[asterisk-users] Asterisk 13.11 realtime problem registering phones
I upgraded my office installation from 13.10 to 13.11 yesterday and
now I am having problems registering phones. Here is what I get on the CLI:
[Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql:
Realtime table general@ps_contacts: column 'qualify_timeout' cannot be
type 'int(10)' (need char)
[Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql:
Realtime table general@ps_contacts: column 'expiration_time' cannot be
type 'bigint(20)' (need char)
[Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1246 require_mysql:
Possibly unsupported column type 'enum('yes','no')' on column
'authenticate_qualify'
[Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql:
Realtime table general@ps_contacts: column 'via_port' cannot be type
'int(11)' (need char)
[Sep 2 15:38:46] ERROR[2098]: res_pjsip_registrar.c:411
register_aor_core: Unable to bind contact
'sip:2001@192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525'
to AOR '2001'
== Contact
2001/sip:2001@192.168.2.203:57776;transport=UDP;rinstance=48d5c7d09b9f2525
has been deleted
The mysql warnings have always been there since version 13.0 and
the "Unable to bind contact..." error has also been present since I
started using PJSIP realtime with Asterisk 13 (13.5 at least). But
starting this version the contact gets deleted immediately after
registration so I cannot receive calls on the phone. I can make calls
from the phone. On the phone web gui it tells me there is a "contact
mismatch". There are some phones working but I am not able to determine
why, no configurations were changed since the upgrade (either on the
database or on the phones themselves). I have two phones of the same
make/model/firmware, one is working and the other one is not and they
basically have the same configuration, only the username and password
change.
I guess realtime finally broke.
--
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Carlos Chávez
+52 (55)9116-91161
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Re: [asterisk-users] Feature Request: what about "core stop panic" ?
On Fri, Sep 2, 2016 at 9:34 AM, Olivier wrote: > Hello, > > I had a recent case where Asterisk stopped due to a segfault. > This reminded me that being sure that whenever such issue occurs, it's > useful to have a core file or various data at hand to analyze and exchange > with support teams. > > How can you double check a running Asterisk system would produce such data > if a segfault arrises ? > > 1. To my knowledge, no tool exists for this. Is it correct ? > 2. If such tool do not exist, what would you say about something like > "core stop panic", "core stop segfault" or whatever ? > How about just running "pkill -SEGV -f /usr/sbin/asterisk" from the shell? Works for me. > > Best Regards > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feature Request: what about "core stop panic" ?
Hello, I had a recent case where Asterisk stopped due to a segfault. This reminded me that being sure that whenever such issue occurs, it's useful to have a core file or various data at hand to analyze and exchange with support teams. How can you double check a running Asterisk system would produce such data if a segfault arrises ? 1. To my knowledge, no tool exists for this. Is it correct ? 2. If such tool do not exist, what would you say about something like "core stop panic", "core stop segfault" or whatever ? Best Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail greeting
> hi.i managed to record my voicemail greeting. the only problem is that after > my greeting the caller hear '...please leave your message after the tone. > when done press the pound key or hangup.' is there a way to get rid of that? > Ideally i would like to have my own recording and then the beep sound. > Try option s : https://wiki.asterisk.org/wiki/display/AST/Application_VoiceMail Regards, -- Bertrand LUPART -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail greeting
hi.i managed to record my voicemail greeting. the only problem is that after my greeting the caller hear '...please leave your message after the tone. when done press the pound key or hangup.' is there a way to get rid of that? Ideally i would like to have my own recording and then the beep sound. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily
Le 02/09/2016 à 11:26, Jonas Kellens a écrit : Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal' (thanks to SIP/myaccount184-3729) Question : how can I read the variable which contains the value 'myaccount184' in the context from-internal ? From SIP_HEADER(TO) ? [...] -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to disable subsequent transfers?
Hi,
Consider the following scenario. A customer's incoming call enters the
system, and after some processing, the call is placed on a queue, where it
will be picked up by an agent.
Then, the agent makes an attended transfer (using asterisk internal
transfer) of this costumer to some other party.
When the transfer is made, none of the endpoints should be able to make
subsequent transfers.
On the Queue statement where the incoming call is placed, the 't' flag is
used to enable the agent to make the transfer.
On the Dial statement used in the attended transfer, neither the 't' nor
the 'T' flags are used.
But, after the transfer is completed, the party that the customer was
transfered to is still able to tranfer.
I'll show some logs from my testing environment:
asterisk1*CLI> features show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# B
Attended Transfer 1
One Touch Monitor
Disconnect Call * *0
Park Call
One Touch MixMonitor
(etc...)
Incoming call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [333@from-ctaadmon:1] NoOp("SIP/ctaadmon-0054", "Entra
el num 11139 con name Sala de formacion 3 exten=333") in new stack
(... some processing, then)
-- Executing [333@from-ctaadmon:26] Queue("SIP/ctaadmon-0054",
"cola_agentes,twWrrestartmixmonitor.py,enviar-canal-por-sendtext") in
new stack
Notice that parameter t is used on the queue application. The agent should
be able to transfer the call.
