[asterisk-users] Asterisk and Teams integration?

[Carlos Chavez]

    Does anyone know of a good solution to integrate Asterisk and MS 
Teams?  Something where you can use the MS Teams client as a regular 
extension?


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Re: [asterisk-users] Deleting voicemail by program

[Carlos Chavez]

    The script included with Asterisk (messages-expire.pl) deletes 
older messages and then renumbers the rest of the messages.  I guess you 
need to do the same.


On 09/10/23 2:24 PM, Mike Diehl wrote:

Unfortunately, I'm using a version of asterisk that is old enough to not
benefit from this... 

Mike.

On Monday, October 9, 2023 3:15:45 PM EDT Michael Bradeen wrote:

Hi Mike,

New AMI actions were recently added to app_voicemail to let you remotely
manipulate a mailbox:
https://github.com/asterisk/asterisk/issues/181

Hope this helps.

BR,
-Mike

On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl  wrote:

Hi all,

I need to be able to delete a voicemail message using a program.

Is is sufficient to simply delete the .wav and .txt files in the spool
directory?
Or do I need to also renumber the remaining files?

For example, let say a given mailbox has 20 messages in it and I want to
delete message number 5.  Can I just delete the 2 files and expect that
asterisk will renumber them?  Or do I need to?

Also, is the answer the same when I migrate to storing voicemails in a
database?

Thanks in advance.

Mike



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[asterisk-users] Can ShanSpy be used on Local Channels?

[Carlos Chavez]

    Does anyone know if Chanspy can be used with local channels? Since 
agents on queues need to use local channels like Local/@from-queue/n 
to make sure that all of their registered extensions ring we are now 
having a problem trying to use ChanSpy to listen to calls.  Since we do 
not know if the agent is on their desk phone or a softphone (which use 
different identifiers) we cannot set a common rule like 
ChanSpy(PJSIP/).  Queuemetrics registers agent extensions as: 
Local/@from-queue but if I try to listen on that channel I get no 
audio.  Am I missing a parameter or is chanspy simply unable to use 
Local channels?


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[asterisk-users] Is there a good Python library for AMI?

[Carlos Chavez]

    I am switching many of my scripts to python and I found pyst2 in my 
search for an Asterisk library.  While it seems to work well for AGI 
acripts it seems very broken when using it for AMI.  Can anyone 
recommend a good and currently supported AMI library for python?


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Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

[Carlos Chavez]

You need to put your external IP in the transport configuration:

external_media_address=X.X.X.X
external_signaling_address=X.X.X.X
external_signaling_port=5060


On 21/06/23 12:36, TTT wrote:

I've split this thread off from another (PJSIP authentication) because I think 
the root cause is something different.I think the problem is the following 
FROM line in my SIP INVITE transaction:

From: "MYNAME" 
;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4

The IP address above is an internal/non-routable IP, so Twilio is rejecting it. 
 For some reason Asterisk is not replacing the private IP with my public IP 
address.  My pjsip.transport.conf contains a stanza for this transport with:

local_net=172.31.0.0/16

Is that all that's needed for Asterisk to replace the from IP with the external 
IP?  I'm not clear on why Asterisk is not substituting the private FROM ip with 
a public one...




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Re: [asterisk-users] PJSIP not performing outbound authentication

[Carlos Chavez]

    Dis you set "outbound_auth" in your endpoint configuration to Twilio?

On 21/06/23 11:19, TTT wrote:

I am using Asterisk 20.3.0 with PJSIP.  I have setup a trunk to my ISP
(Twilio) who requires outbound authentication.  My pjsip.auth.conf contains:

[Twilio]
type=auth
auth_type=userpass
password=mysecret
username=myun

However, my calls using the trunk are rejected with a 403. Using pjsip
logging I notice that the outgoing invite does not have an authentication
line. Why is Asterisk not sending credentials to the ISP? SIP transactions
are:
  > INVITE
  < 100 TRYING
  < 403 FORBIDDEN

Or is this normal?  Must Twilio respond with a 407 which will cause Asterisk
to authenticate?



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Re: [asterisk-users] Multiple phones on same PJSIP account

[Carlos Chavez]

    Are you using PJSIP_DIAL_CONTACTS in your dialplan?  You need to 
use that in the Dial command and not just PJSIP/XXX for all extensions 
to ring.


On 21/06/23 9:52, TTT wrote:

Ok I've got multiple phone sets registered with the same extension/secret.

However, this causes a strange problem.  If I have 3 phone sets registered on 
extension 123, and I then call extension 123 (from extension 456), only a 
SINGLE phone set will ring.

Is this by design or a bug?  Does only the most recently registered phone set 
ring when I call the extension?  Seems odd...is there a way to change it so ALL 
phones on the same extension will ring?  (I'm using SNOM + PANASONIC + Aastra 
phones)

Thanks
Brian

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Antony Stone
Sent: Monday, June 19, 2023 10:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Multiple phones on same PJSIP account

On Monday 19 June 2023 at 16:26:05, TTT wrote:


That begs another interesting question...with analog phones picking up
two extensions on the same "line" allow multiple people to participate
on the call (without a "conference" feature)

Does this become possible with multiple phones on the same PJSIP account?

No.


Antony.

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[asterisk-users] MixMonitor not recording through transfer

[Carlos Chavez]

    I have the following scenario:

Agent calls external number

Mixmonitor starts recording call

After agent speaks with customer they need to transfer them to an 
extension that will simply play a message


Customer hangs up

    The problem is that the recording stops the moment the agent 
transfers the call to the other extension.  We need the recording to 
include the message from the other extension.  We use Asterisk 16 on 
this server.  I know that the AUDIOHOOK_INHERIT function was deprecated 
long ago so I should not need anything extra to keep recording through 
transfers, or am I wrong?


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Re: [asterisk-users] Run asterisk -rx "command" and get plain text output

[Carlos Chavez]

    The "-n" option only works on startup and cannot be used when 
Asterisk is already running (I tried and I get an error).  We are using 
version 18.12.1.  The output I want to capture is:


asterisk -rx "queue show"

    If I capture the output to a file or straight to the command line I 
get the colorized output.


    I found a way to strip the ANSI codes using SED:

asterisk -rx "queue show" | sed -e 's/\x1b\[[0-9;]*m//g'


On 8/3/2022 10:35 AM, Joel Serrano wrote:

Have you tried adding “-n”?

Also, what version of asterisk are you using? newer versions only have 
colorized output when your are connected to the console (-r) not for 
remote commands (-rx)




On Wed, Aug 3, 2022 at 08:21 Carlos Chavez  wrote:

 I usually like to have the colorized output when looking at
asterisk output but I need to get some info by running "asterisk -rx"
and get just plain text output so I can mail it.  Right now I get
ANSI
codes in the output.  Is there a way to get plain text output for
just
that script and not disable colors for everything?

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[asterisk-users] Run asterisk -rx "command" and get plain text output

[Carlos Chavez]

    I usually like to have the colorized output when looking at 
asterisk output but I need to get some info by running "asterisk -rx" 
and get just plain text output so I can mail it.  Right now I get ANSI 
codes in the output.  Is there a way to get plain text output for just 
that script and not disable colors for everything?


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Re: [asterisk-users] Installing and configuring Opus?

[Carlos Chavez]

    If you compiles Asterisk by hand you need to make sure that 
codec_opus was selected (make menuconfig to check selections).  If you 
installed it from another source make sure that Opus is included (maybe 
an extra package).  Also, make sure that you modules.conf file is not 
explicitly blocking it (if you have autoload on).


    From the Asterisk cli do "modules show like codec" and you should 
see something like this:


codec_opus.so  OPUS Coder/Decoder   
0  Running extended


    If it is not there then you have not enabled it.

On 11/07/22 11:13, Brant Merryman wrote:

Hello Friends,

I am recently beginning to work with Asterisk. I am using version 
18.3.0 on Ubuntu 20.04. I have the Hello World example running and I 
am able to connect to Asterisk using Zoiper on a Mac to test it.


The next thing I would like to do is have Asterisk use Opus. It is 
currently using G.711 u-law for the call. What do I need to do in 
order to have it use Opus? I already set Opus as the preferred codec 
in Zoiper. I think I might need to install something and perhaps 
configure something in Asterisk.


I have already seen this page:

https://wiki.asterisk.org/wiki/display/AST/Codec+Opus

I configured *codecs.conf* as follows:

[opus]
type=opus
max_playback_rate=16000 ; Limit bandwidth to narrow band
fec=yes ;


However, doing so had no effect. What do I need to do in order to 
install Opus and configure it?


Thank you.

Brant Merryman



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[asterisk-users] How to use mixmonitor when transfering a call

[Carlos Chavez]

    I am having a problem with my recordings.  Mixmonitor is called in 
the "macro" when you dial an extension.  If that call is transferred to 
another extension then the recording is reset and we lose the recording 
for the original call.  How can I tell Mixmonitor to keep recording and 
not reset?  I have the "a" option set in mixmonitor to append and not 
overwrite.  Is there a way to know if Mixmonitor has already started on 
this call and avoid executing it again when transferred?  This is what I 
use:


[ Context 'grabar' created by 'pbx_config' ]
  's' =>    1. 
Set(MONITOR_FILE=${STRFTIME(${EPOCH},,%Y)}/${STRFTIME(${EPOCH},,%m)}/${UNIQUEID}.wav) 
[extensions.conf:0]

    2. Mixmonitor(${MONITOR_FILE},ba) [extensions.conf:0]
    3. Return() [extensions.conf:0]

    On every transfer the recording starts again at the moment. I 
thought the "a" option would simply append to the file and keep going.  
Any ideas?


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Re: [asterisk-users] R2 error Seize Timeout

[Carlos Chavez]

    The provider is the timing source.  Both wanpipe1.conf and 
system.conf have the timing sources set to the remote side:


TE_CLOCK     = NORMAL


span=1,1,0,CAS,HDB3

    I still have a feeling that the problem is on the providers side as 
during testing we never saw the issue.


    I have modified wanpipe1.conf to be CAS but the strange thing is 
that the freepbx gui does show CAS there but sets CCS on the 
configuration file.  Now I have to wait and see if the problem persists.


On 08/03/22 11:54, Duncan Turnbull wrote:

It’s been a r we hike since we used these cards.  This example may help

https://wiki.freepbx.org/plugins/servlet/mobile?contentId=73007457#content/view/73007457

My thinking is it sounds like a timing error. Make sure your provider 
is the timing source. Once it loses time you will get dropped calls 
until it resyncs


Good luck




On 9/03/2022, at 4:25 AM, Steinwendtner  wrote:

Hello,

I must admit that I have never set up an asterisk system with R2 
signalling. But from the config files


point of view, you stated TE_SIG_MODE in wanpipe1.conf as ccs which 
should be cas, right ?


If this does not help, you need to connect an external E1 Monitor.

Regards,

Hans

Am 08.03.22 um 06:41 schrieb Carlos Chavez:

    Last month we switched a Panasonic pbx with a Freepbx 16
appliance.  We use a single E1 in MFC/R2 (Mexico) with Telmex as a
provider.  This was connected for a couple of days for testing with no
problems before the client moved offices to a new location.  In the new
location we are now having a problem every few days where we get the
following error:

[2022-03-07 07:30:11] ERROR[3469][C-004c] chan_dahdi.c: Chan 10 -
Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF
state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08
[2022-03-07 07:30:44] ERROR[29573] chan_dahdi.c: Chan 10 - Protocol
error. Reason = Seize Timeout, R2 State = Clear Forward Transmitted, MF
state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08
[2022-03-07 07:32:15] ERROR[3704][C-004e] chan_dahdi.c: Chan 10 -
Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF
state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08
[2022-03-07 07:32:52] ERROR[29573] chan_dahdi.c: Chan 10 - Protocol
error. Reason = Seize Timeout, R2 State = Clear Forward Transmitted, MF
state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08

    When we see that error the E1 will no longer send or receive
calls.  Our solution has been to stop and restart Asterisk and
Wanconfig/Dahdi to restore service.  Since restarting solves it I am
wondering if the problem is on my side and not on the providers.  So far
it happens once or twice a week.  When we report this to the provider
they simply state that the problem is on our side (it is their default
position) unless we can provide evidence to the contrary.  Any
recommendations on how to debug this?

Here is wanpipe1.conf:
[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE     = AFT
S514CPU     = A
CommPort     = PRI
AUTO_PCISLOT     = NO
PCISLOT     = 4
PCIBUS  = 8
FE_MEDIA    = E1
FE_LCODE    = HDB3
FE_FRAME    = NCRC4
FE_LINE        = 1
TE_CLOCK     = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE    = NO
TE_RX_SLEVEL    = 430
HW_RJ45_PORT_MAP = DEFAULT
LBO         = 120OH
FE_TXTRISTATE    = NO
MTU         = 1500
UDPPORT        = 9000
TTL        = 255
IGNORE_FRONT_END    = NO
TDMV_SPAN        = 1
TDMV_DCHAN        = 16
TE_AIS_MAINTENANCE = NO #NO: defualt  YES: Start port in AIS
Blue Alarm and keep line down
#wanpipemon -i w1g1 -c Ttx_ais_off to
disable AIS maintenance mode
#wanpipemon -i w1g1 -c Ttx_ais_on to
enable AIS maintenance mode
TDMV_HW_DTMF        = NO # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT        = NO        # YES: receive fax 1100hz events
from hardware
HWEC_OPERATION_MODE = OCT_NORMAL    # OCT_NORMAL: echo cancelation
enabled with nlp (default)
        # OCT_SPEECH: improves software
tone detection by disabling NLP (echo possible)
        # OCT_NO_ECHO:disables echo
cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL   = NO # NO: default  YES: remove dtmf out of
incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION    = NO # NO: default  YES: reduces noise on the
line - could break fax
HWEC_ACUSTIC_ECHO   = NO # NO: default  YES: enables acustic echo
cancelation
HWEC_NLP_DISABLE    = NO # NO: default  YES: guarantees software
tone detection (possible echo)
HWEC_TX_AUTO_GAIN   = 0 # 0: disable   -40-0: default tx audio
level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN   = 0 # 0: disable   -40-0: default tx audio
level to be maintained (-20 default)
HWEC_TX_GAIN    = 0     # 0: disable   -24-24: db values to
be applied to tx signal
HWEC_RX_GAIN    = 0     # 0: disable   -24-24: db values to
be applied to tx signal

[w1g1]
ACTIVE_CH

[asterisk-users] R2 error Seize Timeout

[Carlos Chavez]

    Last month we switched a Panasonic pbx with a Freepbx 16 
appliance.  We use a single E1 in MFC/R2 (Mexico) with Telmex as a 
provider.  This was connected for a couple of days for testing with no 
problems before the client moved offices to a new location.  In the new 
location we are now having a problem every few days where we get the 
following error:


[2022-03-07 07:30:11] ERROR[3469][C-004c] chan_dahdi.c: Chan 10 - 
Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF 
state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08
[2022-03-07 07:30:44] ERROR[29573] chan_dahdi.c: Chan 10 - Protocol 
error. Reason = Seize Timeout, R2 State = Clear Forward Transmitted, MF 
state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08
[2022-03-07 07:32:15] ERROR[3704][C-004e] chan_dahdi.c: Chan 10 - 
Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF 
state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08
[2022-03-07 07:32:52] ERROR[29573] chan_dahdi.c: Chan 10 - Protocol 
error. Reason = Seize Timeout, R2 State = Clear Forward Transmitted, MF 
state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08


    When we see that error the E1 will no longer send or receive 
calls.  Our solution has been to stop and restart Asterisk and 
Wanconfig/Dahdi to restore service.  Since restarting solves it I am 
wondering if the problem is on my side and not on the providers.  So far 
it happens once or twice a week.  When we report this to the provider 
they simply state that the problem is on our side (it is their default 
position) unless we can provide evidence to the contrary.  Any 
recommendations on how to debug this?


