Changing Volume Via Dialplan

Report
Question

Greetings everyone, I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried setting x and y to -30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10, and 100 and there appears to be no change on the phone volume. I can see that the Set(VOLUME) instruction is being executed on the Asterisk CLI. I have also tried using Set(CHANNEL(txgain)=x) and Set(CHANNEL(rxgain)=y) and those don't seem to have any effect either. I…

Asterisk Users 14 days ago 2 Answers

SRV Lookups In Asterisk 11

Report
Question

Hello,

Can anyone advise on the status of SRV lookups in Asterisk 11? (specifically 11.17.1)

Is there any difference given how the Dial is done, and how supported are weights and priorities?

Thanks in advance,

Asterisk Users 15 days ago 1 Answer

Asterisk 13 Chan_sip Trunk Appending @string To Dialled Number

Report
Question

Hello,

I'm having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I try dialling out via this trunk, something appends '@CUBE' onto the end of the dialled number, as per the following examples;

Asterisk log; app_dial.c: Called SIP/test/0429123456@CUBE chan_sip.c: Got SIP response 500 "Internal Server Error" back from 172.22.4.12:5060

In the SIP SDP; INVITE sip:0429920437%40CUBE@172.22.4.12 SIP/2.0. To: .

As you can see, the @CUBE carries over into the SIP URI as %40CUBE. The FPBX trunk name and outbound route were called CUBE (afaik, purely descriptive) but I changed them to something…

Asterisk Users 17 days ago 12 Answers

786 000 Files Limit CentOS 7 - Asterisk

Report
Question

Hi Markus

Thanks for the reply, I have set those files previously as well...

It seems the problem for me is on my CentOS 7 box that the Asterisk binary does not "know" that these are the limits, and imposes a 1024 open file limit count for some reason.

It seems using prlimit to force the running binary to use a higher file limit is working - but it has to be done manually each time the asterisk process is started, the prlimit cronjob I set up to force the file limit higher does not seem to work - you have to physically…

Asterisk Users 23 days ago 2 Answers

786 000 Files Limit CentOS 7 - Asterisk Keep Complaining

Report
Question

Hi Tony

Thanks for replying.

I suspected something like that, though repeatedly running

lsof | wc -l

Always stays quite low - 100 000 open files, which is still 8 times less than the system maximum as confirmed by running ulimit -n

I also note that this number will increase to about 125 000 but never go higher than that, then, as calls hang up, decreate again - during times when the CLI is spammed with 100s of "broken pipe" errors due to insuffiecient file descriptors, this number never reaches beyond 125 000 out of the available 800 000 open files.

If I grep by…

Asterisk Users 24 days ago 5 Answers

Asterisk RealTime Sippeers, Rtcachefriends=yes, Phones Lose Registration

Report
Question

Hello, we have an issue where after a couple of days, a few random phones will lose registration. I don't notice any particular pattern. Out of 200, only about 5-10 will be suffering at any given time and we won't know until the user complains they are not receiving calls. "sip show peers" does not show the phone in the list. I see packets coming into the server, but asterisk is not responding. Rebooting the phone to signal another REGISTER, yields no results. It seems Asterisk simply ignores the packets. I don't see anything in the logs that are relevant…

Asterisk Users 24 days ago 0 Answers

Load-balancing AMI And Load-balancing FastAGI?

Report
Question

Hi,

I am starting a new project to develop a predictive dialler system.

- Agents can start receiving calls from the queue if agent press "Available" button on the browser which will unpause the queue on Asterisk.

- About 100-150 concurrents calls on a Asterisk box

- Call-out initiated. Other end answers. Passes AMD. Lands in Queue and direct to agents that is available and call is recorded.

- Update state of the call (Ringing, Talking, etc) on the database.

- Listen the events such as Hang Up from customer, check if call is successfully originated or what the failure, etc.

- Agent will have ability to…

Asterisk Users 25 days ago 1 Answer