Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

[Jonas Kellens]

Hello


you mean while placing a video call ? What info am I looking for in the 
debug output ?





Kind regards.

J.



On 21-04-17 12:28, Marcelo Terres wrote:

Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 20 April 2017 at 12:42, Jonas Kellens  wrote:

Hello

in sip.conf I have ;

videosupport=yes




Kind regards.

J.



On 20-04-17 13:09, Marcelo Terres wrote:

I suppose that you enable the video support on sip.conf, right?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 19 April 2017 at 13:18, Jonas Kellens  wrote:

Hello

using asterisk 1.8.32.3

I am not able to make a call with video support. I do not know what I am
missing to make this video call.

Codec h264 should be supported.


sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
  It does not indicate anything about your configuration.
  INTBINARY  HEX   TYPE   NAME
DESCRIPTION

---
1 (1 <<  0)(0x1)  audio   g723
(G.723.1)
2 (1 <<  1)(0x2)  audiogsm
(GSM)
4 (1 <<  2)(0x4)  audio   ulaw
(G.711 u-law)
8 (1 <<  3)(0x8)  audio   alaw
(G.711 A-law)
   16 (1 <<  4)   (0x10)  audio   g726aal2
(G.726 AAL2)
   32 (1 <<  5)   (0x20)  audio  adpcm
(ADPCM)
   64 (1 <<  6)   (0x40)  audio   slin
(16
bit Signed Linear PCM)
  128 (1 <<  7)   (0x80)  audio  lpc10
(LPC10)
  256 (1 <<  8)  (0x100)  audio   g729
(G.729A)
  512 (1 <<  9)  (0x200)  audio  speex
(SpeeX)
 1024 (1 << 10)  (0x400)  audio   ilbc
(iLBC)
 2048 (1 << 11)  (0x800)  audio   g726
(G.726 RFC3551)
 4096 (1 << 12) (0x1000)  audio   g722
(G722)
 8192 (1 << 13) (0x2000)  audio siren7
(ITU
G.722.1 (Siren7, licensed from Polycom))
16384 (1 << 14) (0x4000)  audiosiren14
(ITU
G.722.1 Annex C, (Siren14, licensed from Polycom))
32768 (1 << 15) (0x8000)  audio slin16
(16
bit Signed Linear PCM (16kHz))
65536 (1 << 16)(0x1)  image   jpeg
(JPEG
image)
   131072 (1 << 17)(0x2)  imagepng
(PNG
image)
   262144 (1 << 18)(0x4)  video   h261
(H.261 Video)
   524288 (1 << 19)(0x8)  video   h263
(H.263 Video)
  1048576 (1 << 20)   (0x10)  video  h263p
(H.263+ Video)
  2097152 (1 << 21)   (0x20)  video   h264
(H.264 Video)
  4194304 (1 << 22)   (0x40)  video  mpeg4
(MPEG4 Video)
  8388608 (1 << 23)   (0x80)  videounknown
(unknown)
 16777216 (1 << 24)  (0x100)  videounknown
(unknown)
 33554432 (1 << 25)  (0x200)   textunknown
(unknown)
 67108864 (1 << 26)  (0x400)   textred
(T.140 Realtime Text with redundancy)
134217728 (1 << 27)  (0x800)   text   t140
(Passthrough T.140 Realtime Text)
268435456 (1 << 28) (0x1000)   textunknown
(unknown)
536870912 (1 << 29) (0x2000)   textunknown
(unknown)
   1073741824 (1 << 30) (0x4000)  (unk)unknown
(unknown)
   2147483648 (1 << 31) (0x8000)  (unk)unknown
(unknown)
   4294967296 (1 << 32)(0x1)  audio   g719
(ITU
G.719)
   8589934592 (1 << 33)(0x2)  audiospeex16
(SpeeX 16khz)
  17179869184 (1 << 34)(0x4)  audiounknown
(unknown)
  34359738368 (1 << 35)(0x8)  audiounknown
(unknown)
  68719476736 (1 << 36)   (0x10)  audiounknown
(unknown)
 137438953472 (1 << 37)   (0x20)  audiounknown
(unknown)
 274877906944 (1 << 38)   (0x40)  audiounknown
(unknown)
 549755813888 (1 << 39)   (0x80)  audiounknown
(unknown)
1099511627776 (1 << 40)  (0x100)  

Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

[Marcelo Terres]

Did you try to activate DEBUG and set the verbosity to a higher level
(100?) to check what Asterisk tells you about?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 20 April 2017 at 12:42, Jonas Kellens  wrote:
> Hello
>
> in sip.conf I have ;
>
> videosupport=yes
>
>
>
>
> Kind regards.
>
> J.
>
>
>
> On 20-04-17 13:09, Marcelo Terres wrote:
>>
>> I suppose that you enable the video support on sip.conf, right?
>>
>> Regards,
>> Marcelo H. Terres 
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 19 April 2017 at 13:18, Jonas Kellens  wrote:
>>>
>>> Hello
>>>
>>> using asterisk 1.8.32.3
>>>
>>> I am not able to make a call with video support. I do not know what I am
>>> missing to make this video call.
>>>
>>> Codec h264 should be supported.
>>>
>>>
>>> sip*CLI> core show codecs
>>> Disclaimer: this command is for informational purposes only.
>>>  It does not indicate anything about your configuration.
>>>  INTBINARY  HEX   TYPE   NAME
>>> DESCRIPTION
>>>
>>> ---
>>>1 (1 <<  0)(0x1)  audio   g723
>>> (G.723.1)
>>>2 (1 <<  1)(0x2)  audiogsm
>>> (GSM)
>>>4 (1 <<  2)(0x4)  audio   ulaw
>>> (G.711 u-law)
>>>8 (1 <<  3)(0x8)  audio   alaw
>>> (G.711 A-law)
>>>   16 (1 <<  4)   (0x10)  audio   g726aal2
>>> (G.726 AAL2)
>>>   32 (1 <<  5)   (0x20)  audio  adpcm
>>> (ADPCM)
>>>   64 (1 <<  6)   (0x40)  audio   slin
>>> (16
>>> bit Signed Linear PCM)
>>>  128 (1 <<  7)   (0x80)  audio  lpc10
>>> (LPC10)
>>>  256 (1 <<  8)  (0x100)  audio   g729
>>> (G.729A)
>>>  512 (1 <<  9)  (0x200)  audio  speex
>>> (SpeeX)
>>> 1024 (1 << 10)  (0x400)  audio   ilbc
>>> (iLBC)
>>> 2048 (1 << 11)  (0x800)  audio   g726
>>> (G.726 RFC3551)
>>> 4096 (1 << 12) (0x1000)  audio   g722
>>> (G722)
>>> 8192 (1 << 13) (0x2000)  audio siren7
>>> (ITU
>>> G.722.1 (Siren7, licensed from Polycom))
>>>16384 (1 << 14) (0x4000)  audiosiren14
>>> (ITU
>>> G.722.1 Annex C, (Siren14, licensed from Polycom))
>>>32768 (1 << 15) (0x8000)  audio slin16
>>> (16
>>> bit Signed Linear PCM (16kHz))
>>>65536 (1 << 16)(0x1)  image   jpeg
>>> (JPEG
>>> image)
>>>   131072 (1 << 17)(0x2)  imagepng
>>> (PNG
>>> image)
>>>   262144 (1 << 18)(0x4)  video   h261
>>> (H.261 Video)
>>>   524288 (1 << 19)(0x8)  video   h263
>>> (H.263 Video)
>>>  1048576 (1 << 20)   (0x10)  video  h263p
>>> (H.263+ Video)
>>>  2097152 (1 << 21)   (0x20)  video   h264
>>> (H.264 Video)
>>>  4194304 (1 << 22)   (0x40)  video  mpeg4
>>> (MPEG4 Video)
>>>  8388608 (1 << 23)   (0x80)  videounknown
>>> (unknown)
>>> 16777216 (1 << 24)  (0x100)  videounknown
>>> (unknown)
>>> 33554432 (1 << 25)  (0x200)   textunknown
>>> (unknown)
>>> 67108864 (1 << 26)  (0x400)   textred
>>> (T.140 Realtime Text with redundancy)
>>>134217728 (1 << 27)  (0x800)   text   t140
>>> (Passthrough T.140 Realtime Text)
>>>268435456 (1 << 28) (0x1000)   textunknown
>>> (unknown)
>>>536870912 (1 << 29) (0x2000)   textunknown
>>> (unknown)
>>>   1073741824 (1 << 30) (0x4000)  (unk)unknown
>>> (unknown)
>>>   2147483648 (1 << 31) (0x8000)  (unk)unknown
>>> (unknown)
>>>   4294967296 (1 << 32)(0x1)  audio   g719
>>> (ITU
>>> G.719)
>>>   8589934592 (1 << 33)(0x2)  audiospeex16
>>> (SpeeX 16khz)
>>>  17179869184 (1 << 34)(0x4)  audiounknown
>>> (unknown)
>>>  34359738368 (1 << 35)(0x8)  audiounknown
>>> (unknown)
>>>  68719476736 (1 << 36)   (0x10)  audiounknown
>>> (unknown)
>>> 137438953472 (1 << 37)   (0x20)  audio 

Re: [asterisk-users] asterisk as non root

[Antony Stone]

On Thursday 20 April 2017 at 18:31:03, Atux Atux wrote:

> root@PBX: /var/www/html $ /etc/init.d/asterisk start
> [ ok ] Starting asterisk (via systemctl): asterisk.service.