Here is wanpipe1.conf:
[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE     = AFT
S514CPU     = A
CommPort     = PRI
AUTO_PCISLOT     = NO
PCISLOT     = 4
PCIBUS  = 8
FE_MEDIA    = E1
FE_LCODE    = HDB3
FE_FRAME    = NCRC4
FE_LINE        = 1
TE_CLOCK     = NORMAL
TE_REF_CLOCK    = 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE    = NO
TE_RX_SLEVEL    = 430
HW_RJ45_PORT_MAP = DEFAULT
LBO         = 120OH
FE_TXTRISTATE    = NO
MTU         = 1500
UDPPORT        = 9000
TTL        = 255
IGNORE_FRONT_END    = NO
TDMV_SPAN        = 1
TDMV_DCHAN        = 16
TE_AIS_MAINTENANCE = NO #NO: defualt  YES: Start port in AIS 
Blue Alarm and keep line down
    #wanpipemon -i w1g1 -c Ttx_ais_off to 
disable AIS maintenance mode
                            #wanpipemon -i w1g1 -c Ttx_ais_on to 
enable AIS maintenance mode

TDMV_HW_DTMF        = NO        # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT        = NO        # YES: receive fax 1100hz events 
from hardware
HWEC_OPERATION_MODE = OCT_NORMAL    # OCT_NORMAL: echo cancelation 
enabled with nlp (default)
                                    # OCT_SPEECH: improves software 
tone detection by disabling NLP (echo possible)
                                    # OCT_NO_ECHO:disables echo 
cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL   = NO    # NO: default  YES: remove dtmf out of 
incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION    = NO    # NO: default  YES: reduces noise on the 
line - could break fax
HWEC_ACUSTIC_ECHO   = NO    # NO: default  YES: enables acustic echo 
cancelation
HWEC_NLP_DISABLE    = NO    # NO: default  YES: guarantees software 
tone detection (possible echo)
HWEC_TX_AUTO_GAIN   = 0 # 0: disable   -40-0: default tx audio 
level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN   = 0 # 0: disable   -40-0: default tx audio 
level to be maintained (-20 default)
HWEC_TX_GAIN    = 0        # 0: disable   -24-24: db values to 
be applied to tx signal
HWEC_RX_GAIN    = 0        # 0: disable   -24-24: db values to 
be applied to tx signal


[w1g1]
ACTIVE_CH    = ALL
TDMV_HWEC    = NO
MTU         = 8

    Here is system.conf

span=1,1,0,CAS,HDB3
cas=1-10,11-15,17-31:1101
echocanceller=oslec,1-10,11-15,17-31
loadzone=mx
defaultzone=mx

    Here is chan_dahdi.conf

signalling=mfcr2
mfcr2_variant=mx
mfcr2_get_ani_first=no
mfcr2_max_ani=10
mfcr2_max_dnis=4
mfcr2_category=national_priority_subscriber
mfcr2_call_files=no
mfcr2_mfback_timeout=-1
mfcr2_metering_pulse_timeout=-1
mfcr2_allow_collect_calls=yes
mfcr2_double_answer=no
mfcr2_immediate_accept=no
mfcr2_accept_on_offer=yes
mfcr2_skip_category=no
mfcr2_forced_release=no
mfcr2_charge_calls=yes
group=0
context=from-digital
channel=>1-10

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[asterisk-users] Local channel sometimes have no audio

[Carlos Chavez]

    We recently upgraded a very old server from Asterisk 1.8 to 18.9 
and we are having a strange issue with calls in queues.  We use 
Queuemetrics to manage our agents and extensions are configured as 
Local/@from-queue/n to connect clients to agents.  This is because 
if you dial PJSIP/ directly it will only call the first registered 
device for that extension and not all of them.  The problem we are 
facing is that sometimes there is no audio on the call coming from the 
queues.  Most of the time everything is fine and it is very random when 
audio does not flow.  I have tried to debug SIP but I do not see 
anything wrong and audio should be flowing, it can take a long time 
between failures so it is very difficult to catch the correct call.


    I have another pbx using the same setup with Asterisk and 
Queuemetrics, only difference is that it uses Asterisk 18.5 and we do 
not have any problems with audio there.  As far as I know only calls 
from the queues have this issue, direct calls to the extensions or 
outgoing calls do not.  Could this be an issue with the Local channel?  
I use the /n because that is the example published in the Queuemetrics 
documentation.  I do not know if removing it will affect the call.  Any 
recommendations to properly debug this issue?


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Re: [asterisk-users] automating "make menuselect" options when building

[Carlos Chavez]

On 08/11/21 11:53, Kingsley Tart wrote:


Hi,

I realise that this is not really specific to Asterisk, but this seems
as sensible a place to ask as any.

If I want to create a script to automate the build of my chosen
Asterisk setup, what's the best way to automate my selections that I
did interactively when I ran "make menuselect" ?

I think doing that created a makeopts file somewhere. If I were to just
copy in the makeopts file I created from a build on another machine, is
there a process that can check that to ensure that everything selected
in that can be built, or is doing "make" and seeing whether that worked
the only way?

Cheers,
Kingsley.


    Just use the something like the following in your script:

make menuselect.makeopts
menuselect/menuselect --enable codec_opus --enable codec_silk --enable 
codec_siren7--enable codec_siren14 menuselect.makeopts


Docs are here:

https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options#UsingMenuselecttoSelectAsteriskOptions-ControllingMenuselect


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Re: [asterisk-users] Delay when dialing...

[Carlos Chavez]

    Thank you.  The server is running dnsmasq locally for DNS 
resolution and all queries resolve properly.  I just added the hostname 
to /etc/hosts and restarted but the delay persists.


On 7/23/2021 1:41 AM, Jean Aunis wrote:

Le 22/07/2021 à 18:32, Carlos Chavez a écrit :
    I started noticing a few days ago that whenever I dial any number 
or extension there is a delay of 5 to 10 seconds before Asterisk 
reacts.  I see nothing on the CLI for that time and then the call 
goes through.  I have checked my network to make sure there is 
nothing slowing down packets between the phones and the server.


    Any settings I should check on the Asterisk side?  This is 
happening with all phones (several brands).



Hi,

I've seen this problem several times when there is no DNS resolution 
of Asterisk's hostname.


Try to add your hostname to /etc/hosts and check if it's better.

Regards,

Jean




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[asterisk-users] Delay when dialing...

[Carlos Chavez]

    I started noticing a few days ago that whenever I dial any number 
or extension there is a delay of 5 to 10 seconds before Asterisk 
reacts.  I see nothing on the CLI for that time and then the call goes 
through.  I have checked my network to make sure there is nothing 
slowing down packets between the phones and the server.


    Any settings I should check on the Asterisk side?  This is 
happening with all phones (several brands).


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[asterisk-users] PJSIP_DIAL_CONTACTS and Queues

[Carlos Chavez]

    Is there a solution to dial multiple contacts for a Queue agent?  
Since the pandemic started many of our customers have begun to move 
agents off site.  Since most of them were using softphones we did not 
have any problems but now we have one case where the agents have a desk 
phone in their cubicle and are using a softphone from home.  For regular 
calls there is no problem as PJSIP_DIAL_CONTACTS dials all the contacts 
for the extension, but Queues are only sending calls to a single (the 
first) contact so Queue calls are going unanswered.  For the moment we 
will power down the office phones but is there a solution for the Queue 
to ring both phones?


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Re: [asterisk-users] Some calls drop after 30 seconds

[Carlos Chavez]

On 08/09/20 4:16, Joshua C. Colp wrote:

On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <mailto:cur...@telecomab.mx>> wrote:


 Some users have complained that their calls drop after about 30
seconds.  Not all, just some.  After looking at the log files the
only
difference I can find from the dropped calls is the following line:

[2020-09-07 11:29:59] VERBOSE[21666][C-0055] bridge.c: Bridge
14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge
technology to native_rtp

 Most calls just do:

[2020-09-07 18:13:56] VERBOSE[15293][C-0084] bridge_channel.c:
Channel PJSIP/1028-012a joined 'simple_bridge' basic-bridge
<626258fc-0649-45c7-b0d3-630a06d2c91b>

 Why are some calls using the simple bridge and others switch
to the
native_rtp bridge?  Could this be a codec problem?  How can I prevent
the switch?


It depends on the channels involved as well as the features in use. To 
prevent direct media from occurring you can set the "direct_media" 
option to "no" on the endpoint. The native_rtp bridge can still be 
used, though, to provide more efficient in-Asterisk forwarding of media.


If that doesn't change things you'd need to examine further, such as 
looking at the SIP trace for a call (pjsip set logger on) as 30 
seconds is close to the amount of time for a lost ACK to a 200 OK, 
which generally indicates a NAT issue.



    Direct media is off for all endpoints (both trunks and phones).  
There is no NAT on either side, the phones are on the local network and 
the trunk provider has a direct link and the pbx has a dedicated 
ethernet port for it.  We have two trunk providers and I only see the 
native rtp bridge used on one of them.  I will do a packet capture on 
the trunk interface to see if something else strange happens.


    Thank you.

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[asterisk-users] Some calls drop after 30 seconds

[Carlos Chavez]

    Some users have complained that their calls drop after about 30 
seconds.  Not all, just some.  After looking at the log files the only 
difference I can find from the dropped calls is the following line:


[2020-09-07 11:29:59] VERBOSE[21666][C-0055] bridge.c: Bridge 
14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge 
technology to native_rtp


    Most calls just do:

[2020-09-07 18:13:56] VERBOSE[15293][C-0084] bridge_channel.c: 
Channel PJSIP/1028-012a joined 'simple_bridge' basic-bridge 
<626258fc-0649-45c7-b0d3-630a06d2c91b>


    Why are some calls using the simple bridge and others switch to the 
native_rtp bridge?  Could this be a codec problem?  How can I prevent 
the switch?


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Re: [asterisk-users] Channels freeze on Confbridge

[Carlos Chavez]

On 25/08/20 7:20, Andrew Yager wrote:

On Sun, 23 Aug 2020 at 18:23, Antony Stone 
> wrote:


On Saturday 22 August 2020 at 22:51:48, Sebastian Nielsen wrote:

> I had a similiar problem, but with calls dropping after 30 sec.
> It turned out that Android didn't support RP-CID (Reverse Party
Caller ID)
> so when I sent the name of the callee to the caller (as some sort of
> "centralized phonebook function") it caused calls to be dropped
as android
> refused to reply on the packets or sent rejections back.

I can see the point you're making here, but what's going to do
this after 30
*minutes* of normal call?


Have seen plenty of ALGs do weird things like this. 30 minutes is a 
nice number, and nice enough that I'd go hunting for ALG issues. It's 
a good multiple of 3 minutes, and quite possibly is some big number 
someone thought to set in something that "no one would ever hit".


A tcpdump would probably show you what's going on if the logs are 
otherwise unclear, and you could also make sure you have sensible RTP 
timeout rules.


Andrew


    We are zeroing in on something with the SIP trunk provider.  We 
have are testing a new carrier and so far we have not seen the same problem.


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[asterisk-users] Channels freeze on Confbridge

[Carlos Chavez]

    I am having a strange problem.  We have an Asterisk 16.12.0 server 
(we have upgraded at least two versions since we found the problem) 
where users complain that confbridge calls end after about 30 minutes or 
so.  The problem is that according to Asterisk the calls are still 
active.  All users are cut off at the same time but a "core show 
channels verbose" still shows channels as active:


CBAnn/902-002f;1 default  s   1 Up  
(None)   (Empty) 04:03:43
CBAnn/902-002f;2 default  s   1 Up  
(None)   (Empty) 04:03:43 6e7710ea-7c0f-4c7e-a


CBAnn/903-0036;2 default  s   1 Up  
(None)   (Empty) 02:47:04 05e10e42-85ec-4120-b
CBAnn/903-0036;1 default  s   1 Up  
(None)   (Empty) 02:47:04


PJSIP/directo-0001b7 oficina  903 2 Up  
ConfBridge   903   8110221265 02:40:43 general 
general 05e10e42-85ec-4120-b


PJSIP/directo-0001af oficina  902 2 Up  
ConfBridge   902   8992596823 04:25:50 general 
general 6e7710ea-7c0f-4c7e-a


I have to manually hangup the channels.  The PSTN provider is a SIP trunk.

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[asterisk-users] One way audio on outgoing calls

[Carlos Chavez]

    I am having a strange problem with a new provider.  We already have 
a couple SIP trunks working fine.  We are trying a new provider but we 
are having one way audio problems with outgoing calls.  Incoming calls 
do have two way audio, only outgoing calls have this problem.  I do not 
see anything odd with a packet capture and using PJSIP history to 
check.  The provider says that on outgoing calls the get random 
characters instead of the media port for RTP.


    We are using Asterisk 16.12.0 with PJSIP.  The server is behind NAT 
so we have external_media_address and external_signaling_address set to 
the public IP and all relevant ports are forwarded to the Asterisk 
server.  The other SIP trunks work fine, only this new provider has a 
problem and only for outgoing calls.


    An rtp set debug on shows only outgoing packets to the media 
address but no incoming packets.  Why would there be a difference that 
makes it work on incoming calls but not on outgoing?


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Re: [asterisk-users] No CID between Asterisk using IAX trunk

[Carlos Chavez]

    Could the difference be that you need to use type=friend for CID to 
work?  Using type=peer we can forgo auth since we are not using public 
infrastructure.  My other trunks do not have allowcallerid=yes so I will 
add that and test it.  Thanks.


