I'm running Asterisk 1.13. When two users enter dtmf 9 it gets ignored.
I can see in the logs that Asterisk is receiving the dtmf but it doesn't do any action. Most phones that I have tries work fine. I see the following in the logs, but no action is taken.
[Apr 15 16:40:08] DEBUG[11094][C-00000018] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 57 (9), at x.x.x.x:37740
[Apr 15 16:40:08] DEBUG[11094][C-00000018] res_rtp_asterisk.c: Creating END DTMF Frame: 57 (9), at x.x.x.x:37740
Any ideas?
I had Wait() in the dialplan instead of WaitExten(). It therefore didn't listing at that time for input leading me down many rabbit holes. It's sorted now.
Related
we have created a Group in Vicidial and added few remote agents to the group. Remote agents have mobile phone numbers. Here are some of the options from the group configurations:
Next Agent Call: longest_wait_time
Queue Priority: 0 - Even
On-Hook Ring Time: 15
Now the queue is working as expected. Agent numbers are dialed based on the wait time. But the problem is that for an incoming call, the system dialed only one number from the queue and hangs up if the call is not answered.
It does not dialed the next number from the queue if the previous number does not receive the call. It only dialed the next number if a new call comes in.
I was wondering how to have the next number from the queue dialed (or redirect the user to an IVR) if the previous number does not picks up?
Vicidial:
Version: 2.14b0.5
SVN Version: 3254
DB Schema Version: 1596
Found a way, in my case all the agents are remote agent on PSTN connections. So I have set the On-Hook = Y for all the remote agent accounts. And now if the previous agent is busy or does not receive the call then the next agent is dialed from the queue.
Need to check if setting the On-Hook property to Y has any other ramification.
Update: You have to set the onhook time considering 5/6 seconds will be spent on the PSTN end to search for the mobile device + couple of seconds to initiate the call + ring time on the mobile device. In my case setting 23 seconds working ok for a 15 seconds ring time on the mobile end.
Sometime in my sip accounts occurs network problem and generates "UnReachable" event.
[2020-04-27 16:57:56] NOTICE[2949] chan_sip.c: Peer '323' is now UNREACHABLE! Last qualify: 6
I also see it in log files. I have 2 question about this situation.
After how many second Asterisk generate this event when can not access to sip account?
Exist some parametr change this reachable timeout second?
You can look at:
qualifyfreq=60 = default value is 60 sec
qualifygap=100
this works if:
qualify=yes|4000 = 4000 ms
I was running a asterisk 11.5 with no error. After installed Opus codec with patch in my asterisk
btw i am using browser based sip softphone jssip
Now whenever I am making audio call between 2 peer, no call is getting establish and throwing below errors:
btw i am using browser based sip softphone jssip
[Sep 29 09:27:36] WARNING[17065][C-00000003]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 29 09:27:36] WARNING[17065][C-00000003]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 29 09:27:36] WARNING[17065][C-00000003]: chan_sip.c:11100 process_sdp_a_audio: Got Opus minptime=10
[Sep 29 09:27:36] WARNING[17065][C-00000003]: chan_sip.c:10437 process_sdp: Rejecting secure audio stream without encryption details: audio 17002 RTP/SAVPF 111 103 104 0 8 106 105 13 126
[1000]
secret=test123
context=local
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,wss,ws
encryption=yes
dial=SIP/8000
callerid=testuser <8000>
callcounter=yes
avpf=yes
icesupport=yes
directmedia=no
Can any one cooperate?
Use tcpdump or wireshark to look at the packets, and make sure that they're actually encrypted packets. I looks like it may be rejecting them because it's not SRTP. However, that is only an educated guess. Additionally at the asterisk command line you might want to look at rtp set debug on.
You may look to Asterisk 12 for Opus support, where it will be officially supported, or so it seems from the development activity for Asterisk 12.
Here's the reference to the issue and discussion of the fix. And you can see that it's been approved by the review board here.
I, too, am interested in using Opus. However, without knowing what patch you are using, and the stability of said patch -- it may be difficult (and rather time consuming) to give specific instruction here. I would recommend reporting the issue to the person/group who released the patch as it may help the community project (if it is such).
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.
I'm using FreePBX from a Trixbox install to manage an Asterisk server. I added a dial group with ringall strategy, but as soon as one person answers, the other extensions in the group are dropped. I'd like to keep ringing these extensions so that everybody that picks up the call lands in a conference with the caller.
It would be acceptable to join the conference first, then dial the group.
Create callfiles for each party you wish to add to the conference. Callfiles are text files that are placed in /var/spool/asterisk/outgoing and cause Asterisk to originate a call based on the contents of the callfile. Make sure that the callfile is r/w by the same user that runs Asterisk. Set the mod date in the future to schedule the outbound call.
The format of the callfile should look something like this:
Channel: Local/8085551212#from-internal
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: ext-meetme
Extension: 202
Priority: 1
...where 8085551212 is the party to call and 202 is the conference room extension. When answered, each call will be placed into the conference. Keep in mind that the call might be answered by an IVR or voicemail, so you might want to require a PIN for the conference or drop the incoming calls to an IVR that will then connect to the conference if the intended party is reached. It could be as simple as, "It's time for the sales conference. Dial # now to join."