Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[Robin Kipp]

Hello all,

thanks to all of you for your truly excellent ideas and suggestions, greatly 
appreciated!
You have definitely given me a lot of things to play around with in the coming 
days.
For starters, I think I will try the built-in Follow Me feature as suggested by 
Karl, as it seems like this would be the easiest in order to get basic 
functionality up and running.
Following that I will definitely try something more complex, such as the setup 
suggested by Adam.
Using some PHP and / or Python, I should even be able to integrate with Swarm 
by Foursquare. In that case, the Asterisk server would be notified if I check 
in in a new country using the Swarm API, and consequently redirect calls to the 
corresponding mobile number. This would be pretty sweet, as it would basically 
allow me to configure call redirection using the Swarm app on my phone.
I will also try changing the SIP port and see what happens, at least it would 
allow me to use SIP over the cell network while I have an appropriately fast 
connection.

Many thanks once again, this has definitely helped me make up my mind about 
what’s possible!
Best wishes,
Robin
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Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[Pete Mundy]

On 29/04/2016, at 3:46 am, A J Stiles  wrote:
> 
> 

> 
> There is no reliable way to distinguish whether a phone was answered by a 
> human being or a machine.  
>  
> If you can't just disable voicemail on all your SIMs then you will need to 
> find out how long each carrier will wait before diverting to voicemail, and 
> then make sure the timeouts in your Dial() statements are short enough not to 
> trigger the carrier's voicemail.  Then use Asterisk's VoiceMail() application 
> to record any message your caller might leave.
> 
> 


To make things worse, in the country I reside not one of the cellular telcos 
have the option of disabling voicemail! If you disable voicemail they still 
terminate the call and instead play a message to the effect of 'This user has 
disabled voicemail. Goodbye'.

We also can't rely on timeouts because if the destination cellphone is out of 
coverage or turned off then the telco will drop to VM (or no-VM warning) almost 
immediately.

Unfortunately for us, the only option is to code up a routine that checks for 
acceptance confirmation from the destination human, a-la what John was 
describing in FreePBX.

Pete



smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[Karl Anderson]

Followme can be configured to accept a keypress to accept or decline the
call. Put something like this in followme.conf:


[general]
featuredigittimeout=>5000
takecall=>1
declinecall=>2
call_from_prompt=>followme/call-from
norecording_prompt=>followme/no-recording
options_prompt=>followme/options
pls_hold_prompt=>followme/pls-hold-while-try
status_prompt=>followme/status
sorry_prompt=>followme/sorry

On Thu, Apr 28, 2016 at 8:46 AM, A J Stiles 
wrote:

> On Thursday 28 Apr 2016, Robin Kipp wrote:
>
> > Hi all,
>
> >
>
> > sorry if the subject is a bit confusing, but I just couldn’t think of a
>
> > good way of better describing the situation…
>
> >
>
> > Basically, I travel a lot and have several SIM cards for my phone from
>
> > local carriers. What I’d like to do now is to setup Asterisk, so that
>
> > people who want to reach me just have to dial one number which forwards
>
> > the call to all my cellphone numbers in turn. I’m still pretty new to
>
> > Asterisk, so I’m unsure which method would be most suitable for this
>
> > scenario.
>
> >
>
> > Theoretically, I could use the dial function to call one number, then
> wait
>
> > a few seconds and then dial another number. In practice, this won’t work
>
> > because as soon as a call is answered by the mobile carrier’s voicemail
>
> > the caller would be connected to that, no other numbers would be called.
>
> > So here’s my question: how can I possibly avoid this situation? Is there
> a
>
> > way for Asterisk to detect such situations and distinguish them from me
>
> > actually trying to answer the call when the correct number is called? Not
>
> > sure if this is technically possible, but figured I’d ask just in case
>
> > there is any sort of solution. I’m aware that it would be best to simply
>
> > use SIP and a SIP client on my phone in order to take the call, but due
> to
>
> > most carriers blocking SIP traffic on their mobile data networks this
>
> > wouldn’t work as soon as I’m not connected to any WiFi. So, in case
>
> > there’s any solution to this problem I’d greatly appreciate if you could
>
> > share that with me! Many thanks and best wishes,
>
> > Robin
>
>
>
> There is no reliable way to distinguish whether a phone was answered by a
> human being or a machine.
>
>
>
> If you can't just disable voicemail on all your SIMs then you will need to
> find out how long each carrier will wait before diverting to voicemail, and
> then make sure the timeouts in your Dial() statements are short enough not
> to trigger the carrier's voicemail. Then use Asterisk's VoiceMail()
> application to record any message your caller might leave.
>
>
>
> You can just chain a whole bunch of Dial() statements one after another
> within an extension because once the first one has been answered, execution
> will proceed to the "h" extension.
>
>
>
> If you install ConnectBot on your mobile phone, you should be able to
> login to your Asterisk server each time you swap in a new SIM card, and
> edit your dialplan so the phone you are using today gets tried first. This
> could even be automated, but talk of such may not be appropriate for a
> non-commercial list.
>
>
>
> Out of politeness to the caller, play them a recorded announcement before
> your bank of Dial() statements, so they know to wait while your Asterisk
> box searches for you.
>
>
>
> Finally, remember: You will be tying up two channels -- and therefore
> maybe two DAHDI spans, depending how the calls are coming into and out of
> your Asterisk box -- with this.
>
>
>
> --
>
> AJS
>
>
>
> Note: Originating address only accepts e-mail from list! If replying
> off-list, change address to asterisk1list at earthshod dot co dot uk .
>
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Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[A J Stiles]