On 02/03/20 12:54, Doug Lytle wrote:

My Asterisk 13 IAX2 trunk posted below:

type=friend
trunk=yes
allowcallerid=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
host=my.super.duper.host
username=my.super.duper.username
secret=my.super.duper.secret
context=sip
qualify=500
qualifysmoothing=yes
requirecalltoken=no
trunk=yes
jitterbuffer=yes
forcejitterbuffer=yes
maxjitterbuffer=300
maxjitterinterps=100
resyncthreshold=1500
tos=ef
cos=5

Doug


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Re: [asterisk-users] No CID between Asterisk using IAX trunk

[Carlos Chavez]

    Not these particular two servers.

On 02/03/20 12:16, Doug Lytle wrote:

  I am trying to troubleshoot two Asterisk servers that have an IAX2
trunk between them.

Carlos,

Had caller-id ever worked between these two systems?

Doug


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[asterisk-users] No CID between Asterisk using IAX trunk

[Carlos Chavez]

    I am trying to troubleshoot two Asterisk servers that have an IAX2 
trunk between them.  Calls come and go but there is no CallerID from the 
remote server either way.  One of the servers is running Asterisk 16 and 
the other is an older 1.8 install (I know, I am trying to get permission 
to update).  The trunk between servers is very simple.  Something like:


Server 1 (Mexico)

[panama]
type=peer
context=oficina
trunk=yes
disallow=all
allow=g729
qualify=yes
requirecalltoken=no
host=10.X.X.141
language=es
callerid=asreceived

Server 2 (Panama)

[mexico]
type=peer
context=oficina
trunk=yes
disallow=all
allow=g729
qualify=yes
requirecalltoken=no
host=10.Y.Y.5
language=es
callerid=asreceived

So from Panama to Mexico we use:

exten => _1XXX,1,Dial(IAX2/mexico/${EXTEN})


Call comes in and is answered but there is no CID in CDR or in the phone 
display.  Other trunks to other servers have no problem sending CID from 
one server to the other (all using IAX).  Any pointers?


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[asterisk-users] DTMF not working on incoming calls

[Carlos Chavez]

    What is  the best way to debug DTMF on a PJSIP trunk?  I have 
cycled through all available options 
('rfc4733','inband','info','auto','auto_info') but my IVR does not 
recognize any options from the remote end. I have also tried changing 
codecs from g729 to alaw or ulaw with the same result.  Outgoing calls 
do not seem to have this problem, just incoming.  This is with Asterisk 
13.29.2 but the problem started with 13.21 before I decided to upgrade 
to the latest 13.x version.  Any pointers?


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Re: [asterisk-users] Stuck "channel"

[Carlos Chavez]

    So a restart is the only way to get rid of it?

On 11/1/2019 9:28 AM, Richard Mudgett wrote:



On Thu, Oct 31, 2019 at 11:05 PM Carlos Chavez <mailto:cur...@telecomab.mx>> wrote:


I have tried both by hand and hitting tab to auto complete:

*CLI> channel request hangup Message/ast_msg_queue
Message/ast_msg_queue is not a known channel


This channel is used for processing all out of dialog SIP MESSAGE 
requests in the dialplan.  It cannot be hung up.


Richard


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Re: [asterisk-users] Stuck "channel"

[Carlos Chavez]

I have tried both by hand and hitting tab to auto complete:

*CLI> channel request hangup Message/ast_msg_queue
Message/ast_msg_queue is not a known channel


On 31/10/19 14:18, Sean Bright wrote:

On 10/31/2019 2:13 PM, Carlos Chavez wrote:
I assume this is something created by Freepbx.  If I do a "channel 
request hangup" it tells me the channel does not exist. Any ideas


Are you trying to hang up "Message/ast_msg_queu" or are you hitting 
the tab key to complete it in the CLI? "Message/ast_msg_queu" is not 
the full channel name, it's being truncated for display purposes.





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[asterisk-users] Stuck "channel"

[Carlos Chavez]

    Since yesterday I have a stuck channel on my Asterisk server and I 
do not know how to eliminate it:



Message/ast_msg_queu macro-dial-one   s  59 Up  
Dial PJSIP/1218/sip:1218@192.1 17:24:07


    I assume this is something created by Freepbx.  If I do a "channel 
request hangup" it tells me the channel does not exist. Any ideas?



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Re: [asterisk-users] Asterisk and CentOS 8

[Carlos Chavez]

    They only problem I have found so far is while trying to install 
Alembic for SQLAlchemy (for realtime configs).  Those are the only 
packages that I cannot get working properly.  Vanilla Asterisk works 
fine  with the only extra package needed being libedit-devel that is not 
included in any "official" repo.  You need to download the Fedora Core 
29 packages to in order to successfully compile Asterisk.  That being 
said, I would not recommend trying to put this in production any time soon.


On 10/17/2019 11:19 AM, George Joseph wrote:
At the current time, we do not recommend attempting to build Asterisk 
on CentOS 8.  Many packages Asterisk uses are not yet available and 
would require building from their sources.  The Asterisk packages are 
also not available in the EPEL 8 or CentOS 8 repositories yet for the 
same reason.


We'll update you when we think it's safe.


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Re: [asterisk-users] Polycom BLF Question

[Carlos Chavez]

    This is done via the custom extension state or hints. Basically you 
create a custom hint for 444 and monitor that on your phone like any 
other extension.  You then enable or disable the hint in the same 
dialplan for 444 and 555.


https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/usingCustomDeviceStates.html


On 9/8/2019 4:32 PM, Alexander Perkins wrote:
Hi All. I have an interesting scenario. We use the Polycom VXX phones 
and have an auto-attendant on our Asterisk system. The receptionist 
can turn the auto-attendant off and on as she would like (she dials 
444 to enable and 555 to disable). However, I’d like to have one of 
the BLFs on her Polycom light up if the auto-attendant is enabled and 
off if it is disabled.


Any suggestions on how I can have the one of the Polycom BLFs stay on 
if the auto-attendant is enabled?


Any help is appreciated

Thanks,
Alex


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Re: [asterisk-users] Wanted: professional softphone

[Carlos Chavez]

On 7/24/19 11:41 PM, Michael Maier wrote:


Hello!

Does anybody by chance know of a softphone which additionally has a management 
suite to fully configure it userID based for Windows clients? Any idea is 
appreciated!

    Zulu from Sangoma allows you to generate a QR code that configures 
everything automatically for each user.  Been using it lately and it 
works very well.  Only downside is that it is only for FreePBX/PBXact.




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[asterisk-users] Calling GOSUB from Macro on Asterisk 1.8

[Carlos Chavez]

    I know we should not be running an Asterisk so old but this 
customer really does not want to replace this particular installation.  
I am having a problem when calling Gosub from a macro.  It seems that if 
I call gosub and return to the macro all Macro related variables like 
MACRO_EXTEN and ARG are gone.  Is that expected or should I look for a 
problem elsewhere?


    On the CLI I can see the call enter the macro, the use the GOSUB 
code and when it returns to try to dial I have something like: 
Dial(${ARG1}) (where ARG1 is a parameter sent to the macro).  I get an 
error:


WARNING[21682] app_dial.c: Dial requires an argument (technology/number)

    This is because the ARG variable is no longer available after the 
GOSUB.  If I remove the gosub everything works as expected.  I know 
macros are deprecated but they are in use in most of my servers and it 
will take a while to replace everything.  Any pointers?


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Re: [asterisk-users] why doesn't extension "s" work ?

[Carlos Chavez]

On 3/28/2019 6:32 PM, sean darcy wrote:


I'm using "s" extension in my dialplan:

[gv-voice]
exten => s,1,Verbose(callerid is "${CALLERID(all)}" or 
"${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)})  ; 
PJSIP_HEADER(read,To)

   same=>n,

But when a call comes in to the gv-voice context, "s" doesn't match 
the extension:


res_pjsip_session.c:2991 new_invite: Call from 'gv-voice' 
(UDP:10.10.10.80:5062) to extension '' rejected because 
extension not found in context 'gv-voice'.


I thought "s" (as in start ?) would match any extension sent to that 
context.


sean

The "s" extension does not work because you are calling a numbered 
extension. If the call was coming in without an explicit number it would 
work. If you want to use your s extension in this case you need to add: 
exten => _X.,1,Goto(s,1) That will redirect any call coming in to your s 
extension.



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Re: [asterisk-users] AMI mulitple calls quickly

[Carlos Chavez]

On 3/12/19 11:03 AM, Steve Edwards wrote:


On Mon, 11 Mar 2019, Jerry Geis wrote:

If I use the AMI interface to originate a call, close the connection, 
open another connection etc...This works. but is slow...


Would opening multiple AMI connections be an option?

    You should be able to send multiple originate commands without 
having to wait as long as you are using the "Async" option in your 
originate action.  That way you just send all your calls and then wait 
for an "OriginateResponse" event to know the result of each of your 
calls.  Just remember to also use an "ActionID" to identify each call so 
you can match the response event.


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Re: [asterisk-users] Various extensions ring once and go to voicemail

[Carlos Chavez]

On 1/14/19 4:30 PM, Thomas Peters wrote:


Here’s what I get:

apbx*CLI> module show like timing

Module Description  Use Count

res_timing_pthread.so pthread Timing Interface 0

res_timing_dahdi.so DAHDI Timing Interface   4

2 modules loaded

So what would you suggest? (And thanks in advance.)

Thomas M. Peters | Sr. Systems Administrator | tpet...@mcts.org 
<mailto:tpet...@mcts.org>
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*From:*asterisk-users  *On 
Behalf Of *Carlos Chavez

*Sent:* Monday, January 14, 2019 4:08 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Various extensions ring once and go to 
voicemail


On 1/14/19 4:02 PM, Duncan Turnbull wrote:

Sent from my iPad


On 15/01/2019, at 10:34 AM, Thomas Peters mailto:tpet...@mcts.org>> wrote:

Duncan:

You may have it right—I took one phone and set the ring time
to 60 seconds. I now get about 4 rings on that one.

I wonder how I can change the timing source.

In one version (and I can’t recall which) asterisk moved to an
internal timing system, to avoid the hardware need.

There should be quite a lot of discussion of it in the archives or
perhaps voipinfo

I don’t know if you can slow the VM processor speed? I am guessing
it is counting something much faster than it used to

Cheers Duncan

*CLI> module show like timing
Module Description  Use Count Status  
Support Level
res_timing_dahdi.so    DAHDI Timing 
Interface   0  Running core
res_timing_pthread.so  pthread Timing 
Interface 0  Running  extended
res_timing_timerfd.so  Timerfd Timing 
Interface 1  Running  core

3 modules loaded

    This will show you what module Asterisk is using for timing.  You 
can try doing a noload on the two you do not need.


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If your Asterisk server is a VM it should not be using DAHDI as a 
timing source.  The res_timing_timerfd.so module would probably be the best 
candidate.
Make sure your /etc/asterisk/modules.conf loads the timerfd module (and make 
sure it was built or installed by your package).

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Re: [asterisk-users] Various extensions ring once and go to voicemail

[Carlos Chavez]

On 1/14/19 4:02 PM, Duncan Turnbull wrote:




Sent from my iPad

On 15/01/2019, at 10:34 AM, Thomas Peters > wrote:



Duncan:

You may have it right—I took one phone and set the ring time to 60 
seconds. I now get about 4 rings on that one.


I wonder how I can change the timing source.



In one version (and I can’t recall which) asterisk moved to an 
internal timing system, to avoid the hardware need.


There should be quite a lot of discussion of it in the archives or 
perhaps voipinfo


I don’t know if you can slow the VM processor speed? I am guessing it 
is counting something much faster than it used to


Cheers Duncan



*CLI> module show like timing
Module Description  Use Count  Status 
Support Level
res_timing_dahdi.so    DAHDI Timing Interface   
0  Running  core
res_timing_pthread.so  pthread Timing Interface 
0  Running  extended
res_timing_timerfd.so  Timerfd Timing Interface 
1  Running  core

3 modules loaded

    This will show you what module Asterisk is using for timing. You 
can try doing a noload on the two you do not need.



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[asterisk-users] Is the R2 list still up?

[Carlos Chavez]

    I am trying to send messages to asterisk...@lists.digium.com but I 
do not get an error or any messages back.  In the archive I do not see 
any messages past November 2018.


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[asterisk-users] Problem receiving calls with Telmex in Mexico...

[Carlos Chavez]

    Hi.  I am having a problem when trying to receive calls via en E1 
from  Telmex using MFC/R2 (MX Variant).  Outgoing calls are fine.  We 
are using a PBXact system with a Digium TE420 (5th Gen) card.  Here is a 
log from the call:


[10:46:37:707] [Thread: 140631230322432] [Chan 1] - Call started at Sat 
Jan 12 10:46:37 2019 on chan 1 [openr2 version 1.3.3, revision (release)]
[10:46:37:707] [Thread: 140631230322432] [Chan 1] - Initialized R2 MF 
detector
[10:46:37:707] [Thread: 140631230322432] [Chan 1] - CAS Tx >> [SEIZE 
ACK] 0x0C

[10:46:37:707] [Thread: 140631230322432] [Chan 1] - CAS Raw Tx >> 0x0D
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - MF Rx << 2 [ON]
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 0

[10:46:37:781] [Thread: 140631230322432] [Chan 1] - Cannot cancel timer 0
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - Getting DNIS digit 2
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - DNIS so far: 2, 
expected length: 4
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - Requesting next DNIS 
with signal 0x31.

[10:46:37:781] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [ON]
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - scheduled timer id 2 
(mf_back_cycle)

[10:46:37:861] [Thread: 140631230322432] [Chan 1] - MF Rx << 2 [OFF]
[10:46:37:861] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [OFF]
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - MF Rx << 8 [ON]
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 2
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - timer id 2 found, 
cancelling it now

[10:46:37:921] [Thread: 140631230322432] [Chan 1] - Getting DNIS digit 8
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - DNIS so far: 28, 
expected length: 4
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - Requesting next DNIS 
with signal 0x31.

[10:46:37:921] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [ON]
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - scheduled timer id 3 
(mf_back_cycle)

[10:46:38:001] [Thread: 140631230322432] [Chan 1] - MF Rx << 8 [OFF]
[10:46:38:001] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [OFF]
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - MF Rx << 0 [ON]
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 3
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - timer id 3 found, 
cancelling it now

[10:46:38:061] [Thread: 140631230322432] [Chan 1] - Getting DNIS digit 0
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - DNIS so far: 280, 
expected length: 4
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - Requesting next DNIS 
with signal 0x31.

