Multiple server register with only one sip account(outbound)

Hi guys.

we serve simple outbound service with 2 Asterisk servers.

I have problem about the below :
I register 2 servers(Called “A”, “B”) with one sip account(Called “account123”) to SIP provider.

A(account123) --------- SIP Provider
B(account123) ---------------|

With “sip show registry”, I can see that 2 servers(with one sip account) success to register to SIP Provider.
So I tried to make call around 1,000. but only B server can handle the request successfully.
When I tried to make call to each server separately, both works fine . So I don’t think A server is not a problem. :smiley:

So I was thinking of problem is using only one sip account with 2 server and then I do googling.
and I found something.

You can register multiple end users with only one sip account but asterisk does not support ringing all the registered phones on single account.
Whenever a new registration comes, asterisk updates its contact info in memory. So if the registration is coming from multiple end users (multiple ip address and port) then the call will be placed to the phone who sent latest registration request. Asterisk does not keep track of all the ip addresses for single account registration.

The above is about inbound. but I think… It can be a problem about outbound also.

our system is like the below
|<---------- our system --------------------->|
Java(engine) <–> PHP(AGI) <–> Asterisk <----------------------------> SIP Provider.

“A” server’s log (asteirsk -r) told me the request has been handled by PHP(AGI).
please see the below log.


[Dec 9 16:40:26] VERBOSE[1269] logger.c: sm_dial.php: file:sm_dial.php - line:60 - GET VARIABLE : CALLEDP=1 | CALLED=6514544894 | CALLERID=6514501201 | CBID=1-705167913 | INTRO=t/traveltags/134280958899_Trumpia_Power_Outage_Voice_Message_Intro_Travel_Tags | MSG=t/traveltags/135134418599_Trumpiaseverewinterweathervoice | VOICE=1 | MODE=live
[Dec 9 16:40:26] VERBOSE[1269] logger.c: sm_dial.php: file:sm_dial.php - line:66 - Destination : 16514544894
[Dec 9 16:40:30] VERBOSE[1275] logger.c: == Parsing ‘/etc/asterisk/manager.conf’: [Dec 9 16:40:30] VERBOSE[1275] logger.c: Found
[Dec 9 16:40:30] VERBOSE[1275] logger.c: == Manager ‘trumpia’ logged on from 127.0.0.1
[Dec 9 16:40:30] VERBOSE[1275] logger.c: == Manager ‘trumpia’ logged off from 127.0.0.1
[color=#FF0000][Dec 9 16:40:30] VERBOSE[1277] logger.c: – Executing [s@trumpia-sendmessage:1] AGI(“Local/s@trumpia-sendmessage-8cc7,2”, “sm_dial.php”) in new stack
[Dec 9 16:40:30] VERBOSE[1277] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/sm_dial.php[/color]
[Dec 9 16:40:30] VERBOSE[1277] logger.c: sm_dial.php: file:class.trumpiacallcenter.php - line:120 - callerid : 6514501201
[Dec 9 16:40:30] VERBOSE[1277] logger.c: sm_dial.php: file:class.trumpiacallcenter.php - line:121 - channel : Local/s@trumpia-sendmessage-8cc7,2
[Dec 9 16:40:30] VERBOSE[1277] logger.c: sm_dial.php: file:class.trumpiacallcenter.php - line:122 - uniqueid : 1355100030.122
[Dec 9 [color=#FF0000]16:40:30[/color]] VERBOSE[1277] logger.c: sm_dial.php: file:class.trumpiacallcenter.php - line:123 - dnid : s

[color=#FF0000]…(nothing to do for 30 secs)
…(It should be logged like “Playing Ment blah blah blah”)[/color]

[Dec 9 [color=#FF0000]16:41:00[/color]] DEBUG[1277] res_agi.c: Local/s@trumpia-sendmessage-8cc7,2 hungup

So If I use 2 sip account for each 2 asterisk server, the problem is solved?
(I must pay for one more sip account, so I didn’t test yet.)

[color=#FF0000]Is my guess(Problem is using only one sip account) right? or What should I do for that? [/color]

Thanks for thaking your time.

That will be because your ITSP’s server has the same restriction as Asterisk (and may even be Asterisk).

If an ITSP is getting significant revenue as a monthly charge on an account, they would probably consider that desirable behaviour, as it prevents reselling the account for outgoing only calls.

[quote=“david55”]That will be because your ITSP’s server has the same restriction as Asterisk (and may even be Asterisk).

If an ITSP is getting significant revenue as a monthly charge on an account, they would probably consider that desirable behaviour, as it prevents reselling the account for outgoing only calls.[/quote]

thanks for reply. david55
I’m gonna ask to our sip provider about this. :smiley:
Have a nice time.

One other thing to note is that ITSPs that accept multiple registrations for the same address of record will typically offer incoming calls to all of them at the same time. That might not be what you want.

I asked about this to our SIP provider.
They said there is no restriction for us.

Any other idea why this is happened? :smiley:

Hello all.

I have outbound call problem.
I have 2 asterisk server(A, B), 2 sip account like the below(I got one more sip account)
(voice engine is for requesting voice call to asterisk server)

voice1 engine -------- L4 ------------- A server(sip account : account1) ----------- SIP provider
voice2 engine -------| |----------- B server(sip account : account2) ----------|

I test case by case to get what is problem.

  1. voice1, voice2 -> L4 (FAILED)
  • I doubt one sip account with multiple server.(please read the above to understand what situation)
  • so I can conclude one sip account is not a problem.
    - failed only on A server.
  1. voie1 -> A, voice2 -> B (FAILED) / voice1 -> B, voice2 -> A (FAILED)
  • I think L4 is not a problem
    - failed only on A server.
  1. voice1, voice2 -> A(SUCCESS) / voice1, voice2 -> B (SUCCESS)
  • I thnks Asterisk server is not a problem.

So, I talked with server engineer about this.
and we start to doubt NFS for voice file sharing.

A real voice file is in A server. B server use it through NFS.

([real voice file] A server) <------- NFS ---------> (B server)

When A, B servers try to use one voice file concurrently, Is this will be caused problem?