I'm somewhat puzzled that your root-user prompt is "$"
instead of the more normal "#", but never mind...

> root@PBX: /var/www/html $ ps aux | grep asterisk
> asterisk  1007  0.7  2.3  67128 23748 ?Ssl  Apr19   8:49 
> /usr/sbin/asterisk -U asterisk -G asterisk

So, the first column of that output shows you that asterisk is
running as the user "asterisk".

On my Debian system I only have "-U asterisk" without the "-G asterisk".

> root  4186  0.0  0.1   4192  1992 pts/0S+   17:30   0:00 grep asterisk

...and the grep command was run by "root"

> root@PBX: /var/www/html $ /usr/sbin/asterisk –rx "sip show peers"
> Privilege escalation protection disabled!
> See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
> Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk
> -r' to connect.

Who does "ls -l" show you that file /var/run/asterisk/asterisk.ctl
is owned by?

On my machine it's:

srwxrwx--- 1 asterisk asterisk 0 Apr 11 10:32 /var/run/asterisk/asterisk.ctl


Antony.

-- 
There's a good theatrical performance about puns on in the West End.  It's a 
play on words.

   Please reply to the list;
 please *don't* CC me.

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

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] asterisk as non root

[Atux Atux]

root@PBX: /var/www/html $ /etc/init.d/asterisk start
[ ok ] Starting asterisk (via systemctl): asterisk.service.
root@PBX: /var/www/html $ ps aux | grep asterisk
asterisk  1007  0.7  2.3  67128 23748 ?Ssl  Apr19   8:49
/usr/sbin/asterisk -U asterisk -G asterisk
root  4186  0.0  0.1   4192  1992 pts/0S+   17:30   0:00 grep
asterisk
root@PBX: /var/www/html $ /usr/sbin/asterisk –rx "sip show peers"
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Asterisk already running on /var/run/asterisk/asterisk.ctl.  Use 'asterisk
-r' to connect.
root@PBX: /var/www/html $



On Thu, Apr 20, 2017 at 1:36 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote:
>
> > Hi. thanks a lot for your replies. I did stop the services and i did
> issued
> > the  the "chown" and "chmod" commands listed in the guide.
> > It is necessary to compile it, instead if using the apt-get version
> > What am i missing?
>
> Let's go back to basics for a moment - you say this is a Debian system -
> in my
> experience Debian already runs Asterisk as the "asterisk" user and not as
> root, so let's see what you have.
>
> 1. Start Asterisk (probably using "/etc/init.d/asterisk start", or maybe
> "service asterisk start")
>
> 2. Check who it's running as: "ps aux | grep asterisk"
>
>
> Antony.
>
>
> --
> What makes you think I know what I'm talking about?
> I just have more O'Reilly books than most people.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
> --
> _
> -- 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
>
-- 
_
-- 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] Asterisk 1.8.32.3 : no video (h.264)

[Jonas Kellens]

Hello

in sip.conf I have ;

videosupport=yes




Kind regards.

J.


On 20-04-17 13:09, Marcelo Terres wrote:

I suppose that you enable the video support on sip.conf, right?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 19 April 2017 at 13:18, Jonas Kellens  wrote:

Hello

using asterisk 1.8.32.3

I am not able to make a call with video support. I do not know what I am
missing to make this video call.

Codec h264 should be supported.


sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
 It does not indicate anything about your configuration.
 INTBINARY  HEX   TYPE   NAME
DESCRIPTION
---
   1 (1 <<  0)(0x1)  audio   g723
(G.723.1)
   2 (1 <<  1)(0x2)  audiogsm   (GSM)
   4 (1 <<  2)(0x4)  audio   ulaw
(G.711 u-law)
   8 (1 <<  3)(0x8)  audio   alaw
(G.711 A-law)
  16 (1 <<  4)   (0x10)  audio   g726aal2
(G.726 AAL2)
  32 (1 <<  5)   (0x20)  audio  adpcm
(ADPCM)
  64 (1 <<  6)   (0x40)  audio   slin   (16
bit Signed Linear PCM)
 128 (1 <<  7)   (0x80)  audio  lpc10
(LPC10)
 256 (1 <<  8)  (0x100)  audio   g729
(G.729A)
 512 (1 <<  9)  (0x200)  audio  speex
(SpeeX)
1024 (1 << 10)  (0x400)  audio   ilbc
(iLBC)
2048 (1 << 11)  (0x800)  audio   g726
(G.726 RFC3551)
4096 (1 << 12) (0x1000)  audio   g722
(G722)
8192 (1 << 13) (0x2000)  audio siren7   (ITU
G.722.1 (Siren7, licensed from Polycom))
   16384 (1 << 14) (0x4000)  audiosiren14   (ITU
G.722.1 Annex C, (Siren14, licensed from Polycom))
   32768 (1 << 15) (0x8000)  audio slin16   (16
bit Signed Linear PCM (16kHz))
   65536 (1 << 16)(0x1)  image   jpeg   (JPEG
image)
  131072 (1 << 17)(0x2)  imagepng   (PNG
image)
  262144 (1 << 18)(0x4)  video   h261
(H.261 Video)
  524288 (1 << 19)(0x8)  video   h263
(H.263 Video)
 1048576 (1 << 20)   (0x10)  video  h263p
(H.263+ Video)
 2097152 (1 << 21)   (0x20)  video   h264
(H.264 Video)
 4194304 (1 << 22)   (0x40)  video  mpeg4
(MPEG4 Video)
 8388608 (1 << 23)   (0x80)  videounknown
(unknown)
16777216 (1 << 24)  (0x100)  videounknown
(unknown)
33554432 (1 << 25)  (0x200)   textunknown
(unknown)
67108864 (1 << 26)  (0x400)   textred
(T.140 Realtime Text with redundancy)
   134217728 (1 << 27)  (0x800)   text   t140
(Passthrough T.140 Realtime Text)
   268435456 (1 << 28) (0x1000)   textunknown
(unknown)
   536870912 (1 << 29) (0x2000)   textunknown
(unknown)
  1073741824 (1 << 30) (0x4000)  (unk)unknown
(unknown)
  2147483648 (1 << 31) (0x8000)  (unk)unknown
(unknown)
  4294967296 (1 << 32)(0x1)  audio   g719   (ITU
G.719)
  8589934592 (1 << 33)(0x2)  audiospeex16
(SpeeX 16khz)
 17179869184 (1 << 34)(0x4)  audiounknown
(unknown)
 34359738368 (1 << 35)(0x8)  audiounknown
(unknown)
 68719476736 (1 << 36)   (0x10)  audiounknown
(unknown)
137438953472 (1 << 37)   (0x20)  audiounknown
(unknown)
274877906944 (1 << 38)   (0x40)  audiounknown
(unknown)
549755813888 (1 << 39)   (0x80)  audiounknown
(unknown)
   1099511627776 (1 << 40)  (0x100)  audiounknown
(unknown)
   219902322 (1 << 41)  (0x200)  audiounknown
(unknown)
   4398046511104 (1 << 42)  (0x400)  audiounknown
(unknown)
   8796093022208 (1 << 43)  (0x800)  audiounknown
(unknown)
  17592186044416 (1 << 44) (0x1000)  audiounknown
(unknown)
  35184372088832 (1 << 45) (0x2000)  audiounknown
(unknown)
  70368744177664 (1 << 46) (0x4000)  audiounknown
(unknown)
 140737488355328 (1 << 47) (0x8000)  audiotestlaw
(G.711 

Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

[Marcelo Terres]

I suppose that you enable the video support on sip.conf, right?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 19 April 2017 at 13:18, Jonas Kellens  wrote:
> Hello
>
> using asterisk 1.8.32.3
>
> I am not able to make a call with video support. I do not know what I am
> missing to make this video call.
>
> Codec h264 should be supported.
>
>
> sip*CLI> core show codecs
> Disclaimer: this command is for informational purposes only.
> It does not indicate anything about your configuration.
> INTBINARY  HEX   TYPE   NAME
> DESCRIPTION
> ---
>   1 (1 <<  0)(0x1)  audio   g723
> (G.723.1)
>   2 (1 <<  1)(0x2)  audiogsm   (GSM)
>   4 (1 <<  2)(0x4)  audio   ulaw
> (G.711 u-law)
>   8 (1 <<  3)(0x8)  audio   alaw
> (G.711 A-law)
>  16 (1 <<  4)   (0x10)  audio   g726aal2
> (G.726 AAL2)
>  32 (1 <<  5)   (0x20)  audio  adpcm
> (ADPCM)
>  64 (1 <<  6)   (0x40)  audio   slin   (16
> bit Signed Linear PCM)
> 128 (1 <<  7)   (0x80)  audio  lpc10
> (LPC10)
> 256 (1 <<  8)  (0x100)  audio   g729
> (G.729A)
> 512 (1 <<  9)  (0x200)  audio  speex
> (SpeeX)
>1024 (1 << 10)  (0x400)  audio   ilbc
> (iLBC)
>2048 (1 << 11)  (0x800)  audio   g726
> (G.726 RFC3551)
>4096 (1 << 12) (0x1000)  audio   g722
> (G722)
>8192 (1 << 13) (0x2000)  audio siren7   (ITU
> G.722.1 (Siren7, licensed from Polycom))
>   16384 (1 << 14) (0x4000)  audiosiren14   (ITU
> G.722.1 Annex C, (Siren14, licensed from Polycom))
>   32768 (1 << 15) (0x8000)  audio slin16   (16
> bit Signed Linear PCM (16kHz))
>   65536 (1 << 16)(0x1)  image   jpeg   (JPEG
> image)
>  131072 (1 << 17)(0x2)  imagepng   (PNG
> image)
>  262144 (1 << 18)(0x4)  video   h261
> (H.261 Video)
>  524288 (1 << 19)(0x8)  video   h263
> (H.263 Video)
> 1048576 (1 << 20)   (0x10)  video  h263p
> (H.263+ Video)
> 2097152 (1 << 21)   (0x20)  video   h264
> (H.264 Video)
> 4194304 (1 << 22)   (0x40)  video  mpeg4
> (MPEG4 Video)
> 8388608 (1 << 23)   (0x80)  videounknown
> (unknown)
>16777216 (1 << 24)  (0x100)  videounknown
> (unknown)
>33554432 (1 << 25)  (0x200)   textunknown
> (unknown)
>67108864 (1 << 26)  (0x400)   textred
> (T.140 Realtime Text with redundancy)
>   134217728 (1 << 27)  (0x800)   text   t140
> (Passthrough T.140 Realtime Text)
>   268435456 (1 << 28) (0x1000)   textunknown
> (unknown)
>   536870912 (1 << 29) (0x2000)   textunknown
> (unknown)
>  1073741824 (1 << 30) (0x4000)  (unk)unknown
> (unknown)
>  2147483648 (1 << 31) (0x8000)  (unk)unknown
> (unknown)
>  4294967296 (1 << 32)(0x1)  audio   g719   (ITU
> G.719)
>  8589934592 (1 << 33)(0x2)  audiospeex16
> (SpeeX 16khz)
> 17179869184 (1 << 34)(0x4)  audiounknown
> (unknown)
> 34359738368 (1 << 35)(0x8)  audiounknown
> (unknown)
> 68719476736 (1 << 36)   (0x10)  audiounknown
> (unknown)
>137438953472 (1 << 37)   (0x20)  audiounknown
> (unknown)
>274877906944 (1 << 38)   (0x40)  audiounknown
> (unknown)
>549755813888 (1 << 39)   (0x80)  audiounknown
> (unknown)
>   1099511627776 (1 << 40)  (0x100)  audiounknown
> (unknown)
>   219902322 (1 << 41)  (0x200)  audiounknown
> (unknown)
>   4398046511104 (1 << 42)  (0x400)  audiounknown
> (unknown)
>   8796093022208 (1 << 43)  (0x800)  audiounknown
> (unknown)
>  17592186044416 (1 << 44) (0x1000)  audiounknown
> (unknown)
>  35184372088832 (1 << 45) (0x2000)  audiounknown
> (unknown)
>  70368744177664 (1 << 46) (0x4000)  audiounknown
> (unknown)
> 140737488355328 (1 << 

Re: [asterisk-users] asterisk as non root

[Antony Stone]

On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote:

> Hi. thanks a lot for your replies. I did stop the services and i did issued
> the  the "chown" and "chmod" commands listed in the guide.
> It is necessary to compile it, instead if using the apt-get version
> What am i missing?

Let's go back to basics for a moment - you say this is a Debian system - in my 
experience Debian already runs Asterisk as the "asterisk" user and not as 
root, so let's see what you have.

1. Start Asterisk (probably using "/etc/init.d/asterisk start", or maybe 
"service asterisk start")

2. Check who it's running as: "ps aux | grep asterisk"


Antony.


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Re: [asterisk-users] asterisk as non root

[Atux Atux]

Hi. thanks a lot for your replies. I did stop the services and i did issued
the  the "chown" and "chmod" commands listed in the guide.
It is necessary to compile it, instead if using the apt-get version
What am i missing?



On Wed, Apr 19, 2017 at 10:47 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Wednesday 19 April 2017 at 18:48:29, Atux Atux wrote:
>
> > Hi.
> > Here is the output of the command
> >
> > root@pbx: ~ $  find / -name asterisk -exec ls -ld '{}' \;
> >
> > drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
> >
> > drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk
> >
> > -rwxr-xr-x 1 root root 9719880 Apr 19 17:27 /usr/src/asterisk-11.25.1/
> main/asterisk
> >
> > drwxrwxr-x 3 1013 users 4096 Apr 19 16:56 /usr/src/asterisk-11.25.1/
> include/asterisk
> >
> > -rwxr-xr-x 1 root root 9719880 Apr 19 17:32 /usr/sbin/asterisk
>
> Okay, those look reasonable to me - however I'm surprised at some which
> are missing:
>
> /var/log/asterisk
> /var/spool/asterisk
> /var/run/asterisk
>
> Did you *stop* Asterisk before trying to change it to run as non-root?
>
> I think Tzafrir Cohen's comments are very well worth following.
>
>
> Antony.
>
> --
> "There has always been an underlying argument that we should open up our
> source code more broadly. The fact is that we are learning from open source
> and we are opening our code more broadly through Shared Source.
>
> Is there value to providing source code? The answer is unequivocally yes."
>
>  - Jason Matusow, head of Microsoft's Shared Source Program, in response
> to leaks of Windows source code on the Internet.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
> --
> _
> -- 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
>
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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] asterisk as non root

[Antony Stone]

On Wednesday 19 April 2017 at 18:48:29, Atux Atux wrote:

> Hi.
> Here is the output of the command
> 
> root@pbx: ~ $  find / -name asterisk -exec ls -ld '{}' \;
>
> drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
>
> drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk
>
> -rwxr-xr-x 1 root root 9719880 Apr 19 17:27 
> /usr/src/asterisk-11.25.1/main/asterisk
>
> drwxrwxr-x 3 1013 users 4096 Apr 19 16:56 
> /usr/src/asterisk-11.25.1/include/asterisk
>
> -rwxr-xr-x 1 root root 9719880 Apr 19 17:32 /usr/sbin/asterisk

Okay, those look reasonable to me - however I'm surprised at some which are 
missing:

/var/log/asterisk
/var/spool/asterisk
/var/run/asterisk

Did you *stop* Asterisk before trying to change it to run as non-root?