[10:46:38:061] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [ON]
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - scheduled timer id 4 
(mf_back_cycle)

[10:46:38:121] [Thread: 140631230322432] [Chan 1] - MF Rx << 0 [OFF]
[10:46:38:121] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [OFF]
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - MF Rx << 0 [ON]
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 4
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - timer id 4 found, 
cancelling it now

[10:46:38:181] [Thread: 140631230322432] [Chan 1] - Getting DNIS digit 0
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - DNIS so far: 2800, 
expected length: 4

[10:46:38:181] [Thread: 140631230322432] [Chan 1] - Done getting DNIS!
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - MF Tx >> 6 [ON]
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - scheduled timer id 5 
(mf_back_cycle)

[10:46:38:261] [Thread: 140631230322432] [Chan 1] - MF Rx << 0 [OFF]
[10:46:38:261] [Thread: 140631230322432] [Chan 1] - MF Tx >> 6 [OFF]
[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Bits changed from 
0x00 to 0x08
[10:46:38:577] [Thread: 140631230322432] [Chan 1] - CAS Rx << [CLEAR 
FORWARD] 0x08
[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Far end 
disconnected. Reason: Normal Clearing

[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Call ended
[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 0

[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Cannot cancel timer 0

Basically all I see on the CLI is that a call comes in to Channel 1 and 
hangs up immediately.  Here is the system.conf:


span=1,1,0,CAS,HDB3
span=2,1,0,CAS,HDB3
span=3,1,0,CAS,HDB3
span=4,1,0,CAS,HDB3
cas=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124:1101
echocanceller=oslec,1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124
loadzone=mx

Here is the relevant chan_dahdi.conf (it is auto generated by PBXact so 
it is in several files:


[general]

; generated by module
#include chan_dahdi_general.conf

; for user additions not provided by module
#include chan_dahdi_general_custom.conf

[channels]
language=es
busydetect=yes
busycount=5
usecallerid=no
callwaiting=no
usecallingpres=no
threewaycalling=no
transfer=yes

[asterisk-users] Asterisk 15 and Cepstral

[Carlos Chavez]

    It seems that app_swift does not work with Asterisk 15 or 16.  I 
just get errors when trying to compile:


[root@pbxoficina app_swift]# ./configure
checking gcc...
checking swift...
checking asterisk...
creating Makefile
  
  *  Now run 'make' to compile app_swift *
  
[root@pbxoficina app_swift]# make

gcc -I/opt/swift/include -I/usr/include -g -Wall -fPIC -D_SWIFT_VER_6 
-D_AST_VER_15 -c -o built/app_swift.o app_swift.c

In file included from app_swift.c:34:0:
/usr/include/asterisk.h:219:2: error: #error "Externally compiled 
modules must declare AST_MODULE_SELF_SYM."

 #error "Externally compiled modules must declare AST_MODULE_SELF_SYM."
  ^
app_swift.c:35:1: error: expected declaration specifiers or ‘...’ before 
string constant

 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 305000 $")
 ^
app_swift.c:35:33: error: expected declaration specifiers or ‘...’ 
before string constant

 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 305000 $")
 ^
In file included from app_swift.c:37:0:
/opt/swift/include/swift.h:392:1: error: unknown type name ‘swift_voice’
 swift_voice * SWIFT_CALLCONV

..

    Is there a better way to use Cepstral voices for TTS on Asterisk?


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Re: [asterisk-users] WebRTC as Softphone substitute ?

[Carlos Chavez]

On 9/26/18 10:20 AM, Matthew Fredrickson wrote:


On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez  wrote:

On 9/26/2018 4:46 AM, Olivier wrote:


Hello,

This morning, I asked myself if WebRTC could be a viable alternative
to softphone deployment.

For me, main issue with Softphones is the amount of work needed for
installation and configuration.
Also, Softphones must be carefully choosen if Deskphone-like quality
is expected.

Now that WebRTC becomes ubiquitous, it might make sense to trade
Softphone features (call history, BLF, ...) for WebRTC deployment
simplicity.

What do you think of this ?
What kind of experience did you met with such WebRTC deployments ?
What about classic telephony features (CallTransfer) ?
Have you tried Cyber Maga Phone 2K ?


  If you can get it to work WebRTC is a good option.  The problem is
that any changes in your network may disrupt it and even trying to
replicate your installation is difficult.  I have it working fine on my
website so customers can call us directly from our web page but I never
could get Cyber Mega Phone 2K to work on the same server.  We used JSSIP
to create the webrtc phone on our website.

We just updated the documentation for how to get CMP2K working on the
wiki [1].  We'd love some feedback if you still have issues getting it
setup so that we can improve the docs.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone

Best wishes,
Matthew Fredrickson

    I followed the procedure indicated in the link but I cannot get 
remote video.  I can only see my own feed.  We do have audio for a 
little while.  For some reason the users get disconnected after a few 
minutes even though you can still see your video feed on screen.  This 
was done with Asterisk 15.6.0


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Re: [asterisk-users] WebRTC as Softphone substitute ?

[Carlos Chavez]

On 9/26/2018 4:46 AM, Olivier wrote:


Hello,

This morning, I asked myself if WebRTC could be a viable alternative 
to softphone deployment.


For me, main issue with Softphones is the amount of work needed for 
installation and configuration.
Also, Softphones must be carefully choosen if Deskphone-like quality 
is expected.


Now that WebRTC becomes ubiquitous, it might make sense to trade 
Softphone features (call history, BLF, ...) for WebRTC deployment 
simplicity.


What do you think of this ?
What kind of experience did you met with such WebRTC deployments ?
What about classic telephony features (CallTransfer) ?
Have you tried Cyber Maga Phone 2K ?



    If you can get it to work WebRTC is a good option.  The problem is 
that any changes in your network may disrupt it and even trying to 
replicate your installation is difficult.  I have it working fine on my 
website so customers can call us directly from our web page but I never 
could get Cyber Mega Phone 2K to work on the same server.  We used JSSIP 
to create the webrtc phone on our website.


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Re: [asterisk-users] AGI timeout option

[Carlos Chavez]

On 9/13/2018 8:04 PM, Patrick Wakano wrote:


Hello list,
Hope you all doing  well!

Recently, I had an issue with a FastAGI PHP script, which under some 
specific situation would run into an infinity loop, consuming all CPU 
resources. This also was preventing Asterisk to terminated the call 
properly because it was waiting for the AGI to return... The 
application uses AGIspeedy to process the AGI calls, not sure if this 
can be affecting this situation somehow
Due to this problem I started looking for some option to timeout the 
AGI call and return to the dialplan after XYZ seconds and so this 
would protect Asterisk preventing the dialplan to get stuck due to 
some external script problem that is actually outside of Asterisk 
control. Does Asterisk provide some control of this sort? I searched 
around and could not find any.

Any idea is appreciated!

Kind regards
Patrick Wakano



I think this is what you may be looking for:

http://php.net/manual/en/function.set-time-limit.php

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[asterisk-users] Segfault on libasteriskpj.so.2

[Carlos Chavez]

    I just finished installing a brand new server with CentOS 7.5 and 
Asterisk 13.22.0 and the minute I a call from the PSTN (from a SIP 
trunk) bridges with any SIP phone Asterisk crashes:


Jul 20 10:59:53 localhost kernel: asterisk[2819]: segfault at 188 ip 
7f158b9e047c sp 7f1568789820 error 4 in 
libasteriskpj.so.2[7f158b976000+152000]
Jul 20 11:00:35 localhost kernel: asterisk[2925]: segfault at 188 ip 
7f44b1ea947c sp 7f700820 error 4 in 
libasteriskpj.so.2[7f44b1e3f000+152000]
Jul 20 11:02:05 localhost kernel: asterisk[3133]: segfault at 188 ip 
7fe5b8f8147c sp 7fe547737820 error 4 in 
libasteriskpj.so.2[7fe5b8f17000+152000]
Jul 20 11:08:25 localhost kernel: asterisk[3515]: segfault at 188 ip 
7f65c332547c sp 7f6551b56820 error 4 in 
libasteriskpj.so.2[7f65c32bb000+152000]


    I found a closed bug: 
https://issues.asterisk.org/jira/browse/ASTERISK-27210 but the status is 
unresolved and closed.  We are not using any DNS in our configurations, 
everything is in IP notation.  Internet is somewhat slow and sometimes 
DNS fails the first time you need to resolve something.  We started on 
13.21.1 but upgraded to 13.22.0 and are still having the same problem.


    This is not the first time I have seen this segfault.  We tried a 
virtual machine some months ago with the same result.  At the time we 
though the problem was related to the very old hypervisor we were using 
but we get the same exact segfault on a new physical server.  Anyone 
else run into this bug?  Any way to fix it or is it back to chan_sip?


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[asterisk-users] No audio on direct call from trunk to SPA-8000

[Carlos Chavez]

    I am having one of those days.  We just replaced an old Asterisk 
1.8 server with a new Asterisk 13.21.1 (both using Freepbx) and almost 
everything is working except for some incoming calls directed to a Cisco 
SPA-8000.  The PSTN trunk is SIP.  Only calls coming from the PSTN to a 
direct DID that just rings an extension on the SPA get no incoming 
audio.  All other calls, including calls from the PSTN that go through 
the main IVR or operator have audio.


    I made sure that the trunk has direct_media=no.  I checked the SPA 
configuration to make sure it is not using NAT.  Only the SPA suffers 
from this as regular SIP phones can receive calls from their DID with no 
problems.  This is the first time I use an SPA analog adapter with 
PJSIP.  They work great with chan_sip so I do not know what maybe wrong 
here.  Anyone using an SPA-8000 with PJSIP?  Any settings I should check 
on the SPA or in Asterisk?


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Re: [asterisk-users] Busy indicator for FXO line or extension

[Carlos Chavez]

On 6/28/18 5:31 AM, bilal ghayyad wrote:


Hello;

Is it possible to configure one button on the IP Phone (like Polycom 
or general SIP Phone) to indicate (at the phone display) that the line 
(the line that is connected for FXO port) is busy or not? If it is not 
busy, the user can press on the button to place outside call.


Also, is it possible to to configure another button to indicate if the 
extension is busy or not?


Regards
Bilal


    That is done through the use of "hints". 
https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints



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[asterisk-users] Setting outgoing CALLERID without changing CDR(src)

[Carlos Chavez]

    I thought I had found and answer to this question by using 
CALLERID(ani) but it seems that only works on versions prior to 12.  On 
Asterisk 13 setting CALLERID(num) before dialing to an external trunk 
always changes CDR(src) to the number you set and the original extension 
number that dialed is lost.  How are you handling this?  Am I forced to 
use a custom field to keep the original caller number?  My billing 
software uses the src field to get the extension that dialed the call.  
Any tips?


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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Carlos Chavez]

On 2/22/18 4:40 PM, Carlos Chavez wrote:


On 2/22/18 3:54 PM, Carlos Chavez wrote:


On 2/22/18 3:46 PM, Antony Stone wrote:


On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote:


On 2/22/18 1:07 PM, Antony Stone wrote:

On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:

   Usually phone companies set the outgoing CallerID for you but

recently we got control over that and are now setting the outgoing
Calleir ID ourselves.  My problem now is that the CDR will put the
outgoing CID in the CDR instead of the extension that dialed and 
that
causes problems for reports.  What is the proper way to set 
outgoing CID

and keeping the original extension number in the CDR?

Surely the CDR field "clid" is your Caller ID, whereas the CDR field
"src" is the originating extension?


Antony.

No, the src field contains the external number and the clid field has
the extension name but also the external number.

Okay, then; what do you get in the "channel" field?




    Channels contains PJSIP/-(id)

    Like I mentioned, the problem really lies in that the software 
used for call reports is coded to the "src" field. Than is why I need 
src to hace the extension number.


The solution to this problem is to set CDR(ani) to the original 
extension number before changing the outgoing callerid.  With this src 
will remain as the extension number.



Sorry, I meant CALLERID(ani).
Set(CALLERID(ani)=${CALLERID(num)})

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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Carlos Chavez]

On 2/22/18 3:54 PM, Carlos Chavez wrote:


On 2/22/18 3:46 PM, Antony Stone wrote:


On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote:


On 2/22/18 1:07 PM, Antony Stone wrote:

On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:

   Usually phone companies set the outgoing CallerID for you but

recently we got control over that and are now setting the outgoing
Calleir ID ourselves.  My problem now is that the CDR will put the
outgoing CID in the CDR instead of the extension that dialed and that
causes problems for reports.  What is the proper way to set 
outgoing CID

and keeping the original extension number in the CDR?

Surely the CDR field "clid" is your Caller ID, whereas the CDR field
"src" is the originating extension?


Antony.

No, the src field contains the external number and the clid field has
the extension name but also the external number.

Okay, then; what do you get in the "channel" field?




    Channels contains PJSIP/-(id)

    Like I mentioned, the problem really lies in that the software 
used for call reports is coded to the "src" field.  Than is why I need 
src to hace the extension number.


The solution to this problem is to set CDR(ani) to the original 
extension number before changing the outgoing callerid.  With this src 
will remain as the extension number.


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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Carlos Chavez]

On 2/22/18 3:46 PM, Antony Stone wrote:


On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote:


On 2/22/18 1:07 PM, Antony Stone wrote:

On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:

   Usually phone companies set the outgoing CallerID for you but

recently we got control over that and are now setting the outgoing
Calleir ID ourselves.  My problem now is that the CDR will put the
outgoing CID in the CDR instead of the extension that dialed and that
causes problems for reports.  What is the proper way to set outgoing CID
and keeping the original extension number in the CDR?

Surely the CDR field "clid" is your Caller ID, whereas the CDR field
"src" is the originating extension?


Antony.

No, the src field contains the external number and the clid field has
the extension name but also the external number.

Okay, then; what do you get in the "channel" field?




    Channels contains PJSIP/-(id)

    Like I mentioned, the problem really lies in that the software used 
for call reports is coded to the "src" field.  Than is why I need src to 
hace the extension number.


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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Carlos Chavez]

On 2/22/18 2:05 PM, Antony Stone wrote:


On Thursday 22 February 2018 at 20:07:47, Antony Stone wrote:


On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:

  Usually phone companies set the outgoing CallerID for you but

recently we got control over that and are now setting the outgoing
Calleir ID ourselves.  My problem now is that the CDR will put the
outgoing CID in the CDR instead of the extension that dialed and that
causes problems for reports.  What is the proper way to set outgoing CID
and keeping the original extension number in the CDR?

Surely the CDR field "clid" is your Caller ID, whereas the CDR field "src"
is the originating extension?

Another thought - if that doesn't automatically work for you (probably depends
on your dialplan / Asterisk setup), then how about setting:
CDR(accountcode)=${CALLERID(number)}
in your dialplan sometime before you set the outbound Caller ID to whatever
your PSTN number is?

Then you have the internal extension number in accountcode and the external
CallerID in clid.


Antony.

Unfortunately I use the accountcode field to put phones in different 
administrative groups and the software the company uses to make reports 
uses the src field to determine who made the call.