Then the agent picks up the call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/1001
-- SIP/1001-0055 connected line has changed. Saving it until answer
for SIP/ctaadmon-0054
-- SIP/1001-0055 is ringing
-- SIP/1001-0055 connected line has changed. Saving it until answer
for SIP/ctaadmon-0054
-- SIP/1001-0055 answered SIP/ctaadmon-0054
(etc...)
SIP/ctaadmon-0054 -> customer channel.
SIP/1001-0055 -> agent channel.
Now that the agent is talking to the customer, the two channels are bridged:
asterisk1*CLI> core show channels concise
SIP/1001-0055!coordinador!333!1!Up!AppQueue!(Outgoing
Line)!1001!!!3!12!SIP/ctaadmon-0054!1472723633.139
SIP/ctaadmon-0054!from-ctaadmon!333!26!Up!Queue!cola_agentes,twWrrestartmixmonitor.py,enviar-canal-por-sendtext!11139!!!3!12!SIP/1001-0055!1472723633.138
Now, the agent makes an attended transfer to 646xx (a 9 digit number,
not shown). To do so, first he sends the '1' key (dtmf),
which is assigned to the attended transfer, then the number which he wants
to transfer the customer to:
[Sep 1 11:54:22] DTMF[18353]: channel.c:4109 __ast_read: DTMF end '1'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:22] DTMF[18353]: channel.c:4135 __ast_read: DTMF begin
emulation of '1' with duration 250 queued on SIP/1001-0055
[Sep 1 11:54:22] DTMF[18353]: channel.c:4271 __ast_read: DTMF end
emulation of '1' queued on SIP/1001-0055
-- Started music on hold, class 'default', on SIP/ctaadmon-0054
-- Playing 'pbx-transfer.gsm' (language 'en')
[Sep 1 11:54:23] DTMF[18353]: channel.c:4109 __ast_read: DTMF end '6'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:23] DTMF[18353]: channel.c:4178 __ast_read: DTMF end
passthrough '6' on SIP/1001-0055
[Sep 1 11:54:23] DTMF[18353]: channel.c:4109 __ast_read: DTMF end '4'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:23] DTMF[18353]: channel.c:4178 __ast_read: DTMF end
passthrough '4' on SIP/1001-0055
[Sep 1 11:54:23] DTMF[18353]: channel.c:4109 __ast_read: DTMF end '6'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:23] DTMF[18353]: channel.c:4178 __ast_read: DTMF end
passthrough '6' on SIP/1001-0055
[Sep 1 11:54:23] DTMF[18353]: channel.c:4109 __ast_read: DTMF end 'x'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:23] DTMF[18353]: channel.c:4178 __ast_read: DTMF end
passthrough 'x' on SIP/1001-0055
[Sep 1 11:54:23] DTMF[18353]: channel.c:4109 __ast_read: DTMF end 'x'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:23] DTMF[18353]: channel.c:4178 __ast_read: DTMF end
passthrough 'x' on SIP/1001-0055
[Sep 1 11:54:23] DTMF[18353]: channel.c:4109 __ast_read: DTMF end 'x'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:23] DTMF[18353]: channel.c:4178 __ast_read: DTMF end
passthrough 'x' on SIP/1001-0055
[Sep 1 11:54:23] DTMF[18353]: channel.c:4109 __ast_read: DTMF end 'x'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:23] DTMF[18353]: channel.c:4178 __ast_read: DTMF end
passthrough 'x' on SIP/1001-0055
[Sep 1 11:54:23] DTMF[18353]: channel.c:4109 __ast_read: DTMF end 'x'
received on SIP/1001-0055, duration 250 ms
[Sep 1 11:54:23] DTMF[18353]: channel.c:4178 __ast_read: DTMF end
passthrough 'x' on SIP/10
[asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily
Hello
when setting a local forward (in this case to extension 23) on a SIP
phone, I see the following on the Asterisk CLI :
[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back
from 11.22.33.44:40670
[Aug 31 14:59:34] -- Now forwarding
Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal'
(thanks to SIP/myaccount184-3729)
Question : how can I read the variable which contains the value
'myaccount184' in the context from-internal ?
The following variables I've tried are empty :
ChannelPeerip=${CHANNEL(peerip)}
Channelrecvip=${CHANNEL(recvip)}
Channelfrom=${CHANNEL(from)}
Channeluri=${CHANNEL(uri)}
Channeluseragent=${CHANNEL(useragent)})
You can see this on the CLI output here :
[Aug 31 14:59:34] -- Executing [23@from-internal:7]
NoOp("Local/23@from-internal-07f5;2", "ChannelPeerip= Channelrecvip=
Channelfrom=") in new stack
[Aug 31 14:59:34] -- Executing [23@from-internal:8]
NoOp("Local/23@from-internal-07f5;2", "Channeluri=
Channeluseragent=") in new stack
Anyone knows the correct variable to read ?
Kind regards
Jonas.
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New to Asterisk? Start here:
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