I think Tzafrir Cohen's comments are very well worth following.


Antony.

-- 
"There has always been an underlying argument that we should open up our source 
code more broadly. The fact is that we are learning from open source 
and we are opening our code more broadly through Shared Source.

Is there value to providing source code? The answer is unequivocally yes."

 - Jason Matusow, head of Microsoft's Shared Source Program, in response to 
leaks of Windows source code on the Internet.

   Please reply to the list;
 please *don't* CC me.

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Re: [asterisk-users] asterisk as non root

[Atux Atux]

Hi.
Here is the output of the command

root@pbx: ~ $  find / -name asterisk -exec ls -ld '{}' \;
drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk
-rwxr-xr-x 1 root root 9719880 Apr 19 17:27
/usr/src/asterisk-11.25.1/main/asterisk
drwxrwxr-x 3 1013 users 4096 Apr 19 16:56
/usr/src/asterisk-11.25.1/include/asterisk
-rwxr-xr-x 1 root root 9719880 Apr 19 17:32 /usr/sbin/asterisk
root@pbx: ~ $


On Wed, Apr 19, 2017 at 5:03 PM, Tzafrir Cohen 
wrote:

> On Wed, Apr 19, 2017 at 04:44:39PM +0300, Atux Atux wrote:
> > hello there. i am running debian 8 in my swerver and i would like to run
> > asterisk as non root.
>
> The Asterisk package included with Debian already does that. Why not
> have a look at it?
>
> > i did follow the
> > https://www.voip-info.org/wiki-Asterisk+non-root without any success.
> when
> > i issue
> > root@PBX: ~ $ asterisk -U asterisk -G asterisk
>
> The options -U and -G are for the case of running Asterisk as root and
> having Asterisk change user and group afterwards. There are a number of
> options that only work that way (real-time priority, special socket
> permissions, IIRC).
>
> Alternatively you can use other mans to change to that user (--chuid or
> start-stop-daemon or User: and Group: in a systemd service file, or
> whatever). And then you don't need those options.
>
> > Privilege escalation protection disabled!
> > See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
>
> Read that text. But it is irrelevant for your situation.
>
> > Unable to access the running directory (Permission denied). Changing to
> '/'
> > for compatibility.
>
> /root is not accessible by the user asterisk. This is mostly harmless,
> but not if you want to have core files (see also -g) and maybe a few
> other minor things.
>
> > Asterisk already running on /var/run/asterisk/asterisk.ctl. Use
> 'asterisk
> > -r' to connect.
>
> Because you already ran that command before. Or already have the system
> copy of asterisk running. Or whatever.
>
> Reading error messages helps.
>
> --
>Tzafrir Cohen
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
> --
> _
> -- 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
>
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Re: [asterisk-users] asterisk as non root

[Tzafrir Cohen]

On Wed, Apr 19, 2017 at 04:44:39PM +0300, Atux Atux wrote:
> hello there. i am running debian 8 in my swerver and i would like to run
> asterisk as non root. 

The Asterisk package included with Debian already does that. Why not
have a look at it?

> i did follow the
> https://www.voip-info.org/wiki-Asterisk+non-root without any success. when
> i issue
> root@PBX: ~ $ asterisk -U asterisk -G asterisk

The options -U and -G are for the case of running Asterisk as root and
having Asterisk change user and group afterwards. There are a number of
options that only work that way (real-time priority, special socket
permissions, IIRC).

Alternatively you can use other mans to change to that user (--chuid or
start-stop-daemon or User: and Group: in a systemd service file, or
whatever). And then you don't need those options.

> Privilege escalation protection disabled!
> See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.

Read that text. But it is irrelevant for your situation.

> Unable to access the running directory (Permission denied). Changing to '/'
> for compatibility.

/root is not accessible by the user asterisk. This is mostly harmless,
but not if you want to have core files (see also -g) and maybe a few
other minor things.

> Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk
> -r' to connect.

Because you already ran that command before. Or already have the system
copy of asterisk running. Or whatever.

Reading error messages helps.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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Re: [asterisk-users] asterisk as non root

[Antony Stone]

On Wednesday 19 April 2017 at 15:44:39, Atux Atux wrote:

> hello there. i am running debian 8 in my swerver and i would like to run
> asterisk as non root. i did follow the
> https://www.voip-info.org/wiki-Asterisk+non-root without any success.

Did you do the very first step:

/etc/init.d/asterisk stop   ?

> when i issue
> root@PBX: ~ $ asterisk -U asterisk -G asterisk
> Privilege escalation protection disabled!
> See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
> Unable to access the running directory (Permission denied).

Did you do all the "chown" and "chmod" commands listed in those guidelines?

> Changing to '/' for compatibility.
> Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk
> -r' to connect.

Er, you can't change to running as non-root without stopping the existing 
(started by root) service first...

> root@PBX: ~ $
> 
> any ideas on how to fix that please?

Show us the output of:

# find / -name asterisk -exec ls -ld '{}' \;


Antony.

-- 
All matter in the Universe can be placed into one of two categories:

1. Things which need to be fixed.
2. Things which need to be fixed once you've had a few minutes to play with 
them.

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[asterisk-users] asterisk as non root

[Atux Atux]

hello there. i am running debian 8 in my swerver and i would like to run
asterisk as non root. i did follow the
https://www.voip-info.org/wiki-Asterisk+non-root without any success. when
i issue
root@PBX: ~ $ asterisk -U asterisk -G asterisk
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Unable to access the running directory (Permission denied). Changing to '/'
for compatibility.
Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk
-r' to connect.
root@PBX: ~ $


any ideas on how to fix that please?
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[asterisk-users] Asterisk 1.8.32.3 : no video (h.264)

[Jonas Kellens]

Hello

using asterisk 1.8.32.3

I am not able to make a call with video support. I do not know what I am 
missing to make this video call.


Codec h264 should be supported.


sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INTBINARY  HEX   TYPE NAME   
DESCRIPTION

---
  1 (1 <<  0)(0x1) audio   g723   
(G.723.1)

  2 (1 <<  1)(0x2) audiogsm   (GSM)
  4 (1 <<  2)(0x4) audio   ulaw   
(G.711 u-law)
  8 (1 <<  3)(0x8) audio   alaw   
(G.711 A-law)
 16 (1 <<  4)   (0x10)  audio g726aal2   
(G.726 AAL2)
 32 (1 <<  5)   (0x20) audio  adpcm   
(ADPCM)
 64 (1 <<  6)   (0x40) audio   slin   
(16 bit Signed Linear PCM)
128 (1 <<  7)   (0x80) audio  lpc10   
(LPC10)
256 (1 <<  8)  (0x100) audio   g729   
(G.729A)
512 (1 <<  9)  (0x200) audio  speex   
(SpeeX)
   1024 (1 << 10)  (0x400) audio   ilbc   
(iLBC)
   2048 (1 << 11)  (0x800) audio   g726   
(G.726 RFC3551)
   4096 (1 << 12) (0x1000) audio   g722   
(G722)
   8192 (1 << 13) (0x2000) audio siren7   
(ITU G.722.1 (Siren7, licensed from Polycom))
  16384 (1 << 14) (0x4000)  audio siren14   
(ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
  32768 (1 << 15) (0x8000) audio slin16   
(16 bit Signed Linear PCM (16kHz))
  65536 (1 << 16)(0x1) image   jpeg   
(JPEG image)
 131072 (1 << 17)(0x2) imagepng   
(PNG image)
 262144 (1 << 18)(0x4) video   h261   
(H.261 Video)
 524288 (1 << 19)(0x8) video   h263   
(H.263 Video)
1048576 (1 << 20)   (0x10) video  h263p   
(H.263+ Video)
2097152 (1 << 21)   (0x20) video   h264   
(H.264 Video)
4194304 (1 << 22)   (0x40) video  mpeg4   
(MPEG4 Video)
8388608 (1 << 23)   (0x80)  video unknown   
(unknown)
   16777216 (1 << 24)  (0x100)  video unknown   
(unknown)
   33554432 (1 << 25)  (0x200)   text unknown   
(unknown)
   67108864 (1 << 26)  (0x400) textred   
(T.140 Realtime Text with redundancy)
  134217728 (1 << 27)  (0x800) text   t140   
(Passthrough T.140 Realtime Text)
  268435456 (1 << 28) (0x1000)   text unknown   
(unknown)
  536870912 (1 << 29) (0x2000)   text unknown   
(unknown)
 1073741824 (1 << 30) (0x4000)  (unk) unknown   
(unknown)
 2147483648 (1 << 31) (0x8000)  (unk) unknown   
(unknown)
 4294967296 (1 << 32)(0x1) audio   g719   
(ITU G.719)
 8589934592 (1 << 33)(0x2)  audio speex16   
(SpeeX 16khz)
17179869184 (1 << 34)(0x4)  audio unknown   
(unknown)
34359738368 (1 << 35)(0x8)  audio unknown   
(unknown)
68719476736 (1 << 36)   (0x10)  audio unknown   
(unknown)
   137438953472 (1 << 37)   (0x20)  audio unknown   
(unknown)
   274877906944 (1 << 38)   (0x40)  audio unknown   
(unknown)
   549755813888 (1 << 39)   (0x80)  audio unknown   
(unknown)
  1099511627776 (1 << 40)  (0x100)  audio unknown   
(unknown)
  219902322 (1 << 41)  (0x200)  audio unknown   
(unknown)
  4398046511104 (1 << 42)  (0x400)  audio unknown   
(unknown)
  8796093022208 (1 << 43)  (0x800)  audio unknown   
(unknown)
 17592186044416 (1 << 44) (0x1000)  audio unknown   
(unknown)
 35184372088832 (1 << 45) (0x2000)  audio unknown   
(unknown)
 70368744177664 (1 << 46) (0x4000)  audio unknown   
(unknown)
140737488355328 (1 << 47) (0x8000)  audio testlaw   
(G.711 test-law)
281474976710656 (1 << 48)(0x1)  video unknown   
(unknown)
562949953421312 (1 << 49)(0x2)  video unknown   
(unknown)
   1125899906842624 (1 << 50)(0x4)  video unknown   
(unknown)
   2251799813685248 (1 << 51)(0x8)  video unknown   
(unknown)
   4503599627370496 (1 << 52)   (0x10)  video unknown   
(unknown)
   9007199254740992 (1 << 53)   (0x20)  video 

