SS7 + mediant 2000 + Asterisk

Hello

I ned Help from configure the mediant 2000 for work with SIP to Asterisk
and SS7 chanel

I was able to setup an Audiocodes Mediant 1000 E1 SIP as pstn for Asterisk but I don’t know nothing about SS7, so if you need I can try help you with the SIP part, however you have to detail much better which problems you have if you want help from someone here :smile:

Regards.

hi,
dont worrry friend i worked on ss7 with asterisk communication. so wht type of help u need, just reply me, i m here to help u
i made a ss7 link with asterisk from siemens v12 exchange. wht type of help u need ?

regards,

[b]Jehanzaib Younis

+92.111.874.874 Ext: 2558
+92.321.5139853
jehanzaib.younis@trgcustomersolutions.com
trgcustomersolutions.com[/b]

Hello
I need configure SS7 + asterisk

Hardware Compaq proliant 360 GD2
CPU 1.4 Ghz *2
RAM 1 Gb
HDD 18.2 *2 RAID 1

Digium TE110P card

how to resolve problem with SS7

Hi,

Can you please help for intial setting up of audiocodes mediant 1000 E1 line with asterisk?

I have hardwares… and seeking a documentation and help to move further…

Thanks in advance
Rizwan

I’m sorry but I don’t have any docs about but should be not too difficult, just register the Mediant to * and setup its E1 interface, here is how I defined the Mediant in sip.conf:

Try and if something goes wrong ask me a specific question and I’ll try to help you.

Regards.

Marco Bruni

Bruni,

Excellent and thanks for your quick response. I appreciate your help.

In mediant 1000 /2000, how should i route all the calls landing in E1 to asterisk. It will be very much helpfull , if you can send me some screen shorts of mediant with sample.

And same from asterisk to mediant routing in asterisk. (is it like simple extension routing calls ?)

I understand there is nothing to be do in zaptel.conf and zapata.conf as for this setup. Am i right?

Looking forward your response.

Thanks,
Rizwan

When the Mediant is registered to * it will routes every call through * by default, just check the Mediant uses * as its proxy (Protocol Definition -> Proxy & Registration -> Enable Proxy) and registrar (Protocol Definition -> Proxy & Registration -> Enable Registration) and it doesn’t use its internal routing table (Protocol Definition -> Proxy & Registration -> Prefer Routing Table).

Yes, the following is a sample:
exten=>_X.,Dial(SIP/mediant/${EXTEN})

Yes, only the sip protocol is involved.

Regards.

Marco Bruni

Hi,

Please find my proposed setup and requirement and needs your help on the same.
have following setup

E1<----->Audiocdes1<–>Asterisk <— >agents.

Now i am planning for H/A and L/b. So i would to like to setup like the following blocks.

E1<----->Audiocdes1<–>Asterisk 1<— >| Common agents for |both asterisk involved and
E2<----->Audiocdes2<–>Asterisk 2<— >| queing involved in it.

As for this setup, i wish to prevent the single point of failure starts from lines,device and asterisk and balance load and between asterisk , audiocdes.

i have gone through the

voip-info.org/wiki/view/Aste … +Solutions

After this going through this page ,little bit confused, which options to select for this setup.

It would appreciate , if you experts advice a solution to go with this setup for H/a and load balancing.

Posted: 21 Nov 2007 23:11 Post subject:


Hi,

As for my plan, only one gateway (audiocodes1) will be active for incoming calls(E1 - active),Second e1 line and audiocodes2 will be act like passive for incoming calls.

So for E1-Audiocodes (A/P) setup.

When calls lands in the audiocodes, i can program in such a way, audiocodes will forward this to both asterisk in some fashion, such that both asterisk will be used simultaneously, also i can program in audiocodes with to forward all the calls if one asterisk fails, to take care of single point of failure.

In this case calls will be routed to both asterisks by some way. asterisks sould be in Active/Active. Both asterisk have identical set of configurations sip,extensions,etc with different ip address configured in nic.

how sould i configure the sip clients, in such way if one asterisk fails , the asterisk to take care of the load from its peers.

Also both are active, load should be shared.

if in case agent 1 registersd to asterisk1 and agent2 registered in asterisk2.

Now a call lands on asterisk2,after ivr , external user wants to talk to agent1, at this situation, how to handle this ? Since agents not registered in asterisk2, but it should not drop the call, it must forward to asterisk1, if it is live else, the call must routed directly from asteriskd to agent1

How shoudl i design like this setup, the same way for outbound calls too…

Any help would be greatly apreciated.

Thanks in advance
Rizwan