On Thursday 28 Apr 2016, Robin Kipp wrote:
> Hi all,
> 
> sorry if the subject is a bit confusing, but I just couldn’t think of a
> good way of better describing the situation…
> 
> Basically, I travel a lot and have several SIM cards for my phone from
> local carriers. What I’d like to do now is to setup Asterisk, so that
> people who want to reach me just have to dial one number which forwards
> the call to all my cellphone numbers in turn. I’m still pretty new to
> Asterisk, so I’m unsure which method would be most suitable for this
> scenario.
> 
> Theoretically, I could use the dial function to call one number, then wait
> a few seconds and then dial another number. In practice, this won’t work
> because as soon as a call is answered by the mobile carrier’s voicemail
> the caller would be connected to that, no other numbers would be called.
> So here’s my question: how can I possibly avoid this situation? Is there a
> way for Asterisk to detect such situations and distinguish them from me
> actually trying to answer the call when the correct number is called? Not
> sure if this is technically possible, but figured I’d ask just in case
> there is any sort of solution. I’m aware that it would be best to simply
> use SIP and a SIP client on my phone in order to take the call, but due to
> most carriers blocking SIP traffic on their mobile data networks this
> wouldn’t work as soon as I’m not connected to any WiFi. So, in case
> there’s any solution to this problem I’d greatly appreciate if you could
> share that with me! Many thanks and best wishes,
> Robin

There is no reliable way to distinguish whether a phone was answered by a 
human being or a machine.  

If you can't just disable voicemail on all your SIMs then you will need to find 
out how long each carrier will wait before diverting to voicemail, and then 
make sure the timeouts in your Dial() statements are short enough not to 
trigger the carrier's voicemail.  Then use Asterisk's VoiceMail() application 
to record any message your caller might leave.

You can just chain a whole bunch of Dial() statements one after another within 
an extension because once the first one has been answered, execution will 
proceed to the "h" extension.

If you install ConnectBot on your mobile phone, you should be able to login to 
your Asterisk server each time you swap in a new SIM card, and edit your 
dialplan so the phone you are using today gets tried first.  This could even be 
automated, but talk of such may not be appropriate for a non-commercial list.

Out of politeness to the caller, play them a recorded announcement before your 
bank of Dial() statements, so they know to wait while your Asterisk box 
searches for you.

Finally, remember:  You will be tying up two channels -- and therefore maybe 
two DAHDI spans, depending how the calls are coming into and out of your 
Asterisk box -- with this.  

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[ka]

Theoretically, I could use the dial function to call one number, then
wait a few seconds and then dial another number. In practice, this
won’t work because as soon as a call is answered by the mobile
carrier’s voicemail the caller would be connected to that, no other
numbers would be called.


I never understood why people have voicemails.  Switch off the voicemail 
feature, so it won't pick up the call.


My solution to this problem is rather manual.  I travel across countries 
as well, so here is what I have:


[...]
exten => dialme,n,Macro(get_mobile_target)
exten => dialme,n,Dial(SIP/${TARGET_PROVIDER}/${TARGET_NUMBER},35,t)
[...]

Now, as you can see, the above uses a macro to set some variables, so 
let's look at it:


Apart from many logging and other, non-relevant features, it calls an 
AGI:


exten => s,n,Agi(get_mobile_target.agi)

What this AGI does, it looks up in an SQL table which provider and which 
destination number to call at any given time.  These can change 
dynamically as I travel around, hence it's in SQL.  How will values be 
updated in SQL?  By two ways.


