Re: [asterisk-users] Crash asterisk res_odbc

[Rodrigo Ramírez Norambuena]

February 29 2016 5:40 AM, "Maxime"  wrote:
> Hi,
> 
> Thank you for the reply.
> 
> My OS is : Debian 7.
> 
> But i have more than 20 servers with the same features/resources (OS, 
> material, ... ) without the
> issue.
> 

Maybe could debug the crash  getting a Backtrace
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

And open a issue on the Jira
https://issues.asterisk.org/jira/browse/ASTERISK

Regards,
--
Rodrigo Ramírez Norambuena
http://www.rodrigoramirez.com

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Re: [asterisk-users] How to recive Incoming calls in Chan Dongle ?

[Vitor Mazuco]

Humm yes, thanks very much !
Em 04/03/2016 18:00, "Ashish Gupta"  escreveu:

> Hi Vitor,
>
> The dongle.conf file contains your configuration setting related to your
> particular dongle. There, set the "context=dongle" (or anything you
> specified in extensions.conf), then provide the "exten=1234" (The
> extensions that will be called in the particular context). Also provide the
> imei number in the end of the file.
>
> Now in the extensions.conf file start with the context you provided in
> dongle.conf(eg. dongle). Now write the following command to answer the call
> coming through the dongle-
>
> [dongle]
> exten => 1234,1,Answer()
>
> HTH,
> Ashish
>
> *Ashish Gupta*
> *B.Tech (ECE) 3rd Year*
> *The LNM Institute of Information Technology*
> *Jaipur, Rajasthan - 302031 , India*
> *Mobile No: +917597056895 <%2B917597056895>*
>
> On Sat, Mar 5, 2016 at 2:15 AM, Vitor Mazuco 
> wrote:
>
>> Hi!
>>
>> How can I setup my Chan Dongle recived calls in my Asterisk?
>>
>> I have to setup in dongle.conf ? Or in extensions.conf?
>>
>> And the code for recive I found this site
>> http://asterisk-service.com/page/chan-dongle-use
>>
>> I have to To save Subscriber Number before?
>>
>> See the error log in my Asterisk
>>
>> pbx.c:6796 __ast_pbx_run: Channel 'Dongle/dongle1-01' sent to
>> invalid extension but no invalid handler:
>> context,exten,priority=URA,+5511965380290,1,Noop(),1
>>
>>
>> Thanks in advanced.
>>
>> --
>> _
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>>http://www.asterisk.org/hello
>>
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>>
>
>
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Re: [asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

[Ryan, Travis]

Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com
(765) 742-1102

We're not the IT departmentWe're the I-TEAM department!


> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Patrick Laimbock
> Sent: Friday, March 04, 2016 10:58 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 13.5 and higher (asterisk
> 13.7.2) quitting
> 
> Hi Travis,
> 
> On 04-03-16 15:23, Ryan, Travis wrote:
> > I start asterisk 13.7.2 and it dies before I can rasterisk into it.
> > I've tried getting a coredump, but it doesn't coredump.  I know there
> > are a lot of errors in the log below, but most of those just say
> it'll
> > not load a module, and no big deal.
> >
> > When launching from commandline (not service script) here is what
> happens.
> >
> > http://pastebin.com/3GFe6fG9
> 
> Two things:
> 
> [Mar  3 15:19:37] WARNING[8439]: loader.c:553 load_dynamic_module:
> Error loading module 'res_monitor.so':
> /usr/lib/asterisk/modules/res_monitor.so: undefined symbol:
> __ast_beep_stop [Mar  3 15:19:37] WARNING[8439]: loader.c:553
> load_dynamic_module: Error loading module 'res_ari_events.so':
> /usr/lib/asterisk/modules/res_ari_events.so: undefined symbol:
> stasis_app_register_all
> 
> Undefined symbol errors are not good. Not sure why that's just a
> WARNING. Maybe something went wrong during the build? The output of the
> build should show you more information. In the mean time try disabling
> these two modules just to see if that clears up the problem.
> 
> [Mar  3 15:19:39]   == Parsing '/etc/asterisk/extensions.conf': Found
> [Mar  3 15:19:39] WARNING[8439]: config.c:2228 config_text_file_load:
> Unterminated comment detected beginning on line 386
> 
> That needs fixing.
> 
> HTH,
> Patrick
> 
> 
> --[Ryan, Travis] '
Thanks, now I'm at least getting a coredump. Ugh. 


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Re: [asterisk-users] How to recive Incoming calls in Chan Dongle ?

