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Problem with Extension

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Hi , i have a IVR that ask "mark 1 to support" "mark 2 to sales" .. etc

This works , but sometimes that option is not working , so if somebody mark 2 , nothing happend , this is not all the time , but sometimes the extension is not working and the cli dosent show nothing wrong . This is my dialplan .

[dialplan]

include => from-pstn

exten => s,1,Answer
exten => s,n,Set(CHANNEL(hangup_handler_wipe)=hangup,s,1(args));
exten => s,n,Set(CHANNEL(musicclass)=music)
exten => s,n,Background(music1) ; (This say that the call must be recorded)
exten => s,n,Background(music2) ; (This say "mark 1 to support" "mark 2 to sales" )
exten => s,n,Background(music2) ; (This repeat)

exten => 1,1,Goto(ventas,s,1)

exten => 2,1,Goto(soporte,s,1)


If somebody can help me
Thank You Very Much
http://www.melhorempresadehospedagem.com

Statistics : Posted by roni777 • on Tue Jul 14, 2015 3:10 pm • Replies 0 • Views 54

sending DTMF to ongoing call from website

sending DTMF to ongoing call from website

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david55 wrote:AMI SendDTMF.

Thank you for response.
How the control will come back to website once the call is originated ?
and how to send second and subsequent requests to the same channel ?

Statistics : Posted by vkalpanar • on Thu Jul 31, 2014 11:28 am • Replies 7 • Views 879

sending DTMF to ongoing call from website

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AMI SendDTMF.

Statistics : Posted by vkalpanar • on Thu Jul 31, 2014 11:28 am • Replies 7 • Views 879

sending DTMF to ongoing call from website

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I want to send DTMF input to ongoing call through php website, how it will be achieved

Trigger call by clicking button on website
Wait for connection
Press the keys in order to send the DTMF tones
Do some processing
Do not hang up
User will trigger a button from website again to send DTMF to the same call

Statistics : Posted by vkalpanar • on Thu Jul 31, 2014 11:28 am • Replies 7 • Views 879

Asterisk AMI and DTMF

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DTMF sent from AMI will go out on the channel you specify, not be seen as incoming. If there is no alternative to using DTMF, you must interpose a Local channel and send the DTMF on the ;1 side of that.

If you do it properly, the caller won't hear the tones.

Statistics : Posted by fuzzylogic78 • on Wed Jul 15, 2015 6:13 am • Replies 1 • Views 92

Asterisk AMI and DTMF

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I have an application where I am using AMI originate to start a call.

This bridges a call between an extension (1001) and my dialplan which waits for DTMF tones from the extension. If the user presses 8, then the dialplan should play another file, if the user presses 8 again it plays another file and so on and so on.

This works fine when digit 8 is presses from a SIP phone, but when trying to use the AMI action: playDTMF it doesn't work.

the tone is hear from the phone and I can see the activity in the asterisk CLI.
To be able to see it in the asterisk logs I first have to create a local channel to send the DTMF too.
While its being displayed in the asterisk CLI it doesn't step over to play the next file.

Here is an example of my asterisk CLI with key presses being made from a phone:
[2015-07-15 11:30:59.831] DTMF[14264][C-00000019]: channel.c:3945 __ast_read: DTMF end '8' received on SIP/1001-00000018, duration 180 ms
[2015-07-15 11:30:59.832] DTMF[14264][C-00000019]: channel.c:3986 __ast_read: DTMF end accepted with begin '8' on SIP/1001-00000018
[2015-07-15 11:30:59.832] DTMF[14264][C-00000019]: channel.c:4015 __ast_read: DTMF end passthrough '8' on SIP/1001-00000018
[2015-07-15 11:30:59.832] DTMF[14267][C-00000019]: channel.c:3945 __ast_read: DTMF end '8' received on Local/112@extensions-00000009;2, duration 180 ms
[2015-07-15 11:30:59.832] DTMF[14267][C-00000019]: channel.c:4015 __ast_read: DTMF end passthrough '8' on Local/112@extensions-00000009;2
-- Executing [112@extensions:80] NoOp("Local/112@extensions-00000009;2", "USERSTOPPED") in new stack
-- Executing [112@extensions:81] NoOp("Local/112@extensions-00000009;2", "60") in new stack
-- Executing [112@extensions:82] NoOp("Local/112@extensions-00000009;2", "8") in new stack
-- Executing [112@extensions:83] Set("Local/112@extensions-00000009;2", "VAR=USERSTOPPED") in new stack
-- Executing [112@extensions:84] Set("Local/112@extensions-00000009;2", "FRE=9") in new stack
-- Executing [112@extensions:85] GotoIf("Local/112@extensions-00000009;2", "1?heard1:notheard9") in new stack
-- Goto (extensions,112,130)

