[asterisk-users] Outgoing PJSIP using Kamailio

2020-04-06 Thread Administrator

Hello,

We have a provider which is using Kamailio as front end. Our asterisk 
13/chan_sip server has no problem to register and pass/receive calls 
form this provider.


Now we want to move to asterisk 16/pjsip and face problem. Registration 
is OK but when we pass a call our INVITE never receive answer from the 
provider. We opened a ticket to their support but in the mean time we 
want to know if someone is using successfully a PJSIP channel against 
Kamailio.


Another one: despite the fact that they use 5061 port, it's not TLS but 
UDP. Our asterisk16 has no TLS configured.


We use wizard which looks like:

[Provider-tootai](!)
;
type = wizard
sends_auth = yes
sends_registrations = yes
accepts_auth = no
accepts_registrations = no
endpoint/call_group = 1
endpoint/pickup_group = 1
endpoint/accountcode = TOOTAi
endpoint/language = fr
endpoint/allow = !all,ulaw,alaw,g729
endpoint/context = incoming-Provider
endpoint/direct_media = no
endpoint/dtmf_mode = inband
registration/retry_interval = 20
registration/max_retries = 0
registration/expiration = 3600
registration/transport = transport-udp
aor/max_contacts = 2
aor/qualify_frequency = 2000

[Provider](Provider-tootai)
;
remote_hosts = sips.provider.eu
endpoint/callerid = "TOOTAi" <00xx xxx xxx xxx>
aor/contact = sip:sips.provider.eu:5061
registration/client_uri = sips:our...@sips.provider.eu
registration/server_uri = sips:sips.provider.eu:5061
outbound_auth/username = OUR_ID
outbound_auth/password = OUR_PWD
identity/match = PROVIDER_IP

Thanks for any hint

--
Daniel

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PJSIP Lockup

2020-04-06 Thread George Joseph
On Thu, Apr 2, 2020 at 11:34 AM Nick Olsen 
wrote:

> Paddy, It's pretty easy to spot from the CLI.
>
> A voicemail gets called. And the screen basically stops scrolling from
> there. Eventually you'll get the "Task processors exceeded 500 queued
> tasks" or something like that. And maybe channels attempting to hangup due
> to lack of RTP (If you have no-rtp timers configured).
>
> Once you find the problem mailbox, You can call it via any method and
> it'll deadlock every time as soon as it tries to play the mailboxes unavail
> greeting. I've never had it occur when there is no unavail greeting. Each
> case deleting the problem recording from the database fixes the issue, And
> subsequent recordings for the same mailbox have no issue.
>

Given that the issue appears to be related to specific rows and not the
database in general, you might want to get a backtrace while the system is
locked as Josh suggested earlier.   Once you get the backtraces, open an
issue ar https://issues.asterisk.org.

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
NOTE: you do NOT need to recompile with the DEBUG_THREADS, MALLOC_DEBUG,
DONT_OPTIMIZE or BETTER_BACKTRACES but the Asterisk binaries need to still
have the symbols in them (un-stripped).



>
> *Nick Olsen*
> Network Engineer
> Office: 321-408-5000 x103
> Mobile: 321-794-0763
>
>
>
> On Wed, Apr 1, 2020 at 9:04 PM Paddy Grice  wrote:
>
>> Hi All
>>
>> This sounds just like a problem I have had and still investigating having
>> moved to 16.9 using chan_sip. I am still trying to repeat the problem it
>> looks from debug that the issue is either voicemail of call transfer but I
>> cant consistently repeat it.
>>
>> Voicemail is using ODBC and I just imported the data from the old system
>> into the new database.
>>
>> Nick - if you have any more info I would be grateful
>>
>> TIA
>>
>> Paddy
>>
>> --
>> *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
>> Behalf Of *Nick Olsen
>> *Sent:* 01 April 2020 18:54
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] PJSIP Lockup
>>
>> We ultimately found this to be a voicemail issue. The voicemail is held
>> in MYSQL as well (via ODBC). And we found when attempting to playback a
>> customers voicemail unavail greeting is when the deadlock would occur
>> (Immediately, every time. Throwing the same "task processors" errors, And
>> making pjsip completely unresponsive). We had imported a number of
>> greetings from a legacy asterisk system and the vast majority of them
>> worked. When we deleted the row containing the customers unavail greeting
>> (making asterisk revert to read the mailbox number) all issues went away.
>> If we re-record the customers unavail greeting it works fine and the
>> problem doesn't reoccur. This was one out of ~250 voicemails imported.
>>
>> Since then we've done a few more migrations and they've all gone smooth
>> with the exception of the most recent one. ~50% of the imported greetings
>> have caused asterisk to deadlock. We've been checking them now at time of
>> migration.
>>
>> What I can't figure out is what it doesn't like about the greeting. It
>> was on a previous asterisk system working fine. The row looks identical to
>> a working one. The only thing I can guess is something about the blob for
>> the recording goes wrong. It would be nice if asterisk handled that more
>> gracefully.
>>
>> I post this mostly just for internet history. To hopefully help the next
>> guy out who has this same issue.
>>
>> *Nick Olsen*
>> Network Engineer
>> Office: 321-408-5000 x103
>> Mobile: 321-794-0763
>>
>>
>>
>> On Mon, Mar 2, 2020 at 8:29 PM Joshua C. Colp  wrote:
>>
>>> On Mon, Mar 2, 2020 at 4:24 PM Nick Olsen <
>>> n...@floridavirtualsolutions.com> wrote:
>>>
 Thanks for the info, Joshua.

 Does PJSIP handle database access the same way Chan_sip did? We had a
 number of boxes running chan_sip referencing the same mysql server without
 issue.

 We're going to attempt to get a backtrace on the next occurance. We're
 also going to run a local copy of the database on the same physical
 asterisk instance and have the system reference it. Just to "throw
 everything at the wall".

>>>
>>> It uses the same underlying API and layer. It can do more frequent
>>> database access though due to queries and because PJSIP is multithreaded.
>>>
>>> --
>>> Joshua C. Colp
>>> Asterisk Technical Lead
>>> Sangoma Technologies
>>> Check us out at www.sangoma.com and www.asterisk.org
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>>