Re: [asterisk-users] Asterisk/FFA version upgrade recommendation

[Ludovic Gasc]

Hi,

We use with success on our production the builtin fax in Asterisk 13, based
on spandsp.
We have several lawyers that use this feature each day with no major issues
to my knowledge.
However, we have enabled T38 on the entire chain and we have a carrier that
handles T38 pretty well.

Before that, we had more frequent failures.

Yours.
--
Ludovic Gasc (GMLudo)
Lead Developer Architect at ALLOcloud
https://be.linkedin.com/in/ludovicgasc

2017-03-12 18:23 GMT+01:00 Mike Diehl :

> Hi all,
>
> I'm needing to upgrade Asterisk from 10.x to whatever the recommended
> version
> is that will allow me to continue to use Fax For Asterisk.
>
> I don't have many upgrade windows, I'd like to get the most bang for my
> buck,
> but I can't afford to be a beta tester on this server.
>
> The FFA site says that it's supported by Asterisk version 12 and lower, but
> version 12 doesn't seem to be supported.  Perhaps my information is
> outdated?
>
> Anyway, I can't go with the spandsp route because my system listens for AMA
> events that spandsp doesn't seem to produce and I can't emulate easily.
>
> Any recommendations would be very welcome.
>
> --
> Mike Diehl
>
>
>
>
> --
> _
> -- 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
>
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Re: [asterisk-users] asterisk-users Digest, Vol 152, Issue 31

[Mike Codjoe]

Dear Saint Michael,

I will be grateful if you could introduce me to the Company that
offers the translation service.

I am really interested in google voice.

Sincerely,

Michael Codjoe

On 29 March 2017 at 17:00,   wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Bounty on Google Voice (Saint Michael)
>
>
> --
>
> Message: 1
> Date: Wed, 29 Mar 2017 12:45:16 -0400
> From: Saint Michael 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: [asterisk-users] Bounty on Google Voice
> Message-ID:
> 

Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

[Matt Fredrickson]

On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins  wrote:

>
> On Fri, Apr 7, 2017 at 9:44 PM, Teijo  wrote:
>
>> Hello,
>>
>> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
>> problem until now which remained was that if dtls_rekey was set to the
>> value other than 0, call hanged up when using chrome after the time where
>> dtls_rekey was set.
>>
>> I suppose that "bad media description" shown in Chrome's window which
>> causes call to fail, has appeared with Chromes newer versions (currently 58
>> beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.
>>
>> Has somebody else encountered this problem, or more better resolved it?
>>
>> Best regards,
>>
>> Teijo
>>
>> --
>> _
>> -- 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
>>
>
>
> Hi Teijo
>
> Take a read of https://nimblea.pe/monkey-business/2017/01/19/webrtc-
> asterisk-and-chrome-57/ :)
>

13.15.0 should address rtcp-mux issues.

If there are still issues outstanding, it might be worth reporting a bug on
issues.asterisk.org.

Best wishes :-)

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

[Teijo]

Thank you Dan for this information.

Best regards,

Teijo

8.4.2017, 15:23, Dan Jenkins kirjoitti:


On Fri, Apr 7, 2017 at 9:44 PM, Teijo  wrote:


Hello,

I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
problem until now which remained was that if dtls_rekey was set to the
value other than 0, call hanged up when using chrome after the time where
dtls_rekey was set.

I suppose that "bad media description" shown in Chrome's window which
causes call to fail, has appeared with Chromes newer versions (currently 58
beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.

Has somebody else encountered this problem, or more better resolved it?

Best regards,

Teijo

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Hi Teijo

Take a read of
https://nimblea.pe/monkey-business/2017/01/19/webrtc-asterisk-and-chrome-57/
:)

Dan





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Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

[Dan Jenkins]

On Fri, Apr 7, 2017 at 9:44 PM, Teijo  wrote:

> Hello,
>
> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only
> problem until now which remained was that if dtls_rekey was set to the
> value other than 0, call hanged up when using chrome after the time where
> dtls_rekey was set.
>
> I suppose that "bad media description" shown in Chrome's window which
> causes call to fail, has appeared with Chromes newer versions (currently 58
> beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.
>
> Has somebody else encountered this problem, or more better resolved it?
>
> Best regards,
>
> Teijo
>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


Hi Teijo

Take a read of
https://nimblea.pe/monkey-business/2017/01/19/webrtc-asterisk-and-chrome-57/
:)

Dan
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[asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

[Teijo]

Hello,

I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only 
problem until now which remained was that if dtls_rekey was set to the 
value other than 0, call hanged up when using chrome after the time 
where dtls_rekey was set.


I suppose that "bad media description" shown in Chrome's window which 
causes call to fail, has appeared with Chromes newer versions (currently 
58 beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus.


Has somebody else encountered this problem, or more better resolved it?

Best regards,

Teijo

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[asterisk-users] Asterisk 13.15.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of
Asterisk 13.15.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.15.0 resolves several issues reported by the
community and would have not been possible without your participation.

*Thank you!*

The following issues are resolved in this release:

*New Features made in this release:*
---
- [ASTERISK-26878 ]
- func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)
- [ASTERISK-26863 ]
- res_pjsip: Add endpoint identification scheme based on a configured SIP
header/value
(Reported by Matt Jordan)
- [ASTERISK-17428 ]
- [patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)

*Bugs fixed in this release:*
---
- [ASTERISK-26851 ]
- res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26897 ]
- chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
- [ASTERISK-26916 ]
- res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-26705 ]
- libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-26850 ]
- res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)
- [ASTERISK-26484 ]
- res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
argument.
(Reported by Vinod Dharashive)
- [ASTERISK-26776 ]
- res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)
- [ASTERISK-26880 ]
- Asterisk crashes when multiple speex users join confbridge with pp_vad
and dtx enabled
(Reported by Kirsty Tyerman)
- [ASTERISK-26862 ]
- app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)
- [ASTERISK-26732 ]
- res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)
- [ASTERISK-26879 ]
- PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
- [ASTERISK-26867 ]
- autochan: Locking in a function ast_autochan_destroy() on destroyed
channel (after masquerade).
(Reported by Krzysztof Trempala)
- [ASTERISK-26869 ]
- res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s
extension
(Reported by Torrey Searle)
- [ASTERISK-26668 ]
- core: Malformed pattern matching extension (various factors) results in
crash
(Reported by xrobau)
- [ASTERISK-26865 ]
- chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)
- [ASTERISK-26872 ]
- Bundled pjproject fails to build when tarball downloaded with curl due to
md5 verification failure in Docker containers (or when there is no terminal)
(Reported by Matt Jordan)
- [ASTERISK-26717 ]
- Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)
- [ASTERISK-26643 ]
- Extra new line in Device field of DeviceStateChange AMI Event after
restart of Asterisk
(Reported by Roman Bedros)
- [ASTERISK-25237 ]
- stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)
- [ASTERISK-26857 ]
- chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
- [ASTERISK-26841 ]
- chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26822 

[asterisk-users] Asterisk 14.4.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of
Asterisk 14.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.4.0 resolves several issues reported by the
community and would have not been possible without your participation.