Compared to the same activity when the DTMF tone is generated by AMI:

[2015-07-15 11:30:59.693] DTMF[14264][C-00000019]: channel.c:4031 __ast_read: DTMF begin '8' received on SIP/1001-00000018
[2015-07-15 11:30:59.693] DTMF[14264][C-00000019]: channel.c:4042 __ast_read: DTMF begin passthrough '8' on SIP/1001-00000018
[2015-07-15 11:30:59.693] DTMF[14267][C-00000019]: channel.c:4031 __ast_read: DTMF begin '8' received on Local/112@extensions-00000009;2
[2015-07-15 11:30:59.693] DTMF[14267][C-00000019]: channel.c:4035 __ast_read: DTMF begin ignored '8' on Local/112@extensions-00000009;2
-- Executing [112@extensions:71] NoOp("Local/112@extensions-00000009;2", "SUCCESS") in new stack
-- Executing [112@extensions:72] NoOp("Local/112@extensions-00000009;2", "-1") in new stack
-- Executing [112@extensions:73] NoOp("Local/112@extensions-00000009;2", "") in new stack
-- Executing [112@extensions:74] Set("Local/112@extensions-00000009;2", "VAR=SUCCESS") in new stack
-- Executing [112@extensions:75] Set("Local/112@extensions-00000009;2", "FRE=8") in new stack
-- Executing [112@extensions:76] GotoIf("Local/112@extensions-00000009;2", "0?heard1:notheard8") in new stack
-- Goto (extensions,112,77)


This is my a snipet of my dialplan:
same =>n,NoOp(Play audioFile)
same =>n,ControlPlayback(dir/filename,4000,*,#,8,0)
same =>n,NoOp(${CPLAYBACKSTATUS})
same =>n,NoOp(${CPLAYBACKOFFSET})
same =>n,NoOP(${CPLAYBACKSTOPKEY})
same =>n,SET(VAR=${CPLAYBACKSTATUS})
same =>n,SET(FRE=1)
same =>n,GotoIf($["${VAR}" = "USERSTOPPED"]?heard1:notheard1)

My AMI command is:
action:playDTMF
channel:Local/112@extensions-00000009;2
digit:8

I'm assuming the DTMF is going to the wrong leg of the call and asterisk isn't passing it to ControlPlayback for recognition.

Could anyone advise how I might be able to get same =>n,ControlPlayback(dir/filename,4000,*,#,8,0) to recognise the DTMF tone sent via AMI?

Statistics : Posted by fuzzylogic78 • on Wed Jul 15, 2015 6:13 am • Replies 1 • Views 92

Asterisk 13.4.0 "Cannot determine best translation path..."