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Re: [asterisk-users] Set external CID but retain internal extension in CDR...

[Carlos Chavez]

On 2/22/18 1:07 PM, Antony Stone wrote:


On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:


  Usually phone companies set the outgoing CallerID for you but
recently we got control over that and are now setting the outgoing
Calleir ID ourselves.  My problem now is that the CDR will put the
outgoing CID in the CDR instead of the extension that dialed and that
causes problems for reports.  What is the proper way to set outgoing CID
and keeping the original extension number in the CDR?

Surely the CDR field "clid" is your Caller ID, whereas the CDR field "src" is
the originating extension?


Antony.

No, the src field contains the external number and the clid field has 
the extension name but also the external number.


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[asterisk-users] Set external CID but retain internal extension in CDR...

[Carlos Chavez]

    Usually phone companies set the outgoing CallerID for you but 
recently we got control over that and are now setting the outgoing 
Calleir ID ourselves.  My problem now is that the CDR will put the 
outgoing CID in the CDR instead of the extension that dialed and that 
causes problems for reports.  What is the proper way to set outgoing CID 
and keeping the original extension number in the CDR?


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[asterisk-users] Queue playing periodic_announce to agent when they answer

[Carlos Chavez]

    I have a very strange problem with my queues today.  When the agent 
answers a call they get the periodic_announce sound played to them.  I 
have a periodic_announce set to 60 seconds and the caller does hear it 
if their call is not answered.  Why would it play it to the agent?  At 
this point I even erased the periodic announcement but it is still being 
played:


[Feb 21 16:02:55] VERBOSE[25566][C-00c2] pbx.c: Executing 
[@incoming:1] Goto("PJSIP/MCM-01c0", "menu-corpo,s,1") in new stack
[Feb 21 16:02:55] VERBOSE[25566][C-00c2] pbx_builtins.c: Goto 
(menu-corpo,s,1)
[Feb 21 16:02:55] VERBOSE[25566][C-00c2] pbx.c: Executing 
[s@menu-corpo:1] Wait("PJSIP/MCM-01c0", "1") in new stack
[Feb 21 16:02:56] VERBOSE[25566][C-00c2] pbx.c: Executing 
[s@menu-corpo:2] Answer("PJSIP/MCM-01c0", "") in new stack
[Feb 21 16:02:56] VERBOSE[25566][C-00c2] pbx.c: Executing 
[s@menu-corpo:3] Set("PJSIP/MCM-01c0", "TIMEOUT(response)=5") in new 
stack
[Feb 21 16:02:56] VERBOSE[25566][C-00c2] func_timeout.c: Response 
timeout set to 5.000
[Feb 21 16:02:56] VERBOSE[25566][C-00c2] pbx.c: Executing 
[s@menu-corpo:4] Queue("PJSIP/MCM-01c0", "vacacional-corpo") in new 
stack
[Feb 21 16:02:56] VERBOSE[25566][C-00c2] res_musiconhold.c: Started 
music on hold, class 'default', on channel 'PJSIP/MCM-01c0'

[Feb 21 16:02:56] VERBOSE[25566][C-00c2] app_queue.c: Called PJSIP/1575
[Feb 21 16:02:56] VERBOSE[25566][C-00c2] app_queue.c: 
PJSIP/1575-01c1 is ringing
[Feb 21 16:03:00] VERBOSE[25566][C-00c2] app_queue.c: 
PJSIP/1575-01c1 answered PJSIP/MCM-01c0
[Feb 21 16:03:00] VERBOSE[25566][C-00c2] file.c: 
 Playing 'sullamada.slin' (language 'es')
[Feb 21 16:03:11] VERBOSE[25566][C-00c2] res_musiconhold.c: Stopped 
music on hold on PJSIP/MCM-01c0
[Feb 21 16:03:11] VERBOSE[25566][C-00c2] bridge_channel.c: Channel 
PJSIP/MCM-01c0 joined 'simple_bridge' basic-bridge 
<0559bfae-8b48-453f-8a1d-239a542edc99>
[Feb 21 16:03:14] VERBOSE[25566][C-00c2] bridge_channel.c: Channel 
PJSIP/MCM-01c0 left 'simple_bridge' basic-bridge 
<0559bfae-8b48-453f-8a1d-239a542edc99>
[Feb 21 16:03:14] VERBOSE[25566][C-00c2] pbx.c: Spawn extension 
(menu-corpo, s, 4) exited non-zero on 'PJSIP/MCM-01c0'


    We are using Realtime to configure Queues on Asterisk 13.19.1

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Re: [asterisk-users] Duplicate CDR's in Mysql

[Carlos Chavez]

On 1/14/18 4:22 PM, Mike Diehl wrote:


Hi all,

I have a problem I've not seen before. My Asterisk server stores CDR's 
via mysql, and I'm getting duplicate records.


For example:

mysql> select uniqueid,count(*) from cdr group by uniqueid having 
count(*)>1;


+--+--+

| uniqueid | count(*) |

+--+--+

| server12-1515090905.2182 | 5 |

| server12-1515091190.2215 | 3 |

+--+--+

2 rows in set (0.68 sec)

If I query for each uniqueid, I see that the records are identical. I 
have a Perl script that goes through and removes the duplicates. 
Otherwise, EVERY CDR would be duplicated.


Now, my Asterisk server was configured with multiple CDR backends, but 
I unloaded those modules. Here is what I have configured during run-time:


*CLI> cdr show status

Call Detail Record (CDR) settings

--

Logging: Enabled

Mode: Simple

Log unanswered calls: No

Log congestion: No

* Registered Backends

---

cdr-custom

Adaptive ODBC


I have exactly the same problem.  All CDRs get duplicated. Using 
ODBC and Mariadb on Asterisk 13.8


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Re: [asterisk-users] Mixmonitor with b option

[Carlos Chavez]

On 1/8/18 9:38 AM, Bertrand LUPART - Linkeo.com wrote:


Hello Carlos,



We have a server that records all calls so we set Mixmonitor with the b 
option to only record calls that are actually bridged. I notice that we have 
lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls 
that were not answered.  If a call is not answered I assume it was never 
bridged so why would Asterisk create a file?

Which version of asterisk are you running? Looks like this has been fixed some 
years ago :
https://reviewboard.asterisk.org/r/2068/diff/



Is there a way to avoid getting those empty files?  It makes finding recordings 
vey slow when there are hundreds of non relevant files in the monitor directory.

You could run a cron job that would periodically delete those 44 bytes files


Dispatching audio files in subdirectories may help performance-wise, for 
example :

  same => 
n,MixMonitor(/absolute/path/${STRFTIME(${EPOCH},,%Y)}/${STRFTIME(${EPOCH},,%m)}/${STRFTIME(${EPOCH},,%d)}/${STRFTIME(${EPOCH},,%Y-%m-%d)}-${CALLERID(num)}-${EXTEN}-${UNIQUEID}.gsm)


OTH

We are running 13.8.4 at the moment which was the latest when 
deployed.  I guess the patch never made it to the trunk.  The problem 
with running a script to cleanup is that you may drag files that are 
open at the moment but still at 44 bytes because they are still waiting 
to be bridged.  I may start using subdirectories as you mention but that 
means I will still have lots of empty files to deal with.


Thanks.

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[asterisk-users] Mixmonitor with b option

[Carlos Chavez]

We have a server that records all calls so we set Mixmonitor with 
the b option to only record calls that are actually bridged. I notice 
that we have lost of 44 byte files in /var/spool/asterisk/monitor which 
correspond to calls that were not answered.  If a call is not answered I 
assume it was never bridged so why would Asterisk create a file?  Is 
there a way to avoid getting those empty files?  It makes finding 
recordings vey slow when there are hundreds of non relevant files in the 
monitor directory.



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Re: [asterisk-users] PJSIP Trunk 401 Unauthorized (Alestra Mexico)

[Carlos Chavez]

On 12/2/17 4:40 PM, Joshua Colp wrote:


On Sat, Dec 2, 2017, at 06:33 PM, Carlos Chavez wrote:

  I am having a really bad day trying to get incoming calls to work
on Asterisk 13 with PJSIP.  We just migrated from Asterisk 1.8 where
everything was working but there seems that something got lost in
translation.  No matter what I try I always get a 401 Unauthorized
message when receiving a call from the PSTN provider.  I can make calls
and the registration is working.  I have tried to set the identify to an
endpoint that does not have an auth defined.  Anyone using Alestra SIP
trunks in Mexico?




My identify is:

=
   endpoint  : Alestra
   match : 200.94.59.150/255.255.255.255
   match_header  :
   srv_lookups   : true


It does not matter if I use the original endpoint or an endpoint with no
auth.  Asterisk will still reject the call.  Any tips? How can I make
sure that the identify is being used?

If you turn up the core debug to level 4 and send it to the console it
will tell you what it is doing. I'd also suggest providing the endpoint
definition, and confirming it was loaded as expected. If it's not then
you can look at the Asterisk console at load time and it will tell you
what it did not like.

Thank you for your help.  It turns out that I did not notice that 
the register for the trunk had the endpoint defined there and therefore 
the identify section is ignored.  I never got a message that the IP 
matched because of this.  I need to keep this in mind for future reference.


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[asterisk-users] PJSIP Trunk 401 Unauthorized (Alestra Mexico)

[Carlos Chavez]

    I am having a really bad day trying to get incoming calls to work 
on Asterisk 13 with PJSIP.  We just migrated from Asterisk 1.8 where 
everything was working but there seems that something got lost in 
translation.  No matter what I try I always get a 401 Unauthorized 
message when receiving a call from the PSTN provider.  I can make calls 
and the registration is working.  I have tried to set the identify to an 
endpoint that does not have an auth defined.  Anyone using Alestra SIP 
trunks in Mexico?


Here is what I get on the cli:

<--- Received SIP request (1092 bytes) from UDP:200.94.59.150:5060 --->
INVITE sip:5547371...@xxx.xxx.xxx.xxx:5060;line=qooanvj SIP/2.0
Via: SIP/2.0/UDP 200.94.59.150:5060;branch=z9hG4bKnvnkof007gngrp80d2g1.1
From: 
;tag=866455524-1512253376938-

To: "MEXICO USERNAME"
Call-ID: BW1622569380212171700499694@10.6.30.9
CSeq: 212444374 INVITE
Contact: 
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Min-SE: 90
Session-Expires: 900;refresher=uac
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 287

v=0
o=BroadWorks 26026640 1 IN IP4 200.94.59.152
s=-
c=IN IP4 200.94.59.152
t=0 0
m=audio 5470 RTP/AVP 18 0 8 100
a=rtpmap:18 G729/8000
a=fmtp:18 annexb:no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:40

<--- Transmitting SIP response (588 bytes) to UDP:200.94.59.150:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
200.94.59.150:5060;received=200.94.59.150;branch=z9hG4bKnvnkof007gngrp80d2g1.1

Call-ID: BW1622569380212171700499694@10.6.30.9
From: 
;tag=866455524-1512253376938-
To: "MEXICO USERNAME" 
;tag=z9hG4bKnvnkof007gngrp80d2g1.1

CSeq: 212444374 INVITE
WWW-Authenticate: Digest 
realm="asterisk",nonce="1512253376/546618e0645f233990bd70d97691ddba",opaque="3b5f610b33037ba2",algorithm=md5,qop="auth"

Server: Asterisk PBX 13.18.3
Content-Length:  0


<--- Received SIP request (434 bytes) from UDP:200.94.59.150:5060 --->
ACK sip:5547371...@xxx.xxx.xxx.xxx:5060;line=qooanvj SIP/2.0
Via: SIP/2.0/UDP 200.94.59.150:5060;branch=z9hG4bKnvnkof007gngrp80d2g1.1
CSeq: 212444374 ACK
From: 
;tag=866455524-1512253376938-
To: "MEXICO 
USERNAME";tag=z9hG4bKnvnkof007gngrp80d2g1.1

Call-ID: BW1622569380212171700499694@10.6.30.9
Max-Forwards: 9
Content-Length: 0


My identify is:

=
 endpoint  : Alestra
 match : 200.94.59.150/255.255.255.255
 match_header  :
 srv_lookups   : true


It does not matter if I use the original endpoint or an endpoint with no 
auth.  Asterisk will still reject the call.  Any tips? How can I make 
sure that the identify is being used?


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Re: [asterisk-users] Confbridge SFU for Asterisk 15

[Carlos Chavez]

On 11/15/17 11:36 AM, Joshua Colp wrote:


On Wed, Nov 15, 2017, at 01:30 PM, Carlos Chavez wrote:




Here is more information from the browser about the session:
https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF

On Asterisk I have icesupport=true in rtp.conf and ice_support=yes on the
endpoint.  I have configured a STUN server in both rtp.conf and
res_stun_monitor.conf

What is the exact network topology? Is Asterisk behind NAT as well with
ports forwarded? If so you should configure a mapping in rtp.conf so
that the internal IP address is mapped to its external IP address in the
ICE candidates, giving a better chance that things will work.

Correct, Asterisk is behind a NAT with SIP and RTP forwarded to the 
server.  We have a static IP address set in the transport definition.  I 
will map the port in rtp.conf and try it again.


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Re: [asterisk-users] Confbridge SFU for Asterisk 15

[Carlos Chavez]

On 11/15/17 11:10 AM, Joshua Colp wrote:


On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote:

On 11/14/17 5:23 PM, Joshua Colp wrote:


On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote:

Trace with 3 clients.  We can hear each other but no video.

https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz

Do you see anything in the Javascript console of the browser? We are
adding the needed media streams by sending a reinvite to the
participants but we don't get any response, which means for some reason
the browser may not have liked what we provided.


This is what I get on the console:
new session - outgoing - [object Object]
cyber_mega_phone.js:78:3
ontrack: audio - 8b7fca5e-bb67-4e8c-8bdb-84fb80ac4cc0 stream
66e4250b-c196-4482-a347-d12772ef865d
cyber_mega_phone.js:111:4
Streams: added 66e4250b-c196-4482-a347-d12772ef865d
cyber_mega_phone.js:225:3
ontrack: video - ad836e20-c0c9-423f-9c42-0aef19c5ca32 stream
66e4250b-c196-4482-a347-d12772ef865d
cyber_mega_phone.js:111:4
confirmed: adding local stream {8bafb537-864a-424b-b5d3-d13ee0b60f8c}
cyber_mega_phone.js:84:5
Streams: added {8bafb537-864a-424b-b5d3-d13ee0b60f8c}
cyber_mega_phone.js:225:3
RTCPeerConnection.getLocalStreams/getRemoteStreams are deprecated. Use
RTCPeerConnection.getSenders/getReceivers instead.
cyber_mega_phone.js:82:17
ICE failed, add a STUN server and see about:webrtc for more details

Looks like for some reason it failed to successfully do ICE negotiation
potentially on the newly added remote streams. Why that is is
environment specific - but the problem does seem to be on the web
browser/client side, not in Asterisk itself. You'd need to figure out
why.

