[asterisk-users] Asterisk Getting Crashed

[Ahmed Chohan]

Hi,

Currently I'm experiencing crashes on Asterisk more recently, see messages
below (crashed reason: segfault signal 6).

abrt-hook-ccpp[19864]: Process 7082 (asterisk) of user 0 killed by SIGABRT
- dumping core

asterisk: ERROR[15373][C-0004e304]: astobj2.c:131 in INTERNAL_OBJ: FRACK!,
Failed assertion bad magic number 0x0 for object 0x7fbd2c

00d170 (0)

After running the backtrace for the coredump, I'm unable to pinpoint the
root cause of it (see partial messages for the backtrace below).
Furthermore, I've checked in the forums and advised the "utils.so" module
issue but I don't think it might be causing this crash.

[root@alpha01 ccpp-2020-06-25-10-46-01-7082]# gdb /usr/sbin/asterisk
coredump

GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-119.el7

Copyright (C) 2013 Free Software Foundation, Inc.

License GPLv3+: GNU GPL version 3 or later 

This is free software: you are free to change and redistribute it.

There is NO WARRANTY, to the extent permitted by law.  Type "show copying"

and "show warranty" for details.

This GDB was configured as "x86_64-redhat-linux-gnu".

For bug reporting instructions, please see:

...

Reading symbols from /usr/sbin/asterisk...done.

[New LWP 15373]

[New LWP 15800]

[New LWP 16125]

[New LWP 15829]

[New LWP 16486]

..

Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.

Program terminated with signal 6, Aborted.

#0  0x7fbe7a65f337 in ?? ()

(gdb) bt full

#0  0x7fbe7a65f337 in ?? ()

No symbol table info available.

#1  0x7fbe7a660a28 in ?? ()

No symbol table info available.

#2  0x0020 in ?? ()

No symbol table info available.

#3  0x in ?? ()

No symbol table info available.


OS I'm running is CentOs 7.7.1908 and the Asterisk version is 13.21-cert3.
Please advise.

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Realtime PJSIP max_streams' issues

[Ahmed Chohan]

After altering the table; changing type from int(11) to varchar for
max_audio_streams & max_video_stream, it is working.

Thanks Josh.

Date: Wed, 23 Oct 2019 14:40:48 -0300
> From: "Joshua C. Colp" 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Realtime PJSIP max_streams' issues
> Message-ID: <28f0701f-3790-459b-8818-04c0975c6...@www.fastmail.com>
> Content-Type: text/plain
>
> On Wed, Oct 23, 2019, at 2:30 PM, Ahmed Chohan wrote:
> > The database I'm using is MySQL v 5.6.46.2, data type I'm using for
> > both parameters is int(11) the one created by the asterisk script; see
> > table structure below.
>
> If you alter it to be a varchar instead does that change the result within
> PJSIP?
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Realtime PJSIP max_streams' issues

[Ahmed Chohan]

