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Parking, Playback For At Least 500 Parallel Calls And Pickup One

Hi list,

we want to setup an Asterisk on Debian Wheezy. There will arrive at least 500 parallel calls. All of them we set to parking. Then the receive a playback. After a certain time, we will pick up one caller. All the other calls will be cancelled now. Can Asterisk handle this or are there some hardcoded limits?
How much ressources you think will be needed for this solution (CPU, RAM, Disk I/O)?

Best regards, Julian Santer

Process Asterisk Stop

Who can help?
I have Asterisk 1.8.3 server on Ubuntu 10.04. Asterisk periodically falls here with this error:

[1227952.625701] asterisk [30237]: segfault at 18 ip 00007ff3504579bc sp
00007ff34ddc3ff0 error 4 in libc-2.11.1.so [7ff3503e0000 +17 a000]

Dialplan To Reach External SIP Phone

If I have Asterisk setup with local SIP phones configured but need to call a SIP phone which is not local but actually on another VLAN, what would the dialplan need to look like?

Two Pid’s Shown In Asterisk Service Status

Hello List,
Good day,
We have faced an issue in Asterisk, the issue is we are not able to make manager connections and we could see two pid’s for the service asterisk status command. The sequence of operations in our setup goes here,
1) Linux machine is rebooting , we have issued a service asterisk restart command.
2) Tried to connect from an application to asterisk.

The manger connections lost. Did anyone faced such issues or any solutions in latest versions of Asterisk to handle such situation. Best regards, Ruban.S

Siemens Limited, IC BT SSP ES R&D IPI 1
SP Infocity, Block B, 2nd Floor,
#40, MGR Salai, Kandanchavadi, Perungudi, Chennai – 600 096
Tel: +91 44 33524336
Mobile: +91 9500084833

Debugging “stuck” Inbound Call

Asterisk 11.1.0 running on Ubuntu 12.04 64 bit Dahdi Digium T1 card

Occasionally, I will find an inbound call that just seems to be stuck, usually in our after-hours menu portion of the dial plan.

This morning I had this one

core show channels concise DAHDI/i1/5184097869-1baf2!MainMenu!s!20!Up!BackGround!custom/aa-night-hello&custom/hours_8:0-17:0_0:0-0:0_0:0-0:0&custom/aa-night-instructions!5184097869!!!3!9393!(None)!sip1-1396004671.285644

which had been there for about 2.5 hours (time from core show channels verbose).

The inbound channel here is to our toll free number on the T1.

When we’ve researched these in the past, we’ve not found a correspondingly long call on the phone bill, leading me to wonder if the call is actually being disconnected, but Asterisk just doesn’t find out.

How can I go about debugging this? Are the dahdi commands that can show me the connection status from the hardware perspective?

AMD With Analog Lines – DIALSTATUS Empty


I would like to use AMD on outgoing calls using analog line. I tested with SPA3102 and cisco2811 as gw and asterisk as well as 11.8.1
Other end is analog number behind another cisco/asterisk, also tested calling a mobile number with the same result.

What I did: dial is done like exten => s,n,Dial(SIP//,,M(myMacro)), which tell Asterisk to execute myMacro when the call is answered by calling party.


exten => s,1,NoOP(Executed when call is answered)
same => n,AMD()
same => n,NoOp(Dial status=${DIALSTATUS})
same => n,NoOp(AMD status=${AMDSTATUS} cause=${AMDCAUSE})
same => n,MacroExit()

Problem is that [myMacro] is executed as soon as the call is going out from the gw (cisco or linksys) and before called party answered. DIALSTATUS is empty (should be ANSWER), AMDSTATUS=NOTSURE and AMDCAUSE=TOOLONG-5000 which seems OK as DIALSTATUS isn’t reliable.

The same dialplan using a SIP trunk is working as expected.

So question is, why, when using analog line, I dont get the right behavior.

Thanks for any hint