on one provider. In some, we need merely list the IP address, in others they allow for the creation of a new "device" (for authentication purposes), but the $$ is a single account. After all, a company with 5 locations is not likely to open 5 accounts for telephone servers and have to keep track of balances in all of them.
chill_master wrote:I'm using g711. I've read reviews of g729 that the quality is not really good. I was opting to use iLBC but it's hard to find a VOIP provider that implements it with a good review.
Try GSM, try g729 too, and YOU hear how it sounds, then conclude.
I agree wholeheartedly, gsm & g729 are excellent, but everyone has their own experience. A lot of factors are involved (stability of your network, load on your server, distance to your VOIP, reliability of the network path to your VOIP ...). Trying to "calculate" is pointless, it's much easier to test. But remember: ONE good call on g729 does not mean you will be able to make 40! I do have clients who use ulaw for a specific number of trunks, and above that switch to g729 to allow for enough lines for overflow.
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Postby okli » Tue Sep 29, 2009 5:10 pm

williamconley wrote:On the contrary, we use MANY asterisk boxes with one "account" on one provider. In some, we need merely list the IP address, in others they allow for the creation of a new "device" (for authentication purposes), but the $$ is a single account. After all, a company with 5 locations is not likely to open 5 accounts for telephone servers and have to keep track of balances in all of them.
My though was for account as username@provider. If there are username1 and username2 with the same VoIP provider having one IP, how would you route the calls via 2 ISPs if OP goes for the 2 NICs option? This is what I can't figure out.
All other variations, with 2 asterisks or 2 IPs of the provider, should be easily doable, but interesting is the above scenario :)
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Postby williamconley » Tue Sep 29, 2009 9:53 pm

okli wrote:My though was for account as username@provider. If there are username1 and username2 with the same VoIP provider having one IP, how would you route the calls via 2 ISPs if OP goes for the 2 NICs option? This is what I can't figure out.
All other variations, with 2 asterisks or 2 IPs of the provider, should be easily doable, but interesting is the above scenario :)
you're gonna have to slow down for me, dude. i'm old. too many acronyms, i lost the question entirely.

2 ISPs (Internet Service Providers)?
OP? (What is an OP?)
2 NIC - this i get.

As I said, My VOIPs (Trunk Providers) allow two methods of authentication: IP & user/pwd. Obviously, if IP is used you merely give the provider both IP addresses and you have no worries.

If you use "user/pwd", once again, you'll have them both and can use them on whichever machine you like, as long as they are correct your call will work.

Now for Inbound calls, you must specify which IP or credentials will get the calls (and if you choose credentials, you must register to accept the calls or they won't know where to send them).
So ask again, sir, cuz ya lost me.
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Postby okli » Wed Sep 30, 2009 2:16 am

Sorry : )
OP- original poster, chill_master in this thread.

Here is it rephrased- 1 asterisk/vicidial with 2 network cards and 2 internet providers.

Assuming there will be only one VoIP provider, who presumably has 1 IP to connect/trunk to, how would you route the calls out of the vicidial box, so both ISPs are utilized, thus lowering the traffic through each of them?

Even if you create another username (account) with that provider, no matter what method of authentication, that new account is still on the same IP, the VoIP provider's IP. So calls cannot be routed based on destination IP, using ROUTE or IPTABLES.

This scenario is closest to the one the original poster is going to have, once he let go the second VoIP provider, assuming a second asterisk box is not going to be used.
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Postby williamconley » Wed Sep 30, 2009 2:35 am

his goal was to use two T1s and his original method was two servers, and two providers.

the impression i got was that he wanted to be able to use more bandwidth than a single T1 would allow, so he was going to split his server and use two providers to "duplicate" his system, but still use one vicidial to control it all so he would not hve to do extra work to run it on a day to day basis.

point is that you don't need two providers simply because you have two servers. you can have 75 servers with one provider, whether they go through the same T1 or several geographically remote servers, one provider is fine.

now, if you have ONE server: you are right that it is not easy to use the same provider yet route through two T1s from a single server (to allow for extra bandwidth when the T1s cannot be married)

in this case, we generally use the port as the routing trigger (you can route based on port in addition to just IP address). as long as your provider can supply SIP on more than one port, you're good to go. not easy, but doable if you have a Net+ technician (or a brainiac or a guru, or just someone really stubborn).
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Postby okli » Wed Sep 30, 2009 3:08 am

I don't think a second box is present yet and the original request was different:
I wanted to to do this for less bandwidth consumption, where I wanted like 5 agents will be using the asteriskA passing thru ISP A and 5 agents using asteriskB passing thru ISP B.

so I don't need to pay for a big bandwidth in one ISP.


The continuation was that setting up a second asterisk box is simply overkill for 10 agents in total, just to lower down bandwidth consumption, given the fact another ISP is going to be used.
A second NIC with another VoIP provider or another account with the same provider, but different IP/port would work just fine, with destination IP/port based routing. Simple and no much to discuss on that.

I hoped you had something under your hat ( white pigeon? ;) ), for the case with 2 NICs/ISPs and a single VoIP provider/single account/trunk/IP/port to it. This would be interesting to figure out.
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Postby chill_master » Thu Oct 01, 2009 6:23 pm

I will have to go back to this at a later time. There's a much higher priority issue that i have yet to fix. About SIP/2.0 404 Number not in e164 format.
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Postby williamconley » Thu Oct 01, 2009 8:37 pm

okli wrote:I hoped you had something under your hat ( white pigeon? ;) ), for the case with 2 NICs/ISPs and a single VoIP provider/single account/trunk/IP/port to it. This would be interesting to figure out.
Yep, gotta at least have different port. major VoIP providers will ask the port to send the calls to, which then allows the use of several ports, and then you can route based on port.

some call it an "endpoint" some call it a "device" but almost all have a way of creating more than one "connection" between your servers and theirs. some even have multiple servers to which you can send traffic (makes it even easier, if quality can be maintained). and these connections often contain the ability to change the port #.
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Postby chill_master » Fri Oct 02, 2009 10:33 am

I signed up to a VOIP provider and they ask me to open my 1000-64000 UDP port for RTP. Isn't that rediculous? talking about security.
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