[Ashish Gupta]

Hi Vitor,

The dongle.conf file contains your configuration setting related to your
particular dongle. There, set the "context=dongle" (or anything you
specified in extensions.conf), then provide the "exten=1234" (The
extensions that will be called in the particular context). Also provide the
imei number in the end of the file.

Now in the extensions.conf file start with the context you provided in
dongle.conf(eg. dongle). Now write the following command to answer the call
coming through the dongle-

[dongle]
exten => 1234,1,Answer()

HTH,
Ashish

*Ashish Gupta*
*B.Tech (ECE) 3rd Year*
*The LNM Institute of Information Technology*
*Jaipur, Rajasthan - 302031 , India*
*Mobile No: +917597056895*

On Sat, Mar 5, 2016 at 2:15 AM, Vitor Mazuco  wrote:

> Hi!
>
> How can I setup my Chan Dongle recived calls in my Asterisk?
>
> I have to setup in dongle.conf ? Or in extensions.conf?
>
> And the code for recive I found this site
> http://asterisk-service.com/page/chan-dongle-use
>
> I have to To save Subscriber Number before?
>
> See the error log in my Asterisk
>
> pbx.c:6796 __ast_pbx_run: Channel 'Dongle/dongle1-01' sent to
> invalid extension but no invalid handler:
> context,exten,priority=URA,+5511965380290,1,Noop(),1
>
>
> Thanks in advanced.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] PJSIP signaling question

[Kevin Long]

I can’t quite figure it out , I went ahead and pulled everything yet again, and 
I made sure to delete everything related to pjproject from my system, all the 
PJ  lib and include files that were in /usr/lib/  ,  I pulled pjproject from 
svn , pulled asterisk code from gerrit, recompiled everything, but still I 
think new TLS transports are being made which fail in my NAT scenarios .  I 
check with: 

tcpdump -i any src host 10.50.55.10  and  'tcp[13] & 2 != 0’ 


I see tcpdump print a new tcp SYN packet when I  try to make a call between 
endpoints and also when Asterisk tries to send OPTIONS command to the endpoint .

From my endpoints, I can call the “echo” applications and the call works fine, 
but I cannot call from one endpoint to another endpoint , even though they are 
both egistered. It does not say “unavailable’ or anything,  I see in the pjsip 
log that an INVITE is  “sent” , but I think the logger is just showing me that 
the INVITE message has been created, but it never reaches the endpoint because 
of the new TLS connection failing because of the NAT. Eventually, the call 
times out with a 408 error in the pjsip log.

I also see some log entries:
[Mar  4 12:29:10] DEBUG[16225] pjsip:   tlsc0x7f311400 TLS connect() error: 
Connection timed out [code=120110]
[Mar  4 12:29:29] DEBUG[16225] pjsip:   tlsc0x7f311400 TLS connect() error: 
Connection timed out [code=120110]




Just to be clear I am getting pjproject like so : 
svn co http://svn.pjsip.org/repos/pjproject/trunk


and asterisk :
git clone -b 13 http://gerrit.asterisk.org/asterisk



then I go to pjproject directory,  create a site_config.h file (to increase TLS 
connectors and set other options recommended on Wiki)

configure pjproject with the following options:

./configure --prefix=/usr --enable-shared --disable-sound --disable-resample 
--disable-video --disable-opencore-amr --with-external-srtp



Then go to asterisk directory

make clean; make distclean; ./boostrap.sh ; ./configure;  make menuselect; 
make; make install;










> On Mar 4, 2016, at 7:33 AM, George Joseph  wrote:
> 
> 
> 
> On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long  wrote:
> Hi George the patch was from here , you wrote it I believe . I pulled 
> asterisk 13 from git, apply this patch which fixed RTP issue , but I think 
> tla transport issue came back for me . 
> 
> https://gerrit.asterisk.org/#/c/2346/
> 
> ​Oh, that one, OK.  ​  It should be merged now so if you 'git pull' on 13 
> now, you should get it.  The transport re-use issue was in pjproject so is it 
> possible that you're not compiling against the latest trunk?
> 
> 
> 
> 
>  
> 
> Thank you
> 
> Sent from my iPhone
> 
> On Mar 4, 2016, at 12:01 AM, George Joseph  
> wrote:
> 
>> 
>> 
>> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long  
>> wrote:
>> 
>> Thanks George I appreciate the info .  Being able to see what codec is in 
>> use for call in progress is very handy sometimes.
>> 
>> As far as the RTP stats goes,  I see there is some info with “rtp” and 
>> “rtcp” commands which can be useful for troubleshooting. A running tally of 
>> # packets or bandwidth used would be awesome in along with the codec in 
>> "pjsip show channels" or something like that.
>> 
>> 
>> Im not certain, but I think the TLS signalling problem from this email may 
>> be happening to me again after patching for another pjsip/NAT issue which 
>> was with the external_media_address not working and the internal IP being 
>> sent in the SDP from asterisk - I applied this patch to the codebase and 
>> recompiled I am seeing the TLS “new transport”  issue again , I think.
>> 
>> ​I've lost track of who's applying what patches to ​which codebase. :)
>> 
>> Which patch did you apply for "external_media_address not working"?
>> 
>>  
>> 
>> Regards,
>> 
>> Kevin Long
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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>