1) I have a web frontend where I can logon and select from a dropdown 
menu, which simcard is my current cellphone number.  If I want, I can 
also select a provider from a dropdown list, as I have more and there 
are different deals with each.  So I won't be calling my US cellphone 
number from a provider that provides me good EU rates and only those.


Usually I use airport wifi or 3G/4G at the departing airport to select 
the cell number from the list.


2) I have DID number which I can call from whichever cellphone I'm 
currently using, where the call is handled by an AGI on the PBX.  If the 
callerID matches one from a pre-defined list (my known sim card 
numbers), it assumes that I'm checking in and the caller ID gets 
selected the same way as if I was making the selection from the web 
interface.  This call costs nothing as the call is never answered, the 
AGI hungs up the call.


I understand that both are rather manual and not exactly textbook 
follow-me but they are pretty simple and it works well for me.


regards
Adam



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Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[Frank Vanoni]

Just a few ideas...


1. Disable all mobile carrier's voicemail and configure a voicemail on
your Asterisk. Let Asterisk handle the unanswered calls.

2. If your SIP provider allows multiple calls at the same time,
configure Asterisk to call all your SIMs at once (instead of calling the
first, wait... calling the second... wait and so on).

3. If your mobile carrier blocks SIP on your data plan, simply configure
Asterisk <-> SIP client on your mobile phone to use another port. 
Or, even better, you can use IAX instead of SIP. On your mobile device
install a client that supports IAX (for example, Zoiper).


Frank


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Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[covici]

I know if you use freepbx on top of asterisk, you get a followme which
calls one or more cell phones and ask for confirmation, maybe the
regular asterisk followme does this as well, but basically this is the
way to do it.

Robin Kipp  wrote:

> Hi all,
> 
> sorry if the subject is a bit confusing, but I just couldn’t think of a good 
> way of better describing the situation…
> 
> Basically, I travel a lot and have several SIM cards for my phone from local 
> carriers. What I’d like to do now is to setup Asterisk, so that people who 
> want to reach me just have to dial one number which forwards the call to all 
> my cellphone numbers in turn. I’m still pretty new to Asterisk, so I’m unsure 
> which method would be most suitable for this scenario.
> 
> Theoretically, I could use the dial function to call one number, then wait a 
> few seconds and then dial another number. In practice, this won’t work 
> because as soon as a call is answered by the mobile carrier’s voicemail the 
> caller would be connected to that, no other numbers would be called.
> So here’s my question: how can I possibly avoid this situation? Is there a 
> way for Asterisk to detect such situations and distinguish them from me 
> actually trying to answer the call when the correct number is called?
> Not sure if this is technically possible, but figured I’d ask just in case 
> there is any sort of solution. I’m aware that it would be best to simply use 
> SIP and a SIP client on my phone in order to take the call, but due to most 
> carriers blocking SIP traffic on their mobile data networks this wouldn’t 
> work as soon as I’m not connected to any WiFi.
> So, in case there’s any solution to this problem I’d greatly appreciate if 
> you could share that with me!
> Many thanks and best wishes,
> Robin
> -- 
> _
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How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] my dahdi dont'n start

[A J Stiles]

On Thursday 28 Apr 2016, Mamadou NGOM wrote:
> Hello,
>  it doesn't work my dahdi yet .for information, i use debian 8 .
> I put the file dahdi.bash   in /etc/init.d and I gave it the permission 755
> but i have  the same error: bash: /etc/init.d/dahdi: No such file or
> directory

You need to name the file just "dahdi", not "dahdi.bash"; because the command 
"service dahdi start" is looking for a file just called "dahdi".  If you run
# mv /etc/init.d/dahdi.bash /etc/init.d/dahdi
then
# service dahdi start
should work.

You probably also need to run
# update-rc.d dahdi defaults
to ensure it starts up everytime the computer is booted up.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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[asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

[Robin Kipp]

Hi all,

sorry if the subject is a bit confusing, but I just couldn’t think of a good 
way of better describing the situation…

Basically, I travel a lot and have several SIM cards for my phone from local 
carriers. What I’d like to do now is to setup Asterisk, so that people who want 
to reach me just have to dial one number which forwards the call to all my 
cellphone numbers in turn. I’m still pretty new to Asterisk, so I’m unsure 
which method would be most suitable for this scenario.