This is one of the annoyances of WebRTC - the browser can be a black box
at time and when things go wrong (like this) it's hard to dig and figure
out what is up.


Here is more information from the browser about the session:
https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF

On Asterisk I have icesupport=true in rtp.conf and ice_support=yes on the 
endpoint.  I have configured a STUN server in both rtp.conf and 
res_stun_monitor.conf


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Re: [asterisk-users] Confbridge SFU for Asterisk 15

[Carlos Chavez]

On 11/14/17 5:23 PM, Joshua Colp wrote:


On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote:

Trace with 3 clients.  We can hear each other but no video.

https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz

Do you see anything in the Javascript console of the browser? We are
adding the needed media streams by sending a reinvite to the
participants but we don't get any response, which means for some reason
the browser may not have liked what we provided.


This is what I get on the console:
new session - outgoing - [object Object]
cyber_mega_phone.js:78:3
ontrack: audio - 8b7fca5e-bb67-4e8c-8bdb-84fb80ac4cc0 stream 
66e4250b-c196-4482-a347-d12772ef865d

cyber_mega_phone.js:111:4
Streams: added 66e4250b-c196-4482-a347-d12772ef865d
cyber_mega_phone.js:225:3
ontrack: video - ad836e20-c0c9-423f-9c42-0aef19c5ca32 stream 
66e4250b-c196-4482-a347-d12772ef865d

cyber_mega_phone.js:111:4
confirmed: adding local stream {8bafb537-864a-424b-b5d3-d13ee0b60f8c}
cyber_mega_phone.js:84:5
Streams: added {8bafb537-864a-424b-b5d3-d13ee0b60f8c}
cyber_mega_phone.js:225:3
RTCPeerConnection.getLocalStreams/getRemoteStreams are deprecated. Use 
RTCPeerConnection.getSenders/getReceivers instead.

cyber_mega_phone.js:82:17
ICE failed, add a STUN server and see about:webrtc for more details

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Re: [asterisk-users] Confbridge SFU for Asterisk 15

[Carlos Chavez]

Trace with 3 clients.  We can hear each other but no video.

https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz


On 11/14/17 5:06 PM, Joshua Colp wrote:

On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote:

On 11/14/17 4:27 PM, Joshua Colp wrote:


On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote:

On 11/14/17 3:55 PM, Joshua Colp wrote:


On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote:

I followed the blog post and I can get video from the conference if
I configure the bridge as follow_talker so I know everything is working
on the pjsip side.  The only problem is that video_mode = sfu is
apparently not valid in either confbridge.conf or via the dialplan and I
get no video with that option.

The option, when set, will show up as "no video" if you do the
"confbridge show" as you mentioned. That's a bug which is why I
mentioned filing an issue. It is still valid though.

Have you confirmed that the maximum number of streams is set using
"pjsip show endpoint"? and that the codecs are correct?


allow  : (ulaw|vp8|h264)
max_audio_streams  : 10
max_video_streams  : 10

   Video and audio work fine if I use follow_talker in the
confbridge.  No video when set to sfu.

What browser are you trying from? Can you provide a SIP trace (pjsip set
logger on)? And what is the output of "core show channel" for each
channel when they are in the video conference bridge?


  We have tried with Firefox (56) and Chrome 61.0.3163.100 on both
Windows and OSX.

SIP trace:
https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/GsXHb9EoRUZuJrZ
Channels:
https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/9W04VCUFQSfVumW

  It appears that the CBAnn channels only have audio a no video.

Those are the announcer channels for playing audio into the conference
bridge. I need to see an attempt with more than 1 participant in the
bridge.



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Re: [asterisk-users] Confbridge SFU for Asterisk 15

[Carlos Chavez]

On 11/14/17 4:27 PM, Joshua Colp wrote:


On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote:

On 11/14/17 3:55 PM, Joshua Colp wrote:


On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote:

   I followed the blog post and I can get video from the conference if
I configure the bridge as follow_talker so I know everything is working
on the pjsip side.  The only problem is that video_mode = sfu is
apparently not valid in either confbridge.conf or via the dialplan and I
get no video with that option.

The option, when set, will show up as "no video" if you do the
"confbridge show" as you mentioned. That's a bug which is why I
mentioned filing an issue. It is still valid though.

Have you confirmed that the maximum number of streams is set using
"pjsip show endpoint"? and that the codecs are correct?


allow  : (ulaw|vp8|h264)
max_audio_streams  : 10
max_video_streams  : 10

  Video and audio work fine if I use follow_talker in the
confbridge.  No video when set to sfu.

What browser are you trying from? Can you provide a SIP trace (pjsip set
logger on)? And what is the output of "core show channel" for each
channel when they are in the video conference bridge?

We have tried with Firefox (56) and Chrome 61.0.3163.100 on both 
Windows and OSX.


SIP trace: 
https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/GsXHb9EoRUZuJrZ
Channels: 
https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/9W04VCUFQSfVumW


It appears that the CBAnn channels only have audio a no video.

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Re: [asterisk-users] Confbridge SFU for Asterisk 15

[Carlos Chavez]

On 11/14/17 3:55 PM, Joshua Colp wrote:


On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote:

  I followed the blog post and I can get video from the conference if
I configure the bridge as follow_talker so I know everything is working
on the pjsip side.  The only problem is that video_mode = sfu is
apparently not valid in either confbridge.conf or via the dialplan and I
get no video with that option.

The option, when set, will show up as "no video" if you do the
"confbridge show" as you mentioned. That's a bug which is why I
mentioned filing an issue. It is still valid though.

Have you confirmed that the maximum number of streams is set using
"pjsip show endpoint"? and that the codecs are correct?


allow  : (ulaw|vp8|h264)
max_audio_streams  : 10
max_video_streams  : 10

Video and audio work fine if I use follow_talker in the 
confbridge.  No video when set to sfu.


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Re: [asterisk-users] Confbridge SFU for Asterisk 15

[Carlos Chavez]

On 11/14/17 3:38 PM, Joshua Colp wrote:


On Tue, Nov 14, 2017, at 05:23 PM, Carlos Chavez wrote:

  I am trying to get the "Mega Phone" demo working on my office PBX
but there seems to be a problem when trying to set the default bridge to
sfu mode.  I have the following configuration in confbridge.conf in the
default_bridge section: video_mode = sfu but when I do a "confbridge
show profile bridge default_bridge" I see:

Video Mode:   no video

  I can change it to follow_talker, last_marked or first_marked and
it does change, it is just the sfu option that does not seem to be
valid.  I am using Asterisk 15.1.2 for my testing.  I even tried to
force the option via Dialplan:

[ Context 'ext' created by 'pbx_config' ]
'1000' => 1. Answer()
[extensions.conf:0]
  2. Set(CONFBRIDGE(bridge,video_mode)=sfu)
[extensions.conf:0]
  3. ConfBridge(guest) [extensions.conf:0]
  4. Hangup() [extensions.conf:0]

  But I get no video at all on the conference.

  Any ideas?

The CLI command doesn't have sfu as a mapping right now. Please file an
issue[1] for that. As for video there's additional configuration that
you have to do in pjsip.conf in order to allow the streams. I'd suggest
following the blog post[2] which has known working configuration.

[1] https://issues.asterisk.org/jira
[2]
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/

I followed the blog post and I can get video from the conference if 
I configure the bridge as follow_talker so I know everything is working 
on the pjsip side.  The only problem is that video_mode = sfu is 
apparently not valid in either confbridge.conf or via the dialplan and I 
get no video with that option.


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[asterisk-users] Confbridge SFU for Asterisk 15

[Carlos Chavez]

I am trying to get the "Mega Phone" demo working on my office PBX 
but there seems to be a problem when trying to set the default bridge to 
sfu mode.  I have the following configuration in confbridge.conf in the 
default_bridge section: video_mode = sfu but when I do a "confbridge 
show profile bridge default_bridge" I see:


Video Mode:   no video

I can change it to follow_talker, last_marked or first_marked and 
it does change, it is just the sfu option that does not seem to be 
valid.  I am using Asterisk 15.1.2 for my testing.  I even tried to 
force the option via Dialplan:


[ Context 'ext' created by 'pbx_config' ]
  '1000' => 1. Answer()   
[extensions.conf:0]
2. Set(CONFBRIDGE(bridge,video_mode)=sfu) 
[extensions.conf:0]

3. ConfBridge(guest) [extensions.conf:0]
4. Hangup() [extensions.conf:0]

But I get no video at all on the conference.

Any ideas?



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[asterisk-users] Asterisk 13.8 compile error

[Carlos Chavez]

I just tried to compile the latest Asterisk 13.8.0 and it stopped 
with several errors on pjsip.  So FYI if you run the install_prereq 
script and then use ./configure --with-pjproject-bundled you will have 
the same problem because the prereq script installs an older version of 
pjproject.  Make sure to uninstall pjproject-2.3-6.el7.x86_64 and 
pjproject-devel-2.3-6.el7.x86_64 before compiling asterisk.  They really 
should remove this from the script but I guess they will do so in 15.



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Re: [asterisk-users] PJSIP trunk to Telynx

[Carlos Chavez]

On 10/20/2017 8:46 PM, Joshua Colp wrote:


On Fri, Oct 20, 2017, at 10:17 PM, Carlos Chavez wrote:

Has anyone used Telynx as a SIP trunk provider?  It works with chan_sip
but it I seem to be having problems trying to set up a PJSIP trunk.  I
always get a 401 Unauthorized when they send me a call.  I know my
username and password are correct since I can register and PJSIP uses
the same information for inbound as for the registration.  Unfortunately
their support department said "PJSIP what?".  It seems mos SIP providers
know Asterisk but are not aware of the important change coming.  I
already got a nasty surprise from Voicepulse stating that they do not
support PJSIP so their service will not work with newer installations.

Generally ITSPs don't authenticate to you, they expect the device or
software to just know the call is from them and to accept it. In PJSIP
this is done by using an identify section and matching based on IP
address. There's also the line option[1] to outbound registration which
works with some equipment, if it works then no identify section is
required.

    Thank you.  I forgot that little detail.  I just changed my 
identify section to use an endpoint that does not have an auth section 
and only uses the IP in the AOR section.  It is working now.



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[asterisk-users] PJSIP trunk to Telynx

[Carlos Chavez]

Has anyone used Telynx as a SIP trunk provider?  It works with chan_sip 
but it I seem to be having problems trying to set up a PJSIP trunk.  I 
always get a 401 Unauthorized when they send me a call.  I know my 
username and password are correct since I can register and PJSIP uses 
the same information for inbound as for the registration.  Unfortunately 
their support department said "PJSIP what?".  It seems mos SIP providers 
know Asterisk but are not aware of the important change coming.  I 
already got a nasty surprise from Voicepulse stating that they do not 
support PJSIP so their service will not work with newer installations.


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Re: [asterisk-users] speech-recog.agi

[Carlos Chavez]

On 10/19/17 3:53 PM, Jonathan H wrote:


That's because it uses a deprecated API and endpoint.

However, funny you should ask this, because I've just finished
updating my Google TTS routine to take advantage of the new
streamlined API.

If you can wait a couple of days, I've stick it up on the repo -
BUT... it's going to require python3.5+, the way I do it...

Would that work for you?

Thank you.  I just realized that I just needed to RTFM.  There is a 
new version of the speech-recog agi that uses the Cloud Speech API.  It 
is listed on the web site almost at the end. That one works perfectly.  
Of course alternatives are always welcome.


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[asterisk-users] speech-recog.agi

[Carlos Chavez]

I want to try using google for speech recognition in Asterisk and I 
found a ready made AGI:


http://zaf.github.io/asterisk-speech-recog/

I have followed all the steps listed in the web site but I keep 
getting this error:


AGI Tx >> 200 result=99981 (timeout) endpos=22720
AGI Rx << VERBOSE "Unable to get recognition data." 3

I made sure all the dependencies are met and that my API key for 
Google Cloud Speech is correct (cut and paste).  Any pointers to get 
this to work or any other quick waysto start using Google for speech 
recognition in Asterisk?  Thanks.



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[asterisk-users] Cepstral, Swift and Asterisk 13

[Carlos Chavez]

Anyone here have a working app_swift with Asterisk 13?  I purchased 
my licenses and followed their install procedure but I do not get any 
audio when I dial a test.  Stranger still is that I can get audio on a 
softphone (Bria) but nowhere else.  I have tried several desk phones and 
softphones and only Bria can get some distorded audio, all others are 
just silent.


I get no error on the console so everything seems to be installed 
properly.  Any pointers?



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[asterisk-users] Confbridge GUI?

[Carlos Chavez]

I have a very old server that is used only for conferences on 
Meetme.  To manage the conference rooms we use Web Meetme.  Now it is 
time to upgrade everything but since Meetme is no longer available I 
need to find a replacement GUI to manage the conference rooms.  Anyone 
know a solution that works with Confbridge?  I found "Asterisk 
Confbridge Manager" from a russian company but it does not let you 
schedule conference rooms that only last a certain time.



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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[Carlos Chavez]

On 2017-08-02 07:08, Nathan Anderson wrote:

Richard Kenner wrote:


But the question here
was *Asterisk*, not kernels.  User-level code has *way* fewer
dependencies.


*Precisely*.  Unless we're talking DAHDI here (which we're not), Linux
& ESXi are red herrings.

Carlos Chavez wrote:


  I am having a very tough time trying to replace an Elastix 2.X
install running as a virtual machine on ESXI 4.


There's no way this has anything to do with ESXi or the version of it
that you are running.  Zero.  Zip.  Zilch.

If you want to prove this to yourself and others, take the *exact*
same binary bits, install them bare-metal on another piece of
hardware, run the same traffic through it, and watch it crash and burn
in the same way.  The only way that I can see this playing out
differently is if the bug (yes, bug) in Asterisk and associated
libraries is extremely timing-dependent, and running it in a VM is
exposing this bug in a way that most bare-metal installations
wouldn't.


I will try using chan_sip
instead of PJSIP to get things running but confidence is not high.


Given that the log entry you pasted into your e-mail references
"libasteriskpj.so", I'd bet $$$ that switching to chan_sip has an
extremely high likelihood of working, assuming that your set-up has no
particular dependencies on PJSIP-specific features that you have to
work around (and if you are migrating from an Asterisk 1.6
installation, I'm guessing it doesn't).

Best of luck,

-- Nathan


I run the CentOS 7.3 / Asterisk 13.17.0 combination of software 
installed from the same sources on multiple servers across a wide 
variety of hardware (both metal and virtual) and this is the only place 
that I have encountered this particular problem.  That is why the only 
variable left is the version of ESXI as newer versions work.  
Unfortunately I do not have a newer server where I can just import this 
same VM to completely eliminate the possibility.