 varchar(40) DEFAULT NULL,
  `redirect_method` enum('user','uri_core','uri_pjsip') DEFAULT NULL,
  `set_var` text,
  `cos_audio` int(11) DEFAULT NULL,
  `cos_video` int(11) DEFAULT NULL,
  `message_context` varchar(40) DEFAULT NULL,
  `force_avp` enum('yes','no') DEFAULT NULL,
  `media_use_received_transport` enum('yes','no') DEFAULT NULL,
  `accountcode` varchar(80) DEFAULT NULL,
  `user_eq_phone` enum('yes','no') DEFAULT NULL,
  `moh_passthrough` enum('yes','no') DEFAULT NULL,
  `media_encryption_optimistic` enum('yes','no') DEFAULT NULL,
  `rpid_immediate` enum('yes','no') DEFAULT NULL,
  `g726_non_standard` enum('yes','no') DEFAULT NULL,
  `rtp_keepalive` int(11) DEFAULT NULL,
  `rtp_timeout` int(11) DEFAULT NULL,
  `rtp_timeout_hold` int(11) DEFAULT NULL,
  `bind_rtp_to_media_address` enum('yes','no') DEFAULT NULL,
  `voicemail_extension` varchar(40) DEFAULT NULL,
  `mwi_subscribe_replaces_unsolicited`
enum('0','1','off','on','false','true','no','yes') DEFAULT NULL,
  `deny` varchar(95) DEFAULT NULL,
  `permit` varchar(95) DEFAULT NULL,
  `acl` varchar(40) DEFAULT NULL,
  `contact_deny` varchar(95) DEFAULT NULL,
  `contact_permit` varchar(95) DEFAULT NULL,
  `contact_acl` varchar(40) DEFAULT NULL,
  `subscribe_context` varchar(40) DEFAULT NULL,
  `fax_detect_timeout` int(11) DEFAULT NULL,
  `contact_user` varchar(80) DEFAULT NULL,
  `preferred_codec_only` enum('yes','no') DEFAULT NULL,
  `asymmetric_rtp_codec` enum('yes','no') DEFAULT NULL,
  `rtcp_mux` enum('yes','no') DEFAULT NULL,
  `allow_overlap` enum('yes','no') DEFAULT NULL,
  `refer_blind_progress` enum('yes','no') DEFAULT NULL,
  `notify_early_inuse_ringing` enum('yes','no') DEFAULT NULL,
  `max_audio_streams` int(11) DEFAULT NULL,
  `max_video_streams` int(11) DEFAULT NULL,
  `webrtc` enum('yes','no') DEFAULT NULL,
  `dtls_fingerprint` enum('SHA-1','SHA-256') DEFAULT NULL,
  `incoming_mwi_mailbox` varchar(40) DEFAULT NULL,
  `bundle` enum('yes','no') DEFAULT NULL,
  `dtls_auto_generate_cert` enum('yes','no') DEFAULT NULL,
  `follow_early_media_fork` enum('yes','no') DEFAULT NULL,
  `accept_multiple_sdp_answers` enum('yes','no') DEFAULT NULL,
  `suppress_q850_reason_headers` enum('yes','no') DEFAULT NULL,
  `trust_connected_line` enum('0','1','off','on','false','true','no','yes')
DEFAULT NULL,
  `send_connected_line` enum('0','1','off','on','false','true','no','yes')
DEFAULT NULL,
  `ignore_183_without_sdp`
enum('0','1','off','on','false','true','no','yes') DEFAULT NULL,
  UNIQUE KEY `id` (`id`),
  KEY `ps_endpoints_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=latin1


-Original Message-
> From: asterisk-users  On Behalf
> Of Joshua C. Colp
> Sent: Tuesday, October 22, 2019 4:30 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Realtime PJSIP max_streams' issues
>
> On Tue, Oct 22, 2019, at 4:21 PM, Ahmed Chohan wrote:
> > Hi,
> >
> > I'm currently using Asterisk 16.4.0 cert version and working on webrtc.
> > For configuration perspective, I'm pretty much done with it but here
> > the real issue I'm currently facing i.e. when setting parameters
> > max_audio_streams & max_video_streams to any positive greater than 0
> > integer value in realtime (DB) of any endpoints. After running command
> > "pjsip show endpoint 100101" it shows '0' but when setting as
> > 'NULL' in DB, showing output to 1 for both parameters.
> >
> > Furthermore, in AOR section, the max_connection is set to 1 for each
> endpoints.
>
> The configuration option for there is max_contacts.
>
> > Please advise, for this issue.
>
> What database are you using? What type is the column? Do any other fields
> exhibit the problem?
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
> www.digium.com & www.asterisk.org
>
>
-- 
Regards,

Ahmed Munir Chohan
-- 
_
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[asterisk-users] Realtime PJSIP max_streams' issues

[Ahmed Chohan]

Hi,

I'm currently using Asterisk 16.4.0 cert version and working on webrtc. For
configuration perspective, I'm pretty much done with it but here the real
issue I'm currently facing i.e. when setting parameters max_audio_streams &
max_video_streams to any positive greater than 0 integer value in realtime
(DB) of any endpoints. After running command "pjsip show endpoint
100101" it shows '0' but when setting as 'NULL' in DB, showing output
to 1 for both parameters.