Theoretically, I could use the dial function to call one number, then wait a 
few seconds and then dial another number. In practice, this won’t work because 
as soon as a call is answered by the mobile carrier’s voicemail the caller 
would be connected to that, no other numbers would be called.
So here’s my question: how can I possibly avoid this situation? Is there a way 
for Asterisk to detect such situations and distinguish them from me actually 
trying to answer the call when the correct number is called?
Not sure if this is technically possible, but figured I’d ask just in case 
there is any sort of solution. I’m aware that it would be best to simply use 
SIP and a SIP client on my phone in order to take the call, but due to most 
carriers blocking SIP traffic on their mobile data networks this wouldn’t work 
as soon as I’m not connected to any WiFi.
So, in case there’s any solution to this problem I’d greatly appreciate if you 
could share that with me!
Many thanks and best wishes,
Robin
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Re: [asterisk-users] Upgrading 13.7 (external pjproject) to 13.9 (bundled pjproject)

[George Joseph]

On Thu, Apr 28, 2016 at 1:10 AM, Dmitriy Serov  wrote:

> Today was another attempt to upgrade to version 13.9 (git).
>
> 1. The result was https://issues.asterisk.org/jira/browse/ASTERISK-25970
>
> Had to temporarily block this contact and look forward to advice of how to
> fix it.
>
> 2. Also, an unpleasant surprise was the increase in CPU usage from 10-50%
> to 200-400% (4 cores). Stable CPU overusage with the same build options
> (DONT_OPTIMIZE, DEBUG_THREADS, BETTER_BACKTRACES, BUILD_NATIVE)
>


Does this also happen with an external pjproject?


>
> 3. After 20 minutes of this work the server has ceased to respond promptly
> to registrations. "core show locks" is attached. Suggestions about solving
> the problem are very welcome.
>
> Dmitriy.
>
>
>
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Re: [asterisk-users] my dahdi dont'n start

[Luca Bertoncello]

Zitat von Mamadou NGOM :


Hello,

 it doesn't work my dahdi yet .for information, i use debian 8 .

I put the file dahdi.bash   in /etc/init.d and I gave it the permission
755 but i have  the same error:

bash: /etc/init.d/dahdi: No such file or directory


Well, if the file's name ist dahdi.bash, you cannot just start dahdi...

I suggest you to rename the file in dahdi.
In Unix you don't need an extension as in Windows.

Regards
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] my dahdi dont'n start

[Mamadou NGOM]

Hello, it doesn't work my dahdi yet .for information, i use debian 8 .I put the file dahdi.bash   in /etc/init.d and I gave it the permission 755 but i have  the same error:bash: /etc/init.d/dahdi: No such file or directoryThanks for your help.> Le 28 avril 2016 à 13:27, A J Stiles  a écrit :> > > On Thursday 28 Apr 2016, Mamadou NGOM wrote:> > Hello,> > you are right, i dont have the file dahdi on /etc/init.d.> > I should like do what ?> > thank for your answer.> > Find my earlier message and save the file I attached to it. Read it, or show > it to a competent programmer whom you trust, and satisfy yourself that it is > not going to do anything nasty. Copy it over onto your Asterisk box using > scp, mv it to /etc/init.d/dahdi and give it 755 permissions. Then try> # service dahdi start> That ought to fix it.> > > By the way: It is considered polite to place your reply *after* the thing you > are replying to, so as to maintain the natural flow of the conversation.> > -- > AJS> > Note: Originating address only accepts e-mail from list! If replying off-> list, change address to asterisk1list at earthshod dot co dot uk .Mamadou NGOMIngénieur Télécommunications & RéseauxMobile: 06 72 45 23 03Skype: Mamadou NumericapNumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015. siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: fina...@numericap.comCentre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52 

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[asterisk-users] codec_opus w/ PLR and FEC for Asterisk 11

[sean darcy]

There is an opus patch for asterisk 11. 
https://github.com/seanbright/asterisk-opus/tree/asterisk-11 . But it 
doesn't have Packet Loss Resilience or Forward Error Correction, both of 
which are important for voip.



2.1.6.  Packet Loss Resilience

Audio codecs often exploit inter-frame correlations to reduce the 
bitrate at a cost in error propagation: after losing one packet, several 
packets need to be received before the decoder is able to accurately 
reconstruct the speech signal. The extent to which Opus exploits 
inter-frame dependencies can be adjusted on the fly to choose a 
trade-off between bitrate and amount of error propagation.


2.1.7.  Forward Error Correction (FEC)

Another mechanism providing robustness against packet loss is the 
in-band Forward Error Correction (FEC). Packets that are determined to 
contain perceptually important speech information, such as onsets or 
transients, are encoded again at a lower bitrate and this re-encoded 
information is added to a subsequent packet.