*Thank you!*

The following issues are resolved in this release:

*New Features made in this release:*
---
- [ASTERISK-26878 ]
- func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)
- [ASTERISK-26863 ]
- res_pjsip: Add endpoint identification scheme based on a configured SIP
header/value
(Reported by Matt Jordan)
- [ASTERISK-17428 ]
- [patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)

*Bugs fixed in this release:*
---
- [ASTERISK-26851 ]
- res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26897 ]
- chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
- [ASTERISK-26916 ]
- res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-26705 ]
- libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-26850 ]
- res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)
- [ASTERISK-26484 ]
- res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
argument.
(Reported by Vinod Dharashive)
- [ASTERISK-26776 ]
- res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)
- [ASTERISK-26880 ]
- Asterisk crashes when multiple speex users join confbridge with pp_vad
and dtx enabled
(Reported by Kirsty Tyerman)
- [ASTERISK-26862 ]
- app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)
- [ASTERISK-26732 ]
- res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)
- [ASTERISK-26879 ]
- PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
- [ASTERISK-26867 ]
- autochan: Locking in a function ast_autochan_destroy() on destroyed
channel (after masquerade).
(Reported by Krzysztof Trempala)
- [ASTERISK-26869 ]
- res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s
extension
(Reported by Torrey Searle)
- [ASTERISK-26668 ]
- core: Malformed pattern matching extension (various factors) results in
crash
(Reported by xrobau)
- [ASTERISK-26865 ]
- chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)
- [ASTERISK-26872 ]
- Bundled pjproject fails to build when tarball downloaded with curl due to
md5 verification failure in Docker containers (or when there is no terminal)
(Reported by Matt Jordan)
- [ASTERISK-26717 ]
- Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)
- [ASTERISK-26643 ]
- Extra new line in Device field of DeviceStateChange AMI Event after
restart of Asterisk
(Reported by Roman Bedros)
- [ASTERISK-25237 ]
- stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)
- [ASTERISK-26857 ]
- chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
- [ASTERISK-26841 ]
- chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26822 

Re: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID

[Motty Cruz]

Hello, I am trying to match SRC in CSV file to clid - Caller ID to user
extension number for stats purposes, however, in CSV file the SRC is the
company number as set in Extensions.conf 

 

exten => _7XXX,1,Set(CALLERID(number)="Company Inc" <3788001800>)

exten => _7XXX,2,Dial(SIP/voip1/13781${EXTEN:1},80)

exten => _7XXX,n,Congestion()

exten => _7XXX,n,Hangup()

 

how would I change it? I have look in cdr.conf and logger.conf

 

Thanks, 

 

From: Motty Cruz [mailto:motty.c...@gmail.com] 
Sent: Monday, April 03, 2017 3:52 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: motty.c...@gmail.com
Subject: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID
as CallerID

 

Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to  log the EXT ID. For
instance, the SRC in the following line should be my ext. number. Does it
make sense? From my extension 4007 I called 78079745, yet in the log below
the first number is 2318001800 which is the main company's number set in
Extensions.conf. 

 



2318001800

78079745

phones

"ITadmin" <2318001800>

SIP/4007-00015c0a

SIP/voip1-00015c0b

Dial

SIP/voip1/78079745,80

4/3/2017 15:30

4/3/2017 15:31

2

0

NO ANSWER

DOCUMENTATION

1.49E+09




 

 

Logger.conf

[general]

dateformat=%F %T

;

; Customize the display of debug message time stamps

; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)

;

; see strftime(3) Linux manual for format specifiers.  Note that there is
also

; a fractional second parameter which may be used in this field.  Use %1q

; for tenths, %2q for hundredths, etc.

;

;dateformat=%F %T   ; ISO 8601 date format

;dateformat=%F %T.%3q   ; with milliseconds

dateformat = %F %T.%3q   ; ISO 8601 date format with milliseconds

;

;

; This makes Asterisk write callids to log messages

; (defaults to yes)

use_callids = yes

 

 

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[asterisk-users] Asterisk 13.13-cert3, 13.14.1, 14.3.1 Now Available (Security Release)

[Asterisk Development Team]

The Asterisk Development Team has announced security releases for
Certified Asterisk
13.13 and Asterisk 13 and 14. The available security releases are released
as versions 13.13-cert3, 13.14.1, and 14.3.1.

These releases are available for immediate download at

http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of these versions resolves the following security
vulnerabilities:

* AST-2017-001: Buffer overflow in CDR's set user
  No size checking is done when setting the user field on a CDR. Thus,
  it is possible for someone to use an arbitrarily large string and write
past
  the end of the user field storage buffer. This allows the possibility of
remote
  code injection.

For a full list of changes in the current releases, please see the
ChangeLogs:

http://downloads.asterisk.org/pub/telephony/certified-
asterisk/releases/ChangeLog-13.13-cert3
http://downloads.asterisk.org/pub/telephony/asterisk/
releases/ChangeLog-13.14.1
http://downloads.asterisk.org/pub/telephony/asterisk/
releases/ChangeLog-14.3.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2017-001.pdf

Thank you for your continued support of Asterisk!
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[asterisk-users] Asterisk community servers are currently unavailable

[George Joseph]

We're investigating

-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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[asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID

[Motty Cruz]

Hello, In Master.csv Asterisk is loggin the Company ID set in
Extensions.conf, but I configured logger.conf to  log the EXT ID. For
instance, the SRC in the following line should be my ext. number. Does it
make sense? From my extension 4007 I called 78079745, yet in the log below
the first number is 2318001800 which is the main company's number set in
Extensions.conf. 

 



2318001800

78079745

phones

"ITadmin" <2318001800>

SIP/4007-00015c0a

SIP/voip1-00015c0b

Dial

SIP/voip1/78079745,80

4/3/2017 15:30

4/3/2017 15:31

2

0

NO ANSWER

DOCUMENTATION

1.49E+09




 

 

Logger.conf

[general]

dateformat=%F %T

;

; Customize the display of debug message time stamps

; this example is the ISO 8601 date format (-mm-dd HH:MM:SS)

;

; see strftime(3) Linux manual for format specifiers.  Note that there is
also

; a fractional second parameter which may be used in this field.  Use %1q

; for tenths, %2q for hundredths, etc.

;

;dateformat=%F %T   ; ISO 8601 date format

;dateformat=%F %T.%3q   ; with milliseconds

dateformat = %F %T.%3q   ; ISO 8601 date format with milliseconds

;

;

; This makes Asterisk write callids to log messages

; (defaults to yes)

use_callids = yes

 

 

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[asterisk-users] Asterisk crash when playing a WAV file to G722 SIP

[Richard Kenner]

I recently upgraded to Asterisk 14.3.0.  When playing a SIP file to a
G722 SIP channel (via chan_sip), I get a crash with the following
traceback.  This is reproducable:

#0  0x0036fdc30265 in raise () from /lib64/libc.so.6
#1  0x0036fdc31d10 in abort () from /lib64/libc.so.6
#2  0x0036fdc69beb in __libc_message () from /lib64/libc.so.6
#3  0x0036fdc7174f in _int_free () from /lib64/libc.so.6
#4  0x0036fdc75a4b in free () from /lib64/libc.so.6
#5  0x0050e19e in ast_frame_free (frame=0x5c35, cache=1) at frame.c:171
#6  0x00502bac in ast_readaudio_callback (s=0x6a5df88) at file.c:921
#7  0x00502d19 in ast_fsread_audio (data=0x5c35) at file.c:952
#8  0x004bb3df in __ast_read (chan=0x7ba68f8, dropaudio=0)
at channel.c:3848
#9  0x00504e51 in waitstream_core (c=0x7ba68f8, 
breakon=0x2b9630672bdb "", forward=0x5e56f8 "", reverse=0x5e56f8 "", 
skip_ms=0, audiofd=-1, cmdfd=-1, context=0x0, cb=0) at file.c:1602
#10 0x005053bf in ast_waitstream (c=0x5c35, 
breakon=0x5fca ) at file.c:1754
#11 0x2b963067272e in playback_exec (chan=0x7ba68f8, 

Does this "ring a bell" to anyone?  It looks like frame chainin has
gotten corrupted somehow, but this should be a straightforward case.


