I am trying to set add a SIP Header to a call before adding it to the Queue.

The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs (verbose 999). In the SIP case, I see it.

Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the call when the Queue…

Asterisk Users 1 months ago 4 Answers

Queues Don't Follow Dialplan If No Members Are Registered


I am running Asterisk 11 on CentOS 6.x. I have configured several queues as follows in extensions.conf:

exten => s,1,Queue(myqueue,rtnC,18)
same => n,Background(user_unavail)
same => n,WaitExten(10)
exten => 1,1,Voicemail(1111@my-vm,s)

This rings the phones in the queue for 18 seconds. If no queue members answer, the caller is then prompted to press 1 and leave a voicemail. This works well when at least 1 member is registered in the queue, however if no members are registered in the queue, the Queue() call never seems to return, and thus the remaining steps in the dialplan…

Asterisk Users 2 months ago 2 Answers

Phones Don't Stop Ringing When Queue Is Answered



I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows:

sip.conf: [169] secret1111 dtmfmode=rfc2833 directmedia=no directrtpsetup=yes canreinvite=no context=main host=dynamic type=friend portP60 call-limit=5 nat=force_rport,comedia callcounter=yes

queues.conf: [queue_level_1] musiconhold

Asterisk Users 5 months ago 3 Answers

Help Debugging A Possible SIP Channel Leak In 11.17.0, Possible Race Condition


I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which…

Asterisk Users 6 months ago 4 Answers

Determining If A Queue Member Is Paused In Dialplan Logic. [1.8]



I'm looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing.

The problem I see is that when the queue pauses an Member it doesn't jump into the dialplan to do so which means my handy device state and asterisk database driven Light for the Member showing their paused status won't update.

My idea for solving this problem is to check the status of my Member in the queue before I send the calls into it and toggle on the Members Paused light at that point in time if they are…

Asterisk Users 6 months ago 3 Answers

Ringtone To A Member Queue


Hello everyone, I configured a queue with dynamic agents for a small proyect, normally only 1 member in the queue for ansering calls would be enough. But I need to configure a ringtone or any other signal to comunicate to the member's queue other call cames into the queue and the member could decide if park the first one o keep the second one in the queue. I appreciate any help from the group. Regards, Diego Huesca

Asterisk Users 7 months ago 0 Answers

Question Regarding Custom Announcements Used By Several Asterisk Servers



Got a question regarding custom announcements in Asterisk.

My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and…

Asterisk Users 8 months ago 4 Answers

Queue Show Vs Queue Log For Calculating Average Hold Time



We're using on CentOS 5 and are trying to get accurate stats for queues.

For a particular customer, when I run queue show I get the following numbers:

has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s

So from that data we look at 17s holdtime And assume that is the average hold time before calls get answered by a queue members.

However, if I calculate the average hold time from out queue log table using the following SQL

select sum(data1)/ count(*) as ave_hold_time from queue_log where time > DATE(NOW()) and queuename='' and…

Asterisk Users 8 months ago 2 Answers

Queue Reload Command



I'm using asterisk 1.8

Does anyone know how to use the queue reload command. The built in help doesn't really help.

queue reload {parameters|membe Reload queues, members, queue rules, or parameters



Asterisk Users 9 months ago 2 Answers

Queues And RingInUse



I have some problems with a Queue and state interface.

In the queue we have 7 deskphones and 7 cordless. Every user have one deskphone and one cordless phone.

I have set up every user with hints, eg:

exten => 802,hint,SIP/9144-802&SIP/9144-902 exten => 902,hint,SIP/9144-902&SIP/9144-802

Queue_members: Queue_name Interface state_interface 9144-vmi SIP/9144-802 hint:802@hints 9144-vmi SIP/9144-902 hint:902@hints

802 = is a cordless phone 902 = is a deskphone Both are in use by same user.

To the problem, when we have RingInUse = 0, we only get calls on one of the devices (because the other one gets in RINGING state). If the user is not on phone we…

Asterisk Users 2 years ago 0 Answers