Furthermore, in AOR section, the max_connection is set to 1 for each
endpoints.

Please advise, for this issue.

-- 
Regards,

Ahmed Munir Chohan
-- 
_
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Re: [asterisk-users] PJSIP Setup Outbound SIP Trunk

[Ahmed Chohan]

Thanks for reply.

After going through the all configurations, there was syntax error with the
dial plan for outbound call i.e. previously I was using
"Dial(PJSIP/trunk_proxy/${EXTEN})" and was unable to make outbound calls.
Later changed to "Dial( PJSIP/${EXTEN}@ trunk_proxy)" it worked as expected
i.e. no need to set auth/reg for the SIP trunk as not setting it up at SIP
Proxy end.

Date: Wed, 16 Oct 2019 13:27:30 -0500
> From: Kevin Harwell 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] PJSIP Setup Outbound SIP Trunk
> Message-ID:
>  e2cuhg0xpdfpkida8zrkokvpv1s4ymqs9kgpop+a...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> On Mon, Oct 14, 2019 at 11:56 AM Ahmed Chohan 
> wrote:
>
> > Hi,
> >
> > I've currently migrating from chan_sip to chan_pjsip, for now I'm able to
> > setup and configured extensions in PJSIP and incoming trunks but unable
> to
> > configure outbound trunk as getting unauth/unregistered trunk endpoint
> > message error message when making outbound calls. However, for inbound
> > calls I'm not facing any issues.
> >
> > I would like to know how can I configured outbound sip trunk bypassing
> > registration and auth?
> >
>
> Where are the messages coming from? Is Asterisk sending an outbound
> registration, but getting rejected? If so make sure your username/password
> credentials are correct.
>
>
> >
> > See below current configuration;
> >
> > [trunk_proxy]
> > type=endpoint
> > transport=transport-udp
> > context=fromsip
> > disallow=all
> > allow=ulaw
> > aors=trunk_proxy
> > force_rport=no
> > direct_media=yes
> > ice_support=no
> > trust_id_inbound=yes
> > outbound_auth=trunk_proxy
> >
> > [trunk_proxy]
> > type=aor
> > contact=sip:10.3.120.208:5060
> >
> > [trunk_proxy]
> > type=identify
> > endpoint=trunk_proxy
> > match=10.3.120.208
> >
> > [trunk_proxy]
> > type=auth
> > auth_type=userpass
> > password=
> > username=sip_proxy
> >
> > [trunk_proxy]
> > type=registration
> > outbound_auth=trunk_proxy
> > server_uri=sip:10.3.120.208:5060
> > client_uri=sip:10.3.120.208:5060
> > auth_rejection_permanent=no
> >
> > --
> > Regards,
> >
> > Ahmed Munir Chohan
> >
> > --
> > _
> > -- 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
>
>
>
> --
> Kevin Harwell
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: https://digium.com & https://asterisk.org
> -- next part --
> An HTML attachment was scrubbed...
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> >
>
>
-- 
Regards,

Ahmed Munir Chohan
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[asterisk-users] PJSIP Setup Outbound SIP Trunk

[Ahmed Chohan]

Hi,

I've currently migrating from chan_sip to chan_pjsip, for now I'm able to
setup and configured extensions in PJSIP and incoming trunks but unable to
configure outbound trunk as getting unauth/unregistered trunk endpoint
message error message when making outbound calls. However, for inbound
calls I'm not facing any issues.

I would like to know how can I configured outbound sip trunk bypassing
registration and auth?