[asterisk-users] How to recive Incoming calls in Chan Dongle ?

[Vitor Mazuco]

Hi!

How can I setup my Chan Dongle recived calls in my Asterisk?

I have to setup in dongle.conf ? Or in extensions.conf?

And the code for recive I found this site
http://asterisk-service.com/page/chan-dongle-use

I have to To save Subscriber Number before?

See the error log in my Asterisk

pbx.c:6796 __ast_pbx_run: Channel 'Dongle/dongle1-01' sent to
invalid extension but no invalid handler:
context,exten,priority=URA,+5511965380290,1,Noop(),1


Thanks in advanced.

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[asterisk-users] Asterisk 13 Realtime MusicOnHold

[Carlos Chavez]

I am having a problem trying to use the realtime database for 
musiconhold for Asterisk 13.  Everything is setup and I can see the mapping:


===> musiconhold (db=general, table=musiconhold)

Only what is in the musiconhold.conf file appears in Asterisk and 
the database is completely ignored.  The database was created using 
alembic and the structure included with Asterisk 13.7.2.  We are using 
MariaDB on CentOS 7 and res_config_mysql for the realtime engine.  All 
other configurations (pjsip, iax, voicemail and queues) are working as 
expected from the database.  How can I debug this problem?


--
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Carlos Chávez
+52 (55)9116-91161


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Re: [asterisk-users] How to control host part of From: field content from the dialplan [SOLVED]

[Olivier]

2016-03-04 18:59 GMT+01:00 Richard Mudgett :

>
>
> On Fri, Mar 4, 2016 at 11:45 AM, Olivier  wrote:
>
>> Hello,
>>
>> I've read SIP Connect 2.0 draft lately.
>>
>> It mentions specific use if either of the following values is present in
>> the From: field of an INVITE message.
>> The values are:
>> sip:unavailable@unkown.invalid
>> sip:anonymous@anonymous.invalid
>>
>> I'm using Asterisk 13 and PJSIP.
>> Which dialplan function can I use if I want to send an outbound SIP call
>> with a From field matching the above values ?
>>
>> I've tried with :
>> Set(CALLERID(num)=unavailable@unkown.invalid)
>>
>> and got:
>> From: "Bob" > >;tag=d74792e3-f646-4dd9-90fe-e4dc62ea728d
>>
>
> That is currently not supported by chan_pjsip.  There is an issue [1] for
> it and a
> corresponding patch on gerrit [2] to resolve it.  The patch is actively
> being
> reviewed/updated to get it merged into the codebase.
>
> [1] https://issues.asterisk.org/jira/browse/ASTERISK-25791
> [2] https://gerrit.asterisk.org/#/c/2293/
>
> Richard
>


So, if the patch gets committed to trunk, then using the following should
do it :
CALLERID(pres)=unavailable
CALLERID(pres)=prohib

Thanks Richard for this prompt and valuable reply.


>
> --
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Re: [asterisk-users] How to control host part of From: field content from the dialplan

[Richard Mudgett]

On Fri, Mar 4, 2016 at 11:45 AM, Olivier  wrote:

> Hello,
>
> I've read SIP Connect 2.0 draft lately.
>
> It mentions specific use if either of the following values is present in
> the From: field of an INVITE message.
> The values are:
> sip:unavailable@unkown.invalid
> sip:anonymous@anonymous.invalid
>
> I'm using Asterisk 13 and PJSIP.
> Which dialplan function can I use if I want to send an outbound SIP call
> with a From field matching the above values ?
>
> I've tried with :
> Set(CALLERID(num)=unavailable@unkown.invalid)
>
> and got:
> From: "Bob"  >;tag=d74792e3-f646-4dd9-90fe-e4dc62ea728d
>

That is currently not supported by chan_pjsip.  There is an issue [1] for
it and a
corresponding patch on gerrit [2] to resolve it.  The patch is actively
being
reviewed/updated to get it merged into the codebase.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-25791
[2] https://gerrit.asterisk.org/#/c/2293/

Richard
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[asterisk-users] How to control host part of From: field content from the dialplan

[Olivier]

Hello,

I've read SIP Connect 2.0 draft lately.