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Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

[Carlos Chavez]

On 2017-08-01 15:48, Doug Lytle wrote:

I am having a very tough time trying to replace an Elastix 2.X
install running as a virtual machine on ESXI 4


Licensed or free ESXI?

I want to say your version is too old.  I'm currently running ESXI 6.0
update 3 at home and Asterisk in a VM under debian without issue.

Doug


The version is licensed and the customer does not want to invest on new 
hardware/software at the moment.  If the ESXI version is too old I need 
to give them definitive proof that the segfaults are caused by that but 
since the old elastix has been running there for years they do not quite 
believe it.


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[asterisk-users] Asterisk 13 on old VMware ESXI 4

[Carlos Chavez]

 I am having a very tough time trying to replace an Elastix 2.X 
install running as a virtual machine on ESXI 4.  I tried using the 
Freepbx 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I 
keep getting random segfaults:


[175711.476685] asterisk[2942]: segfault at 188 ip 7fc6c41abffc sp 
7fc608575890 error 4 in libasteriskpj.so.2[7fc6c4144000+14c000]


 I then proceeded to install a CentOS 7.3 VM and compiled Asterisk 
13.17.0 by hand.  We are still using Freepbx 14 for the front end.  We 
did some testing over the weekend and calls were coming in and out and 
all extensions were registered.  Come Monday Asterisk started 
segfaulting again with exactly the same error.  Maybe VMware is too old 
to support the newer CentOS and Asterisk?  The Elastix install is based 
on CentOS 5 and Asterisk 1.6.  I have no idea how to approach this.  It 
only segfaults when there are only more than a couple simultaneous 
calls, that is why testing with only a couple of calls worked.


 There are several core dump files but I really do not know how to 
use them for debugging Asterisk.  Any ideas?  I will try using chan_sip 
instead of PJSIP to get things running but confidence is not high.


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[asterisk-users] Asterisk install_prereq

[Carlos Chavez]

 Is there a reason for Asterisk 13.17.0 to download and install 
pjproject-devel-2.3-6.el7.x86_64.rpm when you run the install_prereq 
script?  Since most people will compile asterisk using the bundled 
version of pjproject this may cause confusion.  And it is also an older 
version than the one used by the bundled option.


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Re: [asterisk-users] Asterisk 13.16.0 segfault

[Carlos Chavez]

On 7/20/17 8:47 AM, Marcelo Terres wrote:

Which version of Asterisk are you using? Are you compiling it with the 
bundle pjproject ?


--with-pjproject-bundled

Regards,

Marcelo H. Terres <mhter...@gmail.com <mailto:mhter...@gmail.com>>
IM: mhter...@jabber.mundoopensource.com.br 
<mailto:mhter...@jabber.mundoopensource.com.br>

https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On 19 July 2017 at 17:03, Carlos Chavez <cur...@telecomab.mx 
<mailto:cur...@telecomab.mx>> wrote:


On 7/19/17 2:37 AM, Marcelo Terres wrote:


This is the pjsip library.

Is it possible to you to update pjsip for the latest version to
test if it solves the problem?

On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cur...@telecomab.mx
<mailto:cur...@telecomab.mx>> wrote:

I am getting frequent segfaults on a new Asterisk
installation.  So far the only message I see is:

Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault
at 188 ip 7fb2d535723f sp 7fb25a11b5c0 error 4 in
libasteriskpj.so.2[7fb2d52e5000+18]
Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault
at 188 ip 7f4afea0c23f sp 7f4a7f7e35c0 error 4 in
libasteriskpj.so.2[7f4afe99a000+18]
Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault
at 188 ip 7f2eb611923f sp 7f2e3aec25c0 error 4 in
libasteriskpj.so.2[7f2eb60a7000+18]
Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault
at 188 ip 7fc5758dd23f sp 7fc4fa6245c0 error 4 in
libasteriskpj.so.2[7fc57586b000+18]
Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault
at 188 ip 7f5f29abb23f sp 7f5eae8285c0 error 4 in
libasteriskpj.so.2[7f5f29a49000+18]

Since this is a Freepbx distro does could the problem be
related to their flavor of Asterisk? I have several other
plain Asterisk servers running on this version without any
problems. Any recommendations on how to debug this?


My solution to this is going to be compiling Asterisk manually
instead of using their pre packaged version as debugging will take
a lot more time.



The Freepbx distro still uses a separate pjproject as far as I 
know.  When I compile my own I always use the bundled version now.


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Re: [asterisk-users] Asterisk 13.16.0 segfault

[Carlos Chavez]

On 7/19/17 2:37 AM, Marcelo Terres wrote:


This is the pjsip library.

Is it possible to you to update pjsip for the latest version to test 
if it solves the problem?


On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cur...@telecomab.mx 
<mailto:cur...@telecomab.mx>> wrote:


I am getting frequent segfaults on a new Asterisk installation. So
far the only message I see is:

Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188
ip 7fb2d535723f sp 7fb25a11b5c0 error 4 in
libasteriskpj.so.2[7fb2d52e5000+18]
Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188
ip 7f4afea0c23f sp 7f4a7f7e35c0 error 4 in
libasteriskpj.so.2[7f4afe99a000+18]
Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault at 188
ip 7f2eb611923f sp 7f2e3aec25c0 error 4 in
libasteriskpj.so.2[7f2eb60a7000+18]
Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault at 188
ip 7fc5758dd23f sp 7fc4fa6245c0 error 4 in
libasteriskpj.so.2[7fc57586b000+18]
Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault at 188
ip 7f5f29abb23f sp 7f5eae8285c0 error 4 in
libasteriskpj.so.2[7f5f29a49000+18]

Since this is a Freepbx distro does could the problem be related
to their flavor of Asterisk?  I have several other plain Asterisk
servers running on this version without any problems.  Any
recommendations on how to debug this?

My solution to this is going to be compiling Asterisk manually 
instead of using their pre packaged version as debugging will take a lot 
more time.


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[asterisk-users] Asterisk 13.16.0 segfault

[Carlos Chavez]

I am getting frequent segfaults on a new Asterisk installation.  So far 
the only message I see is:


Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 
7fb2d535723f sp 7fb25a11b5c0 error 4 in 
libasteriskpj.so.2[7fb2d52e5000+18]
Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip 
7f4afea0c23f sp 7f4a7f7e35c0 error 4 in 
libasteriskpj.so.2[7f4afe99a000+18]
Jul 18 09:22:57 pbxbogota kernel: asterisk[28471]: segfault at 188 ip 
7f2eb611923f sp 7f2e3aec25c0 error 4 in 
libasteriskpj.so.2[7f2eb60a7000+18]
Jul 18 09:25:49 pbxbogota kernel: asterisk[28949]: segfault at 188 ip 
7fc5758dd23f sp 7fc4fa6245c0 error 4 in 
libasteriskpj.so.2[7fc57586b000+18]
Jul 18 09:31:17 pbxbogota kernel: asterisk[29203]: segfault at 188 ip 
7f5f29abb23f sp 7f5eae8285c0 error 4 in 
libasteriskpj.so.2[7f5f29a49000+18]


Since this is a Freepbx distro does could the problem be related to 
their flavor of Asterisk?  I have several other plain Asterisk servers 
running on this version without any problems.  Any recommendations on 
how to debug this?


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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/20/17 2:37 PM, Kseniya Blashchuk wrote:

If SIP goes to the same provider then yes. Still I would check a 
packet capture for better understanding. BTW, did you try iax debug?


чт, 20 апр. 2017 г. в 19:46, Carlos Chavez <cur...@telecomab.mx 
<mailto:cur...@telecomab.mx>>:


On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:


Can it happen that the routes lead the traffic through another
interface? Did you try a packet capture with tcpdump? Do the
packets really leave the usb adapter? Can asymmetric routing be
in effect?
Maybe there were some static routes that disappeared when the
adapter was unplugged...


On Thu, Apr 20, 2017, 12:41 AM Antony Stone
<antony.st...@asterisk.open.source.it
<mailto:antony.st...@asterisk.open.source.it>> wrote:

On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:

> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then
reconnected...
> >
> > 1. What's the name of the network device for this
adapter?  Is it the
> > same name as it previously had?
> >
> > 2. What does 'ifconfig' say the IP address is for this
adapter?
> >
> > 3. What do you have in /etc/asterisk/iax.conf for
'bindaddr' and
> > 'bindport'?
> >
> > 4. Do you have SIP connections on the same network
interface, and are
> > those working as normal?
> >
> >
> > Antony.
>
> 1- No changes to device names.  eth0 is the main link to
the network,
> eth1 (also internal) goes to a SIP provider and eth2 (the
USB adapter)
> goes to another SIP provider.  All IAX trunks use eth0
>
> 2- ifconfig gives the proper IP and netmask for all interfaces
>
> 3- We do not specify bindaddr or bindport in the config
file as the
> default is to bind to 0.0.0.0
>
> 4- We had to make new SIP trunks to replace the IAX2 trunks
to all
> servers.  The SIP trunk is working with no problems. 
Except for two SIP

> links to PSTN all internal extensions use the same network
interface.

Ugh :(

Sorry, I have no more ideas, then.

I hope someone else comes into this thread with a helpful
suggestion.


If routing was the problem then the SIP trunk would not work. 
Usually IAX2 is a little more forgiving about routing than SIP.


The new SIP trunks are replacing the IAX2 trunks to our other 
Asterisk servers and use exactly the same network paths, that is why I 
know it is not a network infrastructure issue.  We did turn on IAX debug 
and we only se the server trying to poke the other servers but there is 
not response or any incoming traffic.


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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/20/17 12:45 AM, Kseniya Blashchuk wrote:

Can it happen that the routes lead the traffic through another 
interface? Did you try a packet capture with tcpdump? Do the packets 
really leave the usb adapter? Can asymmetric routing be in effect?
Maybe there were some static routes that disappeared when the adapter 
was unplugged...



On Thu, Apr 20, 2017, 12:41 AM Antony Stone 
<antony.st...@asterisk.open.source.it 
<mailto:antony.st...@asterisk.open.source.it>> wrote:


On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:

> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then
reconnected...
> >
> > 1. What's the name of the network device for this adapter?  Is
it the
> > same name as it previously had?
> >
> > 2. What does 'ifconfig' say the IP address is for this adapter?
> >
> > 3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and
> > 'bindport'?
> >
> > 4. Do you have SIP connections on the same network interface,
and are
> > those working as normal?
> >
> >
> > Antony.
>
> 1- No changes to device names.  eth0 is the main link to the
network,
> eth1 (also internal) goes to a SIP provider and eth2 (the USB
adapter)
> goes to another SIP provider.  All IAX trunks use eth0
>
> 2- ifconfig gives the proper IP and netmask for all interfaces
>
> 3- We do not specify bindaddr or bindport in the config file as the
> default is to bind to 0.0.0.0
>
> 4- We had to make new SIP trunks to replace the IAX2 trunks to all
> servers.  The SIP trunk is working with no problems. Except for
two SIP
> links to PSTN all internal extensions use the same network
interface.

Ugh :(

Sorry, I have no more ideas, then.

I hope someone else comes into this thread with a helpful suggestion.


If routing was the problem then the SIP trunk would not work. 
Usually IAX2 is a little more forgiving about routing than SIP.


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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/19/17 4:23 PM, Antony Stone wrote:


On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote:


On 4/19/17 4:09 PM, Antony Stone wrote:

On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:

I have a server that had been operating for a few years now with

IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks
now say UNREACHABLE.

...snip...


So far the only thing different is that the receive queue for port 4569
is not zero like all the other servers:

udp   128760  0 0.0.0.0:45690.0.0.0:*

Basically all packets for IAX2 are getting stuck in the queue.
Any

suggestions?

Have you tried rebooting the router which connects this machine to the
Internet?

It sounds like a stale connection-tracking table entry to me.


Antony.

  We have already tried that.  One of the servers that has an IAX
trunk to this server is on the same local network so that eliminates any
firewall/router in the way.  We disabled iptables just in case too.

Hm :(

You say the USB ethernet adapter got unplugged and then reconnected...

1. What's the name of the network device for this adapter?  Is it the same
name as it previously had?

2. What does 'ifconfig' say the IP address is for this adapter?

3. What do you have in /etc/asterisk/iax.conf for 'bindaddr' and 'bindport'?

4. Do you have SIP connections on the same network interface, and are those
working as normal?


Antony.

1- No changes to device names.  eth0 is the main link to the network, 
eth1 (also internal) goes to a SIP provider and eth2 (the USB adapter) 
goes to another SIP provider.  All IAX trunks use eth0


2- ifconfig gives the proper IP and netmask for all interfaces

3- We do not specify bindaddr or bindport in the config file as the 
default is to bind to 0.0.0.0


4- We had to make new SIP trunks to replace the IAX2 trunks to all 
servers.  The SIP trunk is working with no problems.  Except for two SIP 
links to PSTN all internal extensions use the same network interface.


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Re: [asterisk-users] IAX2 getting stuck

[Carlos Chavez]

On 4/19/17 4:09 PM, Antony Stone wrote:


On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:


   I have a server that had been operating for a few years now with
IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks
now say UNREACHABLE.

...snip...


So far the only thing different is that the receive queue for port 4569 is
not zero like all the other servers:

udp   128760  0 0.0.0.0:45690.0.0.0:*

   Basically all packets for IAX2 are getting stuck in the queue. Any
suggestions?

Have you tried rebooting the router which connects this machine to the
Internet?

It sounds like a stale connection-tracking table entry to me.


Antony.

We have already tried that.  One of the servers that has an IAX 
trunk to this server is on the same local network so that eliminates any 
firewall/router in the way.  We disabled iptables just in case too.


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[asterisk-users] IAX2 getting stuck

[Carlos Chavez]

 I have a server that had been operating for a few years now with 
IAX2 trunks to several other servers.  Since yesterday all IAX2 trunks 
now say UNREACHABLE.  No configuration changes have been made and no 
upgrades have been done.  The server is running 11.16.0 (yes, we are 
planning upgrades but lets solve this first).  The server has a USB 
ethernet adapter that got disconnected yesterday by accident and since 
it was reconnected we started having the problem.  We have rebooted many 
times and even replaced the USB adapter but still the same issue, and 
the USB adapter is not the main ethernet link to the local network, it 
is just for a link to a provider.  So far the only thing different is 
that the receive queue for port 4569 is not zero like all the other servers:


udp   128760  0 0.0.0.0:45690.0.0.0:*

 Basically all packets for IAX2 are getting stuck in the queue. Any 
suggestions?