See below current configuration;

[trunk_proxy]
type=endpoint
transport=transport-udp
context=fromsip
disallow=all
allow=ulaw
aors=trunk_proxy
force_rport=no
direct_media=yes
ice_support=no
trust_id_inbound=yes
outbound_auth=trunk_proxy

[trunk_proxy]
type=aor
contact=sip:10.3.120.208:5060

[trunk_proxy]
type=identify
endpoint=trunk_proxy
match=10.3.120.208

[trunk_proxy]
type=auth
auth_type=userpass
password=
username=sip_proxy

[trunk_proxy]
type=registration
outbound_auth=trunk_proxy
server_uri=sip:10.3.120.208:5060
client_uri=sip:10.3.120.208:5060
auth_rejection_permanent=no

-- 
Regards,

Ahmed Munir Chohan
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[asterisk-users] App_Originate for over dynamic conference bridge

[Ahmed Chohan]

Hi,

I would like to know for App_Originate that how I can pass the argument to
next dialplan as listed below partial dialplan;


same => n,GotoIf($[${COUNT} > 0 ]?bridge_conference,1)
same => n,Set(__TMPEXTEN=${EXTEN}1)
same => n,ExecIf($[0${CONFBRIDGE_INFO(parties,${EXTEN})} ==
0]?Originate(Local/bridge_conference@main_con,app,ConfBridge,${TMPEXTEN}))
same => n(defaultBridge),ConfBridge(${EXTEN},,,user_menu)
 same => n,HangUp()

exten => bridge_conference,1,NoOp(argeument: ${TMPEXTEN})
 same => n,ConfBridge(${TMPEXTEN},,,user_menu)

While executing above dialplan the value (argument) in ${TMPEXTE} is not
passing through when moved to dial pattern. However when I use below
dialplan it works good.

same => n,GotoIf($[${COUNT} > 0 ]?bridge_conference,1)
same => n,ExecIf($[0${CONFBRIDGE_INFO(parties,${EXTEN})} ==
0]?Originate(Local/bridge_conference@main_con,app,ConfBridge,${EXTEN}))
same => n(defaultBridge),ConfBridge(${EXTEN},,,user_menu)
same => n,HangUp()

exten => bridge_conference,1,NoOp()
 same => n,ConfBridge(${EXTEN},,,user_menu)

My main objective is to create dynamic conference bridge instead of static
conference bridge if the number of participants increased to 1/10/100.. n.


-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] asterisk-users Digest, Vol 168, Issue 14

[Ahmed Chohan]

The limit 10 is just an assumption. For real number I've set in my
conference server is 350 max participants to join. As per the load testing
I've performed on the server, after 400+ participants voice is getting
choppy so I've set the max limit to 350 globally for safe reason.

If 5 other participants want to join the bridge even max limit has been
reached, please advise to merge remaining participants that joined
dynamically created bridge B with A.


Date: Thu, 23 Aug 2018 00:01:12 +0200
> From: Antony Stone 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Subject: Re: [asterisk-users] Merging 2 conference bridges
> Message-ID: <201808230001.12651.antony.st...@asterisk.open.source.it>
> Content-Type: Text/Plain;  charset="iso-8859-15"
>
> On Wednesday 22 August 2018 at 23:49:29, Ahmed Chohan wrote:
>
> > Hi,
> >
> > I would like to know how can I achieve merge 2 conference rooms in same
> > asterisk server. For example 10 users joined bridge A and max user limit
> is
> > set to 10. If more than 10 users try to join this bridge A, 11th user
> > should join to the dynamically created bridge B and merge with bridge A.
> So
> > that all eleven participants should be able to talk to each other.
>
> My first question upon seeing this is:
>
>  - if you want all 11 people to be able to talk to each other, why do you
> set
> a 10-participant limit on the original conference?
>
>
> Antony.
>
> --
> I conclude that there are two ways of constructing a software design: One
> way
> is to make it so simple that there are _obviously_ no deficiencies, and
> the
> other way is to make it so complicated that there are no _obvious_
> deficiencies.
>
>  - C A R Hoare
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
>

-- 
Regards,

Ahmed Munir Chohan
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[asterisk-users] Merging 2 conference bridges

[Ahmed Chohan]

Hi,

I would like to know how can I achieve merge 2 conference rooms in same
asterisk server. For example 10 users joined bridge A and max user limit is
set to 10. If more than 10 users try to join this bridge A, 11th user
should join to the dynamically created bridge B and merge with bridge A. So
that all eleven participants should be able to talk to each other.


Please advise.

-- 
Regards,

Ahmed Munir Chohan
-- 
_
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Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

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

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