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Hello all. Among other things I do on my home Asterisk PBX, one of the I features I am using is forwarding calls for a cell number that I keep "parked" to another cell number that I am using every day using chan_dongle.
Everything was working great up and including to Asterisk 11. After my upgrade to 13, I have been encountering this issue: when I try to forward the call using the Local channel (for which I have some routing/failover in place), I get an error like this:

-- Executing [+16475551234@cellinbound:4] Dial("Dongle/dongle0-0100000006", "local/4165551234@internal,60,wWxXT") in new stack
-- Called local/4165551234@internal
[2015-07-15 13:02:22] ERROR[13481][C-00000007]: translate.c:1284 ast_translator_best_choice: Cannot determine best translation path since one capability supports no formats
[2015-07-15 13:02:22] WARNING[13481][C-00000007]: channel.c:6341 ast_channel_make_compatible_helper: No path to translate from Local/4165551234@internal-00000003;1 to Dongle/dongle0-0100000006

If I use an outbound peer directly rather than the Local channel, everything works OK but I lose the outbound peer failover.
Any idea ?

Statistics : Posted by xoy74 • on Wed Jul 15, 2015 11:10 am • Replies 0 • Views 51

Login failure: Secure Connection Failed on this site

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Howdy,

IT folks are looking into it.

Cheers

Statistics : Posted by gossamer • on Wed Jul 15, 2015 12:34 pm • Replies 1 • Views 69

Login failure: Secure Connection Failed on this site

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Hi,
When trying to login from my Linux desktop using either Chrome or Firefox, I receive the following error:

Secure Connection Failed
An error occurred during a connection to login.digium.com. SSSL received a weak ephermeral Diffie-Hellman key in Server Key Exchange handshake message. (Error code: ssl_error_weak_server_ephemeral_dh_key)

What is the source of this error? Is it my security settings?

Any help greatly appreciated. I'm only able to login using this Windows desktop.

Thanks,
Alex

Statistics : Posted by gossamer • on Wed Jul 15, 2015 12:34 pm • Replies 1 • Views 69

res_ari_mailboxes.so: undefined symbol

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Hi, I'm using the asterisk-13.1.1 RPM included with fedora22 with dahdi downloaded and compiled from git and receiving the following error:

Code: [Jul 15 11:25:51] WARNING[31751] loader.c: Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json
[Jul 15 11:25:51] WARNING[31751] loader.c: Module 'res_ari_mailboxes.so' could not be loaded.


This is upgraded from fedora21 where it all worked properly. Is this a fedora RPM problem or an asterisk problem? I'd rather not have to compile that from source too.

I also have a number of other errors/warnings, and I don't know if they're related, but I'm really stuck. These are from the messsages file when starting asterisk. Where do I go about troubleshooting these problems?

The actual problem I'm having is that asterisk answers the calls but cannot communicate with any IP phones.

Code: [Jul 15 11:25:50] Asterisk 13.1.1 built by mockbuild @ buildhw-06.phx2.fedoraproject.org on a x86_64 running Linux on 2015-01-30 20:30:35 UTC
[Jul 15 11:25:51] NOTICE[31751] cdr.c: CDR simple logging enabled.
[Jul 15 11:25:51] NOTICE[31751] loader.c: 220 modules will be loaded.
[Jul 15 11:25:51] WARNING[31751] res_phoneprov.c: Unable to find a valid server address or name.
[Jul 15 11:25:51] WARNING[31751] res_crypto.c: Unable to open key file /usr/share/asterisk/keys/voicepulse20060419.pub: Permission denied
[Jul 15 11:25:51] WARNING[31751] res_crypto.c: Unable to open key file /usr/share/asterisk/keys/voicepulse01.pub: Permission denied
[Jul 15 11:25:51] ERROR[31751] ari/config.c: No configured users for ARI
[Jul 15 11:25:51] WARNING[31751] loader.c: Error loading module 'res_ari_mailboxes.so': /usr/lib64/asterisk/modules/res_ari_mailboxes.so: undefined symbol: stasis_app_mailbox_to_json
[Jul 15 11:25:51] WARNING[31751] loader.c: Module 'res_ari_mailboxes.so' could not be loaded.
[Jul 15 11:25:51] WARNING[31751] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39.
[Jul 15 11:25:51] WARNING[31751] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47.
[Jul 15 11:25:52] NOTICE[31751] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Jul 15 11:25:52] NOTICE[31751] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Jul 15 11:25:52] WARNING[31751] pbx.c: Unable to register extension '_X' priority 1 in 'mainmenu-night', already in use
[Jul 15 11:25:52] WARNING[31751] pbx_config.c: Unable to register extension at line 220 of extensions.conf
[Jul 15 11:25:52] WARNING[31751] pbx.c: Unable to register extension 's' priority 2 in 'incoming-sales', already in use
[Jul 15 11:25:52] WARNING[31751] pbx_config.c: Unable to register extension at line 292 of extensions.conf
[Jul 15 11:25:52] ERROR[31751] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
[Jul 15 11:25:52] WARNING[31751] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages


Statistics : Posted by gossamer • on Wed Jul 15, 2015 2:41 pm • Replies 0 • Views 46

Asterisk stops processing calls under heavy load

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@david55, I've opened a ticket for this already with Digium's IT folks.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 11 • Views 781

Asterisk stops processing calls under heavy load

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At the moment, the forum server's encryption is unsafe as far as Debian Iceweasel is concerned, which means that it is not easy for me to access it from home, where I have some time. Consequently you are going to have to do more of the leg work and actually work out the cycle of locks, or the locks that seem to be being held for too long, and the threads that seem to be running when you would expect them to be waiting.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 11 • Views 781

Asterisk stops processing calls under heavy load

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Oh, and I removed CDR logging via odbc/mysql. It now logs directly to filesystem with csv files.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 11 • Views 781

Asterisk stops processing calls under heavy load

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Hi, we had the same issue occur almost immediately. Have not had much time to come back here because we had chaos with the system going down during busy hours (100 concurrent calls). We had to move them to their old Asterisk server which does not suffer from this issue.

Things we have tried:

Installed Asterisk 11 on brand new hardware (HP Server, Dual Xeon [32 cores total], 96GB RAM, RAID 10 10k Hard Drives). I know, overkill. This was all we had available at the time. Same issue occurs with this new server. I can successfully reproduce the issue with sipp on a simple dialplan:

Code: [from-sip-test]
exten => 696969,1,Answer
exten => 696969,n,Musiconhold(default)
exten => 696969,4,Hangup


After about 30 calls, (10 calls per second), sipp starts receivinv transmit errors (no response from host).

command:
Quote:netstat -anNp | grep 5060


Shows thousands of packets queued.

Asterisk only shows about 10 calls.
See this in the error log:

Code: res_musiconhold.c: Failed to write frame to 'SIP/sip-test-00000024': Resource temporarily unavailable


Htop shows over 80GB of RAM free, CPU spikes to almost 95% during this time (all 32 cores).

Backtrace:
http://pastebin.com/qY6X5mqm

Core show locks:
http://pastebin.com/cUdFm5mB

I tried different Asterisk versions to 11.12, 11.16, same results. This was running on 11.18. Old server which does not suffer from this issue (as far as I can tell) is running 11.12 x64 on debian. This test was done on Ubuntu 14, 64 bit.

Not sure what else to try.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 11 • Views 781

Asterisk stops processing calls under heavy load

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I have now moved the mysql database and put it on it's own dedicated server. Let's see how it functions today.

Thanks

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 11 • Views 781

Asterisk stops processing calls under heavy load

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I agree, it looks as though something is up with the underlying database connection.

This is personally why I don't like tightly coupling Asterisk to databases. It's handy at times, but if your database acts up then Asterisk reflects that.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 11 • Views 781

Asterisk stops processing calls under heavy load

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You are looking for circles of two or more locks where threads are holding one and waiting on the other, or, in this case, signs that a lock is being held indefinitely for other reasons, namely the odbc lock at 0xab3e488 in this case.

It looks to me as though your ODBC connection for call logging has stalled and everything is waiting for it.

That's likely to be a third party problem and is one of the reasons why I would want to keep call logging as simple as possible.

Statistics : Posted by illizit • on Thu Jul 02, 2015 8:56 pm • Replies 11 • Views 781

Asterisk stops processing calls under heavy load

Asterisk stops processing calls under heavy load

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