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Re: [asterisk-users] Triggering an AGI script when a queued call is answered

[Carlos Chavez]

On 11/24/16 11:20 AM, A J Stiles wrote:


Many years ago, I used to have an AGI script that fired on an incoming call,
did some database lookups and ended up raising a notification on the screen of
the person whose phone was ringing, with the details looked up from the
incoming caller ID.

All that fell by the wayside when Debian Squeeze introduced KDE4 and the
notification system I had created stopped working.  And some time after that,
we introduced queues instead of everyone having their own direct inbound
number .

Now, some tie-wearer is dribbling on me to bring back the old system.


I am confident that I could write something that will work with the new cross-
desktop notification model  (and in any case, that is a matter for Elsewhere On
The Internet).  However, I am going to need to hook it into Asterisk somehow.


What I think I need is for an event to fire when someone answers a queued call;
then I can run an AGI script, or execute a script using the System() command.
Within my script, I need the variable ${CALLERID(num)} to look up the caller's
details from their number, and the answering extension to decide where to send
the notification.

Is there a way of specifying in the dialplan or queue configuration that I want
to execute a script when an agent answers?

So far, all I can think of is joining local channels into the queue instead of
the actual phones, so I get to run a bit of dialplan where I can kick off the
AGI script and then Dial() the actual extension; but that could get terribly
unwieldy if not done extremely carefully.


(Of course, the manager in question also insists for me to implement all this
without a moment's downtime.  Kids, this is what happens when your brain is
deprived of oxygen .)

See 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Queue 
for information on running an AGI, macro or gosub when the caller is 
connected to an agent.


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[asterisk-users] sorcery.conf mappings

[Carlos Chavez]

Is there some documentation for all the available sorcery.conf 
mappings for realtime?  Asterisk already includes some tables in the 
database that are not enabled by default on the sorcery.conf like 
transports and outbound registrations.  There are no examples in the 
file on how to enable them.  Where can I find some documentation to 
enable those mappings?



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[asterisk-users] Force hangup not working on stuck channel

[Carlos Chavez]

I am unable to force a hangup on a channel that has been stuck for over 
two days:


IAX2/from-CD-11006   oficina  27701 Up  
Dial IAX2/to-CD/2883   3467130007  46:24:59 Sotelo   
   Sotelo  IAX2/to-CD-20713


I have tried "hangup request IAX2/from-CD-11006" several times but no 
joy.  I also see the following in the CLI:


[Nov  3 10:05:54] WARNING[2879]: chan_iax2.c:4936 handle_call_token: Too 
much delay in IAX2 calltoken timestamp from address X.X.X.X


This is an IAX2 trunk between two Asterisk 1.8 servers (I know it is old 
but new client so haven't had time yet to upgrade to 13).  Because this 
channels is stuck
 all other calls between servers are not working.  The only way I have 
found to resolve the problem is to stop and restart Asterisk.  This is 
obviously a great inconvinience so is there a way for force iax to 
unload even if there are channels in use?  Or any other way to kill 
these stubborn channels?


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Re: [asterisk-users] Conference Call like conference done by mobile!

[Carlos Chavez]

On 10/5/16 7:04 AM, Mandar Khire wrote:


hi,
I trying to solve one scenario:-
As I can make call from mobile phone to my friend1. As he accept it, I 
put him on hold, & dial friend2.
As he also accept it, I put him on hold & follow same procedure till 
friend6.
The I click on 'Merge call' & I can talk to all 6 friends at a time & 
they can talk each other.


Can I write This scene by dialplan?How?

I used Confbridge but its different type of conference.

Need help.
Thanks.

What you are mentioning is a function of the phone an not of 
Asterisk.  The phone has to support all those channels and mix them 
locally.  Most phones only do three way calling but some can do more.  
What is the problem with dumping everyone into a Confbridge conference 
room?  Same result at the end.



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Re: [asterisk-users] Mysql PJSIP realtime > 13.10?

[Carlos Chavez]

On 9/12/16 4:21 PM, George Joseph wrote:




On Mon, Sep 12, 2016 at 3:01 PM, Carlos Chavez 
<cur...@telecomabmex.com <mailto:cur...@telecomabmex.com>> wrote:


On 9/12/16 3:39 PM, George Joseph wrote:




On Mon, Sep 12, 2016 at 2:31 PM, George Joseph
<gjos...@digium.com <mailto:gjos...@digium.com>> wrote:



On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez
<cur...@telecomabmex.com <mailto:cur...@telecomabmex.com>> wrote:

  Has anyone successfully used Mysql realtime PJSIP with
Asterisk 13.11?  I have tried 13.11, 13.11.1 and 13.11.2
but I always get the following error now:

Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'qualify_timeout' cannot be type 'int(10)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'expiration_time' cannot be type 'bigint(20)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1246
require_mysql: Possibly unsupported column type
'enum('yes','no')' on column 'authenticate_qualify'
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'via_port' cannot be type 'int(11)' (need char)
[Sep 12 14:42:35] ERROR[24498]: res_pjsip_registrar.c:411
register_aor_core: Unable to bind contact
'sip:2001@192.168.2.165:5060
<http://sip:2001@192.168.2.165:5060>;transport=udp' to
AOR '2001'
  == Contact 2001/sip:2001@192.168.2.165
<mailto:sip%3A2001@192.168.2.165>:5060;transport=udp has
been deleted

Up until 13.10 everything was working despite the
warnings about field types.  Now my phones will not
register.  I can make calls but not receive.  All
database modifications are done through alembic so they
are supposed to be up to date.  The only way I can find
to solve this issue right now is to restore a 13.10
backup for both the database and Asterisk.


res_config_mysql has been in "extended" support for some time
now and it's possible it just will no longer work.  We only
test alembic changes with postgres or odbc now. Your best bet
is to convert to res_odbc.


Oh yeah, if you really do need res_config_mysql, go ahead and
open an issue at issues.asterisk.org <http://issues.asterisk.org>
and we'll take a look.  Since we don't test with it though, we
might not notice if it gets broken again in the future unless
someone reports it.



I have solved the problem for the moment by changing the
ps_contacts table with the "recommendations" res_config_mysql is
giving.  I just modified all the fields to varchar and now my
phones are registering.  Obviously this is not a solution as the
database needs to be modified by alembic on future versions and it
will keep breaking.

I tried to migrate to res_config_odbc about 6 months ago but
my Asterisk kept crashing.  I was told that the crashes were due
to the version of ODBC distributed by CentOS 7 and that I would
have to compile my own to be able to solve the issue.  Has this
been solved?  Is the RPM ODBC package included with CentOS 7 still
bugged?  I try to avoid using packages not included in the
distribution as they make upgrades a pain later on.


Although CentOS7 is still on 2.3.1 for unixodbc, there were changes 
made in 13.10.0 that should prevent the crashes.

http://lists.digium.com/pipermail/asterisk-dev/2016-June/075582.html


Ok, thank you for the help.  I'll be trying out ODBC later today 
and test it for a few days.  You should probably mark res_config_mysql 
as deprecated now since it will not work with anything higher than 13.10 
as is.



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Re: [asterisk-users] Mysql PJSIP realtime > 13.10?

[Carlos Chavez]

On 9/12/16 3:39 PM, George Joseph wrote:




On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjos...@digium.com 
<mailto:gjos...@digium.com>> wrote:




On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez
<cur...@telecomabmex.com <mailto:cur...@telecomabmex.com>> wrote:

Has anyone successfully used Mysql realtime PJSIP with
Asterisk 13.11?  I have tried 13.11, 13.11.1 and 13.11.2 but I
always get the following error now:

Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'qualify_timeout' cannot be type 'int(10)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'expiration_time' cannot be type 'bigint(20)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1246
require_mysql: Possibly unsupported column type
'enum('yes','no')' on column 'authenticate_qualify'
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'via_port' cannot be type 'int(11)' (need char)
[Sep 12 14:42:35] ERROR[24498]: res_pjsip_registrar.c:411
register_aor_core: Unable to bind contact
'sip:2001@192.168.2.165:5060
<http://sip:2001@192.168.2.165:5060>;transport=udp' to AOR '2001'
  == Contact 2001/sip:2001@192.168.2.165
<mailto:sip%3A2001@192.168.2.165>:5060;transport=udp has been
deleted

Up until 13.10 everything was working despite the warnings
about field types.  Now my phones will not register.  I can
make calls but not receive.  All database modifications are
done through alembic so they are supposed to be up to date. 
The only way I can find to solve this issue right now is to

restore a 13.10 backup for both the database and Asterisk.


res_config_mysql has been in "extended" support for some time now
and it's possible it just will no longer work.  We only test
alembic changes with postgres or odbc now.  Your best bet is to
convert to res_odbc.


Oh yeah, if you really do need res_config_mysql, go ahead and open an 
issue at issues.asterisk.org <http://issues.asterisk.org> and we'll 
take a look.  Since we don't test with it though, we might not notice 
if it gets broken again in the future unless someone reports it.



I have solved the problem for the moment by changing the 
ps_contacts table with the "recommendations" res_config_mysql is 
giving.  I just modified all the fields to varchar and now my phones are 
registering.  Obviously this is not a solution as the database needs to 
be modified by alembic on future versions and it will keep breaking.


I tried to migrate to res_config_odbc about 6 months ago but my 
Asterisk kept crashing.  I was told that the crashes were due to the 
version of ODBC distributed by CentOS 7 and that I would have to compile 
my own to be able to solve the issue.  Has this been solved?  Is the RPM 
ODBC package included with CentOS 7 still bugged?  I try to avoid using 
packages not included in the distribution as they make upgrades a pain 
later on.


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Re: [asterisk-users] Mysql PJSIP realtime > 13.10?

[Carlos Chavez]

On 9/12/16 3:21 PM, Annus Fictus wrote:


Hello,

is there any reason you don't use ODBC with MySQL?

Regards


El 12/09/2016 a las 15:14, Carlos Chavez escribió:
Has anyone successfully used Mysql realtime PJSIP with Asterisk 
13.11?  I have tried 13.11, 13.11.1 and 13.11.2 but I always get the 
following error now:


Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 
require_mysql: Realtime table general@ps_contacts: column 
'qualify_timeout' cannot be type 'int(10)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 
require_mysql: Realtime table general@ps_contacts: column 
'expiration_time' cannot be type 'bigint(20)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1246 
require_mysql: Possibly unsupported column type 'enum('yes','no')' on 
column 'authenticate_qualify'
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 
require_mysql: Realtime table general@ps_contacts: column 'via_port' 
cannot be type 'int(11)' (need char)
[Sep 12 14:42:35] ERROR[24498]: res_pjsip_registrar.c:411 
register_aor_core: Unable to bind contact 
'sip:2001@192.168.2.165:5060;transport=udp' to AOR '2001'
  == Contact 2001/sip:2001@192.168.2.165:5060;transport=udp has been 
deleted


Up until 13.10 everything was working despite the warnings about 
field types.  Now my phones will not register.  I can make calls but 
not receive.  All database modifications are done through alembic so 
they are supposed to be up to date.  The only way I can find to solve 
this issue right now is to restore a 13.10 backup for both the 
database and Asterisk.






Unless you use non packaged odbc on CentOS 7, Asterisk keeps crashing.

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[asterisk-users] Mysql PJSIP realtime > 13.10?

[Carlos Chavez]

Has anyone successfully used Mysql realtime PJSIP with Asterisk 
13.11?  I have tried 13.11, 13.11.1 and 13.11.2 but I always get the 
following error now:


Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: 
Realtime table general@ps_contacts: column 'qualify_timeout' cannot be 
type 'int(10)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: 
Realtime table general@ps_contacts: column 'expiration_time' cannot be 
type 'bigint(20)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1246 require_mysql: 
Possibly unsupported column type 'enum('yes','no')' on column 
'authenticate_qualify'
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: 
Realtime table general@ps_contacts: column 'via_port' cannot be type 
'int(11)' (need char)
[Sep 12 14:42:35] ERROR[24498]: res_pjsip_registrar.c:411 
register_aor_core: Unable to bind contact 
'sip:2001@192.168.2.165:5060;transport=udp' to AOR '2001'
  == Contact 2001/sip:2001@192.168.2.165:5060;transport=udp has been 
deleted


Up until 13.10 everything was working despite the warnings about 
field types.  Now my phones will not register.  I can make calls but not 
receive.  All database modifications are done through alembic so they 
are supposed to be up to date.  The only way I can find to solve this 
issue right now is to restore a 13.10 backup for both the database and 
Asterisk.


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Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161


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[asterisk-users] How to get a list of DAHDI channels

[Carlos Chavez]

Anyone know an efficient way to get a list of the DAHDI channels?  
Is there an AMI or ARI variable to get a list of all the channels?



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Re: [asterisk-users] Asterisk 13.11 realtime problem registering phones

[Carlos Chavez]

On 2016-09-04 10:11, Max Grobecker wrote:

Hi,


Am 02.09.2016 um 22:48 schrieb Carlos Chavez:
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).


I hope you find this concerning...

Have you upgraded your MySQL realtime tables to the new schema as
introduced with Asterisk 13?
->
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13#UpgradingtoAsterisk13-RealTime

It's likely a database error (i.e. a required, but missing table
field) causes this issue.
But even if not, you are getting rid of the warning messages ;-)



Hi Max, I do find it concerning ;) (please do not take it the wrong way, 
I just found the usage of the word here funny).


I am using alembic to configure all my database tables so in theory that 
is the "official" way to do it.  Even so mysql keeps sending those 
warnings.  I guess the res_mysql_config code does not follow the same 
standards as odbc.  Like I mentioned, the warnings have been there since 
Asterisk 13.0 but everything was working at least.  I always run the 
alembic upgrade when installing a new version of Asterisk 13 to make 
sure realtime tables are up to date.


I not only had to go back to Asterisk 13.10 but had to revert to a 
previous backup of the database so I guess what really broke things was 
the database update.


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Carlos Chávez
dCAP #1349
+52 (55)9116-91161

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[asterisk-users] Asterisk 13.11 realtime problem registering phones

[Carlos Chavez]

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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[asterisk-users] PJSIP hints unreliable...

[Carlos Chavez]

I find that using hints with PJSIP on Asterisk 13 is very 
unreliable compared to regular SIP.  I see many phones as unavailable 
when they are in fact available.  Usually hints will work fine for a 
while after a phone registers but after a while it will remain at 
unavailable while it is idle.  If the phone is in use I will get the 
correct status from the hint (InUse, OnHold, Ringing) but it will go 
back to unavailable after ther call is done.


IAX2 endpoints do not have this problem and their hints work all 
the time.  Since moving to PJSIP we are registering several devices 
using the same account and I find that these endpoints tend to dissapear 
from my console after one of the devices unregisters even though there 
is still another one active.


Any recommendations or configurations needed to stabilize hints for 
PJSIP?



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Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161


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