Blended Carriers

[omarrodriguezt]

Hello,
There is any way to dial an outbound campaign using blended carriers?

I mean, to use more than 1 carrier automatically in a live campaign.
Thank you in advance.

[mflorell]

We have been over this a couple times, there is nothing built-in that easily lets you blend carriers. You can do the trailing-digit method of blending, as well as the failover method for using more than one carrier, but we have not devised a systematic way to blend outbound calls across multiple carriers.

[omarrodriguezt]

Indeed.
Thank you for answer.

[marmu]

this is really something we need... since every ISDN/SIP-interface we use to dial out is represented as a carrier. Without using multiple carriers we could only dial out to 30 lines with one server.

as far as I understand this:
failover method results in something like this?

Code: Select all

exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@carrier1,,Tor)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@carrer2,,Tor)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@carrier3,,Tor)

I think I get this "failover-method". Can someone tell me how good this performs? And what is exactly a "failover"? Will carrier2 be used, if there are no more free channels on carrier1?

But what is the trailing-digit method?
Does it mean we have to setup prefixes for each campaign, which will result in assigning carriers to campaigns? This is not what we are looking for. I guess we'd have to use the failover-method.

searched the forum for "multiple carrier(s)", "blending", "blend" etc. Didn't find so much info...

Our goal is 240 lines on several servers. Of Course we want as much lines per server as possible, which means havin several carriers (per dailing-server).

Would it be a better Idea to have one carrier within vicidial and code something within asterisk for distributing the calls?

Thanks for any help in advance.

Cheers,
Marcus

[mflorell]

For very large setups we typically set up a gateway using Asterisk and/or OpenSIPs to distribute calls.

The method you describe will result in some logging issues because the termination of the call will possibly be logged more than once.

The trailing digit is mentioned a couple times in the forums already, but here it is again for you:

exten => _1NXXNXXXXX[0-2],1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXX[0-2],n,Dial(SIP/${EXTEN}@carrier1,,Tor)

exten => _1NXXNXXXXX[3-5],1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXX[3-5],n,Dial(SIP/${EXTEN}@carrier2,,Tor)

exten => _1NXXNXXXXX[6-9],1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXX[6-9],n,Dial(SIP/${EXTEN}@carrier3,,Tor)


This will distribute calls mostly proportional across the 3 carriers without logging issues.

[marmu]

thanks matt! I really looked through the forum before posting...

I will check that out as soon as I got multiple carriers in one server. Had to check that before investing.

[marmu]

This works quite well for us now. The only problem is that calls are not being equally distributed between the two carriers this way. We thought about writing some code for that, but this doesn't seem to be required. The system works and we are spending our time on more important stuff right now

[omarrodriguezt]

Sounds good

[williamconley]

For even distribution you could use "groups" in asterisk channels and set up a counter, calling the carrier with the least active calls. It's more complex, but should not noticeably increase server load. You can even put in maximums per carrier that way (using groups and counting active calls per group).