It mentions specific use if either of the following values is present in
the From: field of an INVITE message.
The values are:
sip:unavailable@unkown.invalid
sip:anonymous@anonymous.invalid

I'm using Asterisk 13 and PJSIP.
Which dialplan function can I use if I want to send an outbound SIP call
with a From field matching the above values ?

I've tried with :
Set(CALLERID(num)=unavailable@unkown.invalid)

and got:
From: "Bob" ;tag=d74792e3-f646-4dd9-90fe-e4dc62ea728d

Regards
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Re: [asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

[Patrick Laimbock]

Hi Travis,

On 04-03-16 15:23, Ryan, Travis wrote:

I start asterisk 13.7.2 and it dies before I can rasterisk into it. I’ve
tried getting a coredump, but it doesn’t coredump.  I know there are a
lot of errors in the log below, but most of those just say it’ll not
load a module, and no big deal.

When launching from commandline (not service script) here is what happens.

http://pastebin.com/3GFe6fG9


Two things:

[Mar  3 15:19:37] WARNING[8439]: loader.c:553 load_dynamic_module: Error 
loading module 'res_monitor.so': 
/usr/lib/asterisk/modules/res_monitor.so: undefined symbol: __ast_beep_stop
[Mar  3 15:19:37] WARNING[8439]: loader.c:553 load_dynamic_module: Error 
loading module 'res_ari_events.so': 
/usr/lib/asterisk/modules/res_ari_events.so: undefined symbol: 
stasis_app_register_all


Undefined symbol errors are not good. Not sure why that's just a 
WARNING. Maybe something went wrong during the build? The output of the 
build should show you more information. In the mean time try disabling 
these two modules just to see if that clears up the problem.


[Mar  3 15:19:39]   == Parsing '/etc/asterisk/extensions.conf': Found
[Mar  3 15:19:39] WARNING[8439]: config.c:2228 config_text_file_load: 
Unterminated comment detected beginning on line 386


That needs fixing.

HTH,
Patrick


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Re: [asterisk-users] PJSIP signaling question

[George Joseph]

On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long 
wrote:

> Hi George the patch was from here , you wrote it I believe . I pulled
> asterisk 13 from git, apply this patch which fixed RTP issue , but I think
> tla transport issue came back for me .
>
> https://gerrit.asterisk.org/#/c/2346/
>

​Oh, that one, OK.  ​  It should be merged now so if you 'git pull' on 13
now, you should get it.  The transport re-use issue was in pjproject so is
it possible that you're not compiling against the latest trunk?






>
> Thank you
>
> Sent from my iPhone
>
> On Mar 4, 2016, at 12:01 AM, George Joseph 
> wrote:
>
>
>
> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long 
> wrote:
>
>>
>> Thanks George I appreciate the info .  Being able to see what codec is in
>> use for call in progress is very handy sometimes.
>>
>> As far as the RTP stats goes,  I see there is some info with “rtp” and
>> “rtcp” commands which can be useful for troubleshooting. A running tally of
>> # packets or bandwidth used would be awesome in along with the codec in
>> "pjsip show channels" or something like that.
>>
>>
>> Im not certain, but I think the TLS signalling problem from this email
>> may be happening to me again after patching for another pjsip/NAT issue
>> which was with the external_media_address not working and the internal IP
>> being sent in the SDP from asterisk - I applied this patch to the codebase
>> and recompiled I am seeing the TLS “new transport”  issue again , I think.
>>
>
> ​I've lost track of who's applying what patches to ​which codebase. :)
>
> Which patch did you apply for "external_media_address not working"?
>
>
>
>>
>> Regards,
>>
>> Kevin Long
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> _
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>   http://www.asterisk.org/hello
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>
>
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[asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

[Ryan, Travis]

I start asterisk 13.7.2 and it dies before I can rasterisk into it. I've tried 
getting a coredump, but it doesn't coredump.  I know there are a lot of errors 
in the log below, but most of those just say it'll not load a module, and no 
big deal.

When launching from commandline (not service script) here is what happens.


http://pastebin.com/3GFe6fG9


Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com
(765) 742-1102

We're not the IT departmentWe're the I-TEAM department!

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Re: [asterisk-users] Rasterisk freeze on 4G link [SOLVED]

[Olivier]

Thank you all for pointing me in the right direction.

Now I learned I have to care about MTU.

