Default Asterisk configuration (only sip.conf changes).
I use call files for calling and I need to hangup after first ring while every dial.
WaitTime: 4 seconds doesn't work sometimes, since it's counting from the beginning (connect to SIP provider etc) and the client doesn't even receive the call.
00359894000001.call
Channel: SIP/flowroute/00359894000001
Extension: 00359894000001
WaitTime: 4
There are no way predict connection time and count how much ring was at called side. Ring create by endpoing equipment, you have no control/info about it.
If the connection time for the SIP provider etc is constant you should just check what it is and add it to your 4 seconds.
Related
We have one problem we've been suffering of for a long long time,It's the unknown callerID received from asterisk that happens on specific situations.
First we have a sip soft phone (sipml5)
and on server side we have
asterisk-11.25.0-0
elastix-4.0.0-1
Setup: we have any cid/did inbound route connects our calls to one ring-group(that have all the extensions)
The unknown caller id shows when we have:
1-Short timeout for call(which lead to make the call stick at asterisk and asterisk resent the call to extensions but with unknown caller id).. possible solution would be to make big timeout.
2-All extensions do hangup and the call stick on asterisk and asterisk resend it to extensions with unknown caller id (possible solution would be to prevent extensions doing hangup unless they answer the call first)
3-Receiving one unknown caller id lead to successive unknown caller id calls.. no solution
What we're trying to solve is the 3rd problem and we have an idea of forcing asterisk to wait for a specific timeout between inbound calls(we tried this manually by not allowing immediate successive calls,make 4-5 seconds delay between calls and it works fine)
We want to know what configuration has to be edited to force this timeout delay between inbound calls.
You can use EPOCH function and store value somewhere(ASTDB?)
After that in dialplan compare it when dialling
No, elastic not support that and will not support it in future(weird request).
I have a FreePBX 13 server set up with a SIP Trunk connection, however for some reason we are not getting the ring back tone for calls going out of the trunk connection.
I was able to implement a work around for this by placing the "Tr" options under "Asterisk Trunk Dial Options" to force Asterisk to produce the ring back tone for outbound calls.
However this only works while manually dialing from a soft-phone / VoIP Phone, when I try to launch a call via the Asterisk AMI "Originate" command we are not getting the ring back tone, even when the "r" option is set on the trunk. This is how the command I'm sending looks:
Action: Originate,
Channel: SIP/{extension},
Context: from-internal,
Exten: {phoneNumber},
Priority: 1,
Callerid: {callerId},
Timeout: 30000,
Async: yes
Any idea about what can I do to force the ring back tone?
This could be due to inband progress.
Try to add the following to your sip.conf
under [general] section,
prematuremedia=no
progressinband=yes
if you are using freepbx, then navigate to your trunk settings in the user interface and add the above 2 configs under the peer settings.
restart asterisk and try after that.
Launch Originate via Local/ channel call, if still not work, launch via custom context which answer first.
This is not an uncommon problem. Your ITSP is being lazy and either not providing or not forwarding RFC-complaince SIP 180/183 responses.
My usual solution is to put the local caller into an MOH state where the MOH is a ring tone, and then when the other end picks up, bridge the two calls.
This is kind of jarring, however, if the remote end is busy or is congested, since you go from "fake ring" to "real error tone". The other option is to reverse your dialing process ... call the remote end and then hook the local end, so that the local end doesn't need to hear ringing.
Of course, you can also reach out to your ITSP and tell them you need them to be RFC SIP 180/183 complaint.
I have a java stasis application on Asterisk 14 using ari4java. It mostly works great. I am now trying to receive an external call and relay it back out. I do following
Incoming call enters Stasis
Create bridge
Add first call(channel) to bridge
Create channel
Add second channel to bridge
Dial( secondChID, "Local/2601", 30)
No matter what I try, the second outbound call gets the callerID of the first inbound call. That is actually OK for many calls, but in this case I want to set another callerId.
Before Dial() I have tried to setChannelVar(CALLERID(num)) and this value I can see in all events coming from Asterisk. But once the SIP call is placed, no sign of my callerID.
I doubt it is the ari4java doing anything wrong as I see the callerID in all the "dial" events. I thought I could force a callerID in sip.conf, but unable to do that too.
I am looking this use case scenario for Asterisk. I am using v 1.8 running a Centos 6.4 Linux distribution.
1.An outbound call is initiated via Asterisk
2.Both the internal extension as well as the outbound call-phone starts to ring.
The first person to pick up (either the internal extension or outbound call-phone) will
hear the a pre recorded message to hold as the call is being connected to the other user
i.e: If outbound call-phone picks up first then he will be asked to hold the line while the call is being connected to internal-phone user.
Any inputs?
My suggestion is almost the same but a little different:
Setup a dynamic meetme room changing the only-person message to "please hold..."
Setup booth calls at same time to destination numbers and set the originate command application parameter as meetme
Create 2 calls
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
After connect use bridge command or conference room.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
I am using asterisk(1.6.2.13) to mass originate to specified numbers that are come from mysql database using perl and AMI.
if I send all calls (simultaneous) to asterisk, it will drop half of them after about 20 seconds. but if I sleep for 1 second between each originate it will process the call clearly. so this will reduce the capacity of origination.
Is there any way to get rid of this limitation?
Asterisk uses a single thread to process all SIP messages, and everything relating to SIP messaging happens in this thread (eg. realtime database access). This imposes an upper limit on the number of calls that can be processed per second. You can monitor the unprocessed SIP packets on the socket queue using something like "netstat -na |grep 5060". I've found it can up to 200 call setups per second, beyond that it will start losing and retransmitting packets and eventually dropping calls.