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[asterisk-users] Asterisk 13.14.0. Debugging DTMF issues

[Olivier]

Hello,

I'm working on a (PJ)SIP trunking Asterisk machine with which I'm facing
issues with DTMF.
Installed version is 13.14.0.


1. In outbound calls SDP, I'm seeing these kind of lines:
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

I would expect events to range from 0 to 15, not to 16, as seen in rfc 4733
examples.
What is this event 16 for ?
Is there a way to configure this ?



2. With channel originated calls using Local prefix, I can read this on
console:
[2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4103 __ast_read:
DTMF begin '#' received on PJSIP/Foo-006f
[2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4114 __ast_read:
DTMF begin passthrough '#' on PJSIP/Foo-006f
[2017-03-30 15:11:44] DTMF[11501][C-0057]: channel.c:4103 __ast_read:
DTMF begin '#' received on Local/2@from-originate-0024;1
[2017-03-30 15:11:44] DTMF[11501][C-0057]: channel.c:4114 __ast_read:
DTMF begin passthrough '#' on Local/2@from-originate-0024;1
[2017-03-30 15:11:44] DTMF[11505][C-0056]: channel.c:4017 __ast_read:
DTMF end '#' received on PJSIP/Foo-006f, duration 180 ms

With pure inbound-outbound calls (calls coming in from PJSIP and leaving
through PJSIP), I get this:
[2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4017 __ast_read:
DTMF end '9' received on PJSIP/Bar-IPO-0093, duration 100 ms
[2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4044 __ast_read:
DTMF begin emulation of '9' with duration 100 queued on
PJSIP/Bar-IPO-0093
[2017-03-30 15:51:01] DTMF[11650][C-006d]: channel.c:4181 __ast_read:
DTMF end emulation of '9' queued on PJSIP/Bar-IPO-0093


Can I get both inbound and outbound DTMF on console ? How ?


3. Looking at DTMF duration (as logged by Asterisk console), I can see that
some (from mobile phone) have a 180ms duration while some, from an other
SIP trunk, have a 100ms duration.

Can I configure tone duration ? How ?
Should I configure this ?
Does this duration any real relation with the way a user presses keys ?


Best regards
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Re: [asterisk-users] Asterisk 13: is CALLERID(num-pres) readable ? [SOLVED]

[Olivier]

Hello,

2017-03-17 15:05 GMT+01:00 Olivier :

> Hello,
>
> From a 13.14.0 system:
>
> same = n,Verbose(0,1-CALLERID(num-pres) is ${CALLERID(num-pres)})
> same = n,Set(CALLERID(num-pres)=prohib)
> same = n,Verbose(0,2-CALLERID(num-pres) is now ${CALLERID(num-pres)})
>


Replying to myself, moving above statements before a Ringing statement
triggered printing of expected values:
1-CALLERID(num-pres) is allowed_not_screened
2-CALLERID(num-pres) is now prohib

I hope this could help others.

Cheers



>
> I would expect to read "2-CALLERID(num-pres) is now prohib" but I get
> "2-CALLERID(num-pres) is ".
>
> I also get "1-CALLERID(num-pres) is ".
>
> 1. Am I missing something ?
> 2. What is the more efficient way to detect a caller requested to be kept
> anonymous ?
>
>
>
> Slts
>
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[asterisk-users] Asterisk 14.4.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 14.4.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.4.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
 * ASTERISK-26878 - func_channel: Add ability to get the callid so
  dialplan has access to it. (Reported by Richard Mudgett)
 * ASTERISK-26863 - res_pjsip: Add endpoint identification scheme
  based on a configured SIP header/value (Reported by Matt Jordan)
 * ASTERISK-17428 - [patch] Allow "Comedian Mail" branding to be
  removed (Reported by John Covert)

Bugs fixed in this release:
---
 * ASTERISK-26850 - res_hep_pjsip: Asterisk insert wrong protocol
  name in "Protocol ID" field in HEP packets (Reported by Max
  Norba)
 * ASTERISK-26484 - res_pjsip_messaging: Crash when using invalid
  URI in MessageSend 'from' argument. (Reported by Vinod
  Dharashive)
 * ASTERISK-26776 - res_pjsip_pubsub: Crash when generating xpidf
  content (Reported by Andrew Green)
 * ASTERISK-26880 - Asterisk crashes when multiple speex users join
  confbridge with pp_vad and dtx enabled (Reported by Kirsty
  Tyerman)
 * ASTERISK-26862 - app_queue: Queue stops calling members with
  local interface after forwarding in previous call (Reported by
  Robert Mordec)
 * ASTERISK-26732 - res_rtp_asterisk: Implement RTCP Multiplexing -
  breaking WebRTC in Chrome (Reported by Dan Jenkins)
 * ASTERISK-26879 - PJSIP external_media_address ignored if no
  local_net options are provided (Reported by Matt Jordan)
 * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use
  same IP as explicit transport (Reported by Richard Begg)
 * ASTERISK-26867 - autochan: Locking in a function
  ast_autochan_destroy() on destroyed channel (after masquerade).
  (Reported by Krzysztof Trempala)
 * ASTERISK-26869 - res_pjsip_refer: blind call transfer w/o a user
  name doesn't go to the s extension (Reported by Torrey Searle)
 * ASTERISK-26668 - core: Malformed pattern matching extension
  (various factors) results in crash (Reported by xrobau)
 * ASTERISK-26865 - chan_iax2: Reload of iax peer results in loss
  of host address/port (Reported by Richard Begg)
 * ASTERISK-26872 - Bundled pjproject fails to build when tarball
  downloaded with curl due to md5 verification failure in Docker
  containers (or when there is no terminal) (Reported by Matt
  Jordan)
 * ASTERISK-26717 - Document the fact that Asterisk HEP support
  only works with the PJSIP channel driver (Reported by Olivier
  Krief)
 * ASTERISK-26643 - Extra new line in Device field of
  DeviceStateChange AMI Event after restart of Asterisk (Reported
  by Roman Bedros)
 * ASTERISK-25237 -  stasis_cache.c:845 caching_topic_exec: -
  misleading ERROR message (Reported by Smirnov Aleksey)
 * ASTERISK-26857 - chan_pjsip: Dialplan function race condition
  (Reported by Joshua Colp)
 * ASTERISK-26841 - chan_sip: Call not cancelled after receiving a
  422 response (Reported by Jean Aunis - Prescom)
 * ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats
  shows wrong codec (Reported by Kevin Harwell)
 * ASTERISK-26353 - res_musiconhold: musiconhold seems to think
  that the general section is a class and issues warning (Reported
  by Jonathan Harris)
 * ASTERISK-26685 - res_pjsip: Crash when using IPv6 and Transport
  ws,wss (Reported by Michael Balen)
 * ASTERISK-24562 - app_voicemail: Cannot set fromstring on a
  per-mailbox basis (Reported by Mark Scholten)
 * ASTERISK-26598 - Saynumber is trying to get "and" from "digits/"
  subfolder (Reported by Jonathan Harris)
 * ASTERISK-17067 - Long lines in call files cause spurious syntax
  error (Reported by Dave Olszewski)
 * ASTERISK-26796 - res_pjsip_transport_websocket: Via header is
  'WS' when it should be 'WSS' (Reported by Jørgen H)
 * ASTERISK-25628 - res_config_pgsql: should match the behavior of
  other drivers so that queue_log can disable adaptive logging
  (Reported by Dmitry Wagin)
 * ASTERISK-26774 - core: Playback URL fails after some time
  (Reported by Igor Gamayunov)
 * ASTERISK-26825 - pjsip.conf.sample: user_agent: still refers to
  branch 12 (Reported by Tzafrir Cohen)
 * ASTERISK-26823 - PJSIP: Persistent subscriptions can cause
  FRACKs if endpoint does not exist (Reported by Mark Michelson)
 * ASTERISK-26623 - res_pjsip: Crash when calling PJSIPShowEndpoint
  (Reported by Jørgen H)
 * ASTERISK-26808 - res_pjsip_outbound_registration doesn't know
  about network change events (Reported by George Joseph)
 * ASTERISK-26705 - libasteriskssl.so not 