Best regards

2016-03-03 21:27 GMT+01:00 Toufic Khreish (Gmail) 
:

> Hello,
>
>
>
>
>
> You need to determine the correct MTU value by doing the following:
>
>
>
> ping www.google.com -f -l 1400 and you go up or down
>
>
>
> An example:
> 1440 Max packet size from Ping Test
> + 28 IP and ICMP headers
> 1468 is your optimum MTU Setting
>
>
>
> Reference : http://www.tp-link.fr/FAQ-190.html
>
>
>
> Once you determine the correct MTU you just set it on the VPN server, it
> will not break any connection the worst will be slow connection that is all.
>
>
>
> From my experience the MTU value for 3G was 1400.
>
>
>
> Best regards
>
> Toufic
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Duncan
> *Sent:* Thursday, March 03, 2016 9:02 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [asterisk-users] Rasterisk freeze on 4G link
>
>
>
> > With this new link, whenever I launch a vim, a nano or a rasterisk
> session, my terminal freezes (rasterisk) or remains empty (nano, vim).
>
> >
> > When a session is frozon, I can open a new one at the same so it
> excludes a basic connectivity loss.
> >
> Usually incorrect MTU gives you this effect. Use ping with MTU size set to
> test and find what works.
>
> I think its value: it's 1272, which amazes me.
>
>
> You will probably also break it with any large text dump eg cat
> /var/log/syslog will also do it
>
>
>
> Yes  "cat /var/log/syslog" also broke my console.
>
> Why would my console break because of inadequate MTU and other PC on the
> same location, seem unaffected ?
>
> Because, they most probably mostly use SMTP and HTTP ?
>
> Is possible to simulate a given MTU on a LAN to reproduce such freezes ?
>
> (the remote location is at the other side of the country and I would like
> to prepare things as much as possible).
>
>
>
>
>
> I think you need to go through a router or some device that can constrain
> the MTU. But live changing your server MTU should be straight forward as
> openvpn should try and reconnect, and you can change the server back. I
> haven't lost connectivity before with this
>
> Also the session is probably timed out rather than gone, in 10-15 mins
> maybe less it will come back (or does for me)
>
> Cheers Duncan
>
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Re: [asterisk-users] PJSIP signaling question

[Kevin Long]

Hi George the patch was from here , you wrote it I believe . I pulled asterisk 
13 from git, apply this patch which fixed RTP issue , but I think tla transport 
issue came back for me . 

https://gerrit.asterisk.org/#/c/2346/

Thank you

Sent from my iPhone

> On Mar 4, 2016, at 12:01 AM, George Joseph  
> wrote:
> 
> 
> 
>> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long  
>> wrote:
>> 
>> Thanks George I appreciate the info .  Being able to see what codec is in 
>> use for call in progress is very handy sometimes.
>> 
>> As far as the RTP stats goes,  I see there is some info with “rtp” and 
>> “rtcp” commands which can be useful for troubleshooting. A running tally of 
>> # packets or bandwidth used would be awesome in along with the codec in 
>> "pjsip show channels" or something like that.
>> 
>> 
>> Im not certain, but I think the TLS signalling problem from this email may 
>> be happening to me again after patching for another pjsip/NAT issue which 
>> was with the external_media_address not working and the internal IP being 
>> sent in the SDP from asterisk - I applied this patch to the codebase and 
>> recompiled I am seeing the TLS “new transport”  issue again , I think.
> 
> ​I've lost track of who's applying what patches to ​which codebase. :)
> 
> Which patch did you apply for "external_media_address not working"?
> 
>  
>> 
>> Regards,
>> 
>> Kevin Long
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>> _
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Re: [asterisk-users] PJSIP signaling question

[George Joseph]

On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long 
wrote:

>
> Thanks George I appreciate the info .  Being able to see what codec is in
> use for call in progress is very handy sometimes.
>
> As far as the RTP stats goes,  I see there is some info with “rtp” and
> “rtcp” commands which can be useful for troubleshooting. A running tally of
> # packets or bandwidth used would be awesome in along with the codec in
> "pjsip show channels" or something like that.
>
>
> Im not certain, but I think the TLS signalling problem from this email may
> be happening to me again after patching for another pjsip/NAT issue which
> was with the external_media_address not working and the internal IP being
> sent in the SDP from asterisk - I applied this patch to the codebase and
> recompiled I am seeing the TLS “new transport”  issue again , I think.
>

​I've lost track of who's applying what patches to ​which codebase. :)

Which patch did you apply for "external_media_address not working"?



>
> Regards,
>
> Kevin Long
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