There is an opus patch for asterisk 13 that includes PLR and FEC. 
https://github.com/traud/asterisk-opus.


We don't have approval to move to 13. So anybody have an opus patch for 
11 that includes PLR and FEC ?


sean



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[asterisk-users] Upgrading 13.7 (external pjproject) to 13.9 (bundled pjproject)

[Dmitriy Serov]

Today was another attempt to upgrade to version 13.9 (git).

1. The result was https://issues.asterisk.org/jira/browse/ASTERISK-25970

Had to temporarily block this contact and look forward to advice of how 
to fix it.


2. Also, an unpleasant surprise was the increase in CPU usage from 
10-50% to 200-400% (4 cores). Stable CPU overusage with the same build 
options (DONT_OPTIMIZE, DEBUG_THREADS, BETTER_BACKTRACES, BUILD_NATIVE)


3. After 20 minutes of this work the server has ceased to respond 
promptly to registrations. "core show locks" is attached. Suggestions 
about solving the problem are very welcome.


Dmitriy.


Setting max files open to 30

===
=== 13.9.0-rc1
=== Currently Held Locks
===
===
===   (): 
 (times locked)
===
=== Thread ID: 0x7fbaedb2c700 LWP:25598 (worker_start started at [ 
1077] threadpool.c worker_thread_start())
=== ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields 
&(_type->wizards)->lock 0x2243420 (1)
main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D)
main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA)
main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2)
res_pjsip/pjsip_options.c:290 find_an_endpoint()
res_pjsip/pjsip_options.c:342 qualify_contact()
res_pjsip/pjsip_options.c:443 qualify_contact_task()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:1320 execute_tasks()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:351 threadpool_execute()
main/threadpool.c:1103 worker_active()
main/threadpool.c:1024 worker_start()
main/utils.c:1235 dummy_start()
:0 start_thread()
:0 __clone() (0x7fbb8bb68860+6D)
=== ---
===
=== Thread ID: 0x7fbaedab0700 LWP:25596 (worker_start started at [ 
1077] threadpool.c worker_thread_start())
=== ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields 
&(_type->wizards)->lock 0x2243420 (1)
main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D)
main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA)
main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2)
res_pjsip/pjsip_options.c:290 find_an_endpoint()
res_pjsip/pjsip_options.c:342 qualify_contact()
res_pjsip/pjsip_options.c:443 qualify_contact_task()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:1320 execute_tasks()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:351 threadpool_execute()
main/threadpool.c:1103 worker_active()
main/threadpool.c:1024 worker_start()
main/utils.c:1235 dummy_start()
:0 start_thread()
:0 __clone() (0x7fbb8bb68860+6D)
=== ---
===
=== Thread ID: 0x7fbb25795700 LWP:25597 (worker_start started at [ 
1077] threadpool.c worker_thread_start())
=== ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields 
&(_type->wizards)->lock 0x2243420 (1)
main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D)
main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA)
main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2)
res_pjsip/pjsip_options.c:290 find_an_endpoint()
res_pjsip/pjsip_options.c:342 qualify_contact()
res_pjsip/pjsip_options.c:443 qualify_contact_task()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:1320 execute_tasks()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+10D)
main/threadpool.c:351 threadpool_execute()
main/taskprocessor.c:852 ast_taskprocessor_execute() (0x5e02d3+8CB8)
main/threadpool.c:1024 worker_start()
main/utils.c:1235 dummy_start()
:0 start_thread()
:0 __clone() (0x7fbb8bb68860+6D)
=== ---
===
=== Thread ID: 0x7fbb254ad700 LWP:25895 (worker_start started at [ 
1077] threadpool.c worker_thread_start())
=== ---> Lock #0 (sorcery.c): RDLOCK 1883 ast_sorcery_retrieve_by_fields 
&(_type->wizards)->lock 0x2243420 (1)
main/backtrace.c:59 __ast_bt_get_addresses() (0x466440+1D)
main/lock.c:866 __ast_rwlock_rdlock() (0x53652a+BA)
main/sorcery.c:1884 ast_sorcery_retrieve_by_fields() (0x5bf860+C2)
res_pjsip/pjsip_options.c:290 find_an_endpoint()
res_pjsip/pjsip_options.c:342 qualify_contact()
res_pjsip/pjsip_options.c:443 qualify_contact_task()
main/taskprocessor.c:852 ast_taskprocessor_execute()