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VoIP Abuse Project

By popular request, we’ve convinced someone from the VoIP Abuse Project to join us tomorrow at noon on VUC. I think many of you will be interested in this topic, so please come by, join in and ask questions.

http://vuc.me for all connection info and links to VoIP Abuse Project

A couple of other features and announcements are scheduled after that segment, including an Astricon update and maybe even something about a mysterious book I have heard about.

Call in from 11:40 AM SIP:200901@login.zipdx.com – prefer g722 and accept g711 – Skype:vuc.me – Call widget on the site during conference hours.

See you there!


Use modprobe to find E1/T1 jumper setting on PRI card

Does anyone know if I could use modprobe command to find out rather than set the jumper on a Digium PRI card?

I want to find out the jumper settings on the card without opening the box which will cause down time.



How To Pick A Codec On The Fly In Asterisk

I’m trying to test an IVR system with recorded prompts and would like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234 ulaw, etc. I know I can set up 3 users where #1 is gsm,  #2 is ulaw and #3 is slin; Need it the other way so I can do DAHDI–> IAX testing.

Daniel Tryba (daniel at tryba dot nl) came in with this interesting solution:

exten => 1234,1,Set(SIP_CODEC=alaw)
exten => 1234,n,Goto(0234,1)
exten => 2234,1,Set(SIP_CODEC=slin)
exten => 2234,n,Goto(0234,1)

Should do the trick.

Asterisk Cluster Scenario

My company has experience in setting up single asterisk setup, but recently one of our customers asked us to set up an asterisk cluster, that must be High Availability and Load Balanced. So I wrote here to have some hint or advice about the configuration we thought.

First of all I’ll explain you the Asterisk Cluster scenario:

The asterisk cluster must serve as Call Center “Hub-and-Spoke”. There are lots of little call-centers (~15 ops), phisically separated, that must appears as a single “Big Virtual Call-Center”. The call-center is inbound and outbound.

Initially the “Virtual Call-Center” will be composed of 15 small call-centers (= ~230 ops), but the prevision is to grow to ~800 ops.

Asterisk must work as B2BUA, because the customer does not want the call-centers contact directly the upstream SIP providers. Asterisk server must also record the call and, maybe, do some transcoding.

The single asterisks instances talk each-other with DUNDi/IAX2 to pass the calls from an op to another op registered on another server. We’ll use a mysql database (NDB cluster) to avoid to have multiple sip.conf files with the registration informations (asterisk-rt). We must, in any case, use asterisk, because the whole call-center has to be integrated with a 3rd part software for predictive calling ad power dialing which has been developed for asterisk.

The SIP servers are behind NAT, because the customer want to use only one public IP address; every asterisk server will have a different RTP port range to avoid rtp conflicts.

And, for now, the LB part works manually registering the ops on different asterisk servers. But now we have to achieve the HA part of the setup.

We started considering a SIP hardware load balancer, like Radware or Brocade, but neither a single nor a couple of this equipments (HA, remember?) do not fit in the customer budget. So, we started thinking about use OpenSIPS (formerly OpenSER) as a SIP load-balancer. (the customer has server hardware that could be used for this, and we will use VRRP for redundancy).

In the OpenSIPS features we read “load balancing with failover”, but we could not find any useful and complete configuration example. Is OpenSIPS able to know if an asterisk server is UP or DOWN, or must we use a 3rd part tool, like mon?

Anyone has experience in use OpenSIPS as SIP load balancer (not to work as “real” SIP proxy)?

Any hint/advice for this part, or for the global setup?

Thanks in advance for the help!

PS: sorry for my weak english
PS2: sorry for cross posting but, if the general setup is more asterisk related, the OpenSIPS part is, obviously, OpenSIP specific.

SIP Phones: Lost When Internet Goes Down In Asterisk

Its a long and old thread, haven’t read it all, but just to let you know this happens when there is no reply from the DNS. So change DNS or install it locally on your asterisk server. At least caching name server should be installed.

—Zeeshan A Zakaria

Asterisk And Digium TC400B

Because of the heavy load and the high expectations of an asterisk server offered as a solution integrated with our CRM software, we were looking into other possibilities than software Licenses for G729 and G723 codecs. To lower the pressure on the processor giving it more space to do more work. We heard of a hardware (PCI CARDS) can be used with Asterisk that does the work. And we stumbled with Digium TC400B.

Could be a newbie’s question.. but does that serve our needs? As we have not pressured a server before up to 1400 extensions with 600 outbound SIP calls (customer’s needs). The server in question is Core I7 16 GB ram and Raid 10 SAS drives. We need to know how many calls with G729 or G723 can this server handle? And as far as we can see this Digium TC400B card can be a cheaper solution If calculating the CPU cost plus the licenses for each channel. One more question.. can we add two of those cards to the server? Will it be efficient?


—Tarek Sawah