[asterisk-users] Asterisk 13.15.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 13.15.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.15.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
 * ASTERISK-26878 - func_channel: Add ability to get the callid so
  dialplan has access to it. (Reported by Richard Mudgett)
 * ASTERISK-26863 - res_pjsip: Add endpoint identification scheme
  based on a configured SIP header/value (Reported by Matt Jordan)
 * ASTERISK-17428 - [patch] Allow "Comedian Mail" branding to be
  removed (Reported by John Covert)

Bugs fixed in this release:
---
 * ASTERISK-26850 - res_hep_pjsip: Asterisk insert wrong protocol
  name in "Protocol ID" field in HEP packets (Reported by Max
  Norba)
 * ASTERISK-26484 - res_pjsip_messaging: Crash when using invalid
  URI in MessageSend 'from' argument. (Reported by Vinod
  Dharashive)
 * ASTERISK-26776 - res_pjsip_pubsub: Crash when generating xpidf
  content (Reported by Andrew Green)
 * ASTERISK-26880 - Asterisk crashes when multiple speex users join
  confbridge with pp_vad and dtx enabled (Reported by Kirsty
  Tyerman)
 * ASTERISK-26862 - app_queue: Queue stops calling members with
  local interface after forwarding in previous call (Reported by
  Robert Mordec)
 * ASTERISK-26732 - res_rtp_asterisk: Implement RTCP Multiplexing -
  breaking WebRTC in Chrome (Reported by Dan Jenkins)
 * ASTERISK-26879 - PJSIP external_media_address ignored if no
  local_net options are provided (Reported by Matt Jordan)
 * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use
  same IP as explicit transport (Reported by Richard Begg)
 * ASTERISK-26867 - autochan: Locking in a function
  ast_autochan_destroy() on destroyed channel (after masquerade).
  (Reported by Krzysztof Trempala)
 * ASTERISK-26869 - res_pjsip_refer: blind call transfer w/o a user
  name doesn't go to the s extension (Reported by Torrey Searle)
 * ASTERISK-26668 - core: Malformed pattern matching extension
  (various factors) results in crash (Reported by xrobau)
 * ASTERISK-26865 - chan_iax2: Reload of iax peer results in loss
  of host address/port (Reported by Richard Begg)
 * ASTERISK-26872 - Bundled pjproject fails to build when tarball
  downloaded with curl due to md5 verification failure in Docker
  containers (or when there is no terminal) (Reported by Matt
  Jordan)
 * ASTERISK-26717 - Document the fact that Asterisk HEP support
  only works with the PJSIP channel driver (Reported by Olivier
  Krief)
 * ASTERISK-26643 - Extra new line in Device field of
  DeviceStateChange AMI Event after restart of Asterisk (Reported
  by Roman Bedros)
 * ASTERISK-25237 -  stasis_cache.c:845 caching_topic_exec: -
  misleading ERROR message (Reported by Smirnov Aleksey)
 * ASTERISK-26857 - chan_pjsip: Dialplan function race condition
  (Reported by Joshua Colp)
 * ASTERISK-26841 - chan_sip: Call not cancelled after receiving a
  422 response (Reported by Jean Aunis - Prescom)
 * ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats
  shows wrong codec (Reported by Kevin Harwell)
 * ASTERISK-26685 - res_pjsip: Crash when using IPv6 and Transport
  ws,wss (Reported by Michael Balen)
 * ASTERISK-24562 - app_voicemail: Cannot set fromstring on a
  per-mailbox basis (Reported by Mark Scholten)
 * ASTERISK-26598 - Saynumber is trying to get "and" from "digits/"
  subfolder (Reported by Jonathan Harris)
 * ASTERISK-17067 - Long lines in call files cause spurious syntax
  error (Reported by Dave Olszewski)
 * ASTERISK-26796 - res_pjsip_transport_websocket: Via header is
  'WS' when it should be 'WSS' (Reported by Jørgen H)
 * ASTERISK-25628 - res_config_pgsql: should match the behavior of
  other drivers so that queue_log can disable adaptive logging
  (Reported by Dmitry Wagin)
 * ASTERISK-26825 - pjsip.conf.sample: user_agent: still refers to
  branch 12 (Reported by Tzafrir Cohen)
 * ASTERISK-26823 - PJSIP: Persistent subscriptions can cause
  FRACKs if endpoint does not exist (Reported by Mark Michelson)
 * ASTERISK-26623 - res_pjsip: Crash when calling PJSIPShowEndpoint
  (Reported by Jørgen H)
 * ASTERISK-26808 - res_pjsip_outbound_registration doesn't know
  about network change events (Reported by George Joseph)
 * ASTERISK-26313 - chan_sip : Asterisk restart seems to be
  required for changing encryption option (Reported by benasse)
 * ASTERISK-26705 - libasteriskssl.so not found when asterisk is
  installed for the 1st time (Reported by George Joseph)
 * ASTERISK-26781 - bridge: Passing the 'p' 

[asterisk-users] Asterisk 13: is CALLERID(num-pres) readable ?

[Olivier]

Hello,

>From a 13.14.0 system:

same = n,Verbose(0,1-CALLERID(num-pres) is ${CALLERID(num-pres)})
same = n,Set(CALLERID(num-pres)=prohib)
same = n,Verbose(0,2-CALLERID(num-pres) is now ${CALLERID(num-pres)})

I would expect to read "2-CALLERID(num-pres) is now prohib" but I get
"2-CALLERID(num-pres) is ".

I also get "1-CALLERID(num-pres) is ".

1. Am I missing something ?
2. What is the more efficient way to detect a caller requested to be kept
anonymous ?



Slts
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[asterisk-users] Asterisk crash in ast_find_ourip

[Patrick Wakano]

Hello list,

We've got an Asterisk crash in one of our servers and the core dump showed
following call tree.
Is this anyhow helpful to someone? Seems like a regular RTP / RTCP handling
that lead to a malloc crash

Grateful for any help!
Cheers,
Patrick

Thread 1 (Thread 0x7f8d6b023700 (LWP 14915)):
#0  0x7f8ead2d9252 in _int_malloc () from /lib64/libc.so.6
#1  0x7f8ead2da6b1 in malloc () from /lib64/libc.so.6
#2  0x7f8ead36f902 in make_request () from /lib64/libc.so.6
#3  0x7f8ead36fa5a in __check_pf () from /lib64/libc.so.6
#4  0x7f8ead332d17 in getaddrinfo () from /lib64/libc.so.6
#5  0x005435f5 in ast_sockaddr_resolve (addrs=0x7f8d6b01fee8,
str=0x7f8d6b01ffb0 "**", flags=768, family=0) at
netsock2.c:304
#6  0x0043484d in resolve_first (addr=0x7f8d6b022250,
name=0x7f8d6b01ffb0 "**", family=,
flag=768) at acl.c:792
#7  0x00434ce6 in ast_find_ourip (ourip=0x7f8d6b022250,
bindaddr=, family=0) at acl.c:970
#8  0x7f8e504f4d37 in ast_rtcp_read (instance=0x7f8e293df0f8) at
res_rtp_asterisk.c:4077
#9  0x7f8e504f5a45 in ast_rtp_read (instance=0x7f8e293df0f8,
rtcp=) at res_rtp_asterisk.c:4233
#10 0x7f8dfecdc7e1 in sip_rtp_read (ast=0x7f8e290f57c8) at
chan_sip.c:8298
#11 sip_read (ast=0x7f8e290f57c8) at chan_sip.c:8401
#12 0x004b44c5 in __ast_read (chan=0x7f8e290f57c8, dropaudio=0) at
channel.c:3874
#13 0x00476655 in bridge_handle_trip
(bridge_channel=0x7f8e28d669f8) at bridge_channel.c:2272
#14 bridge_channel_wait (bridge_channel=0x7f8e28d669f8) at
bridge_channel.c:2442
#15 0x00477658 in bridge_channel_internal_join
(bridge_channel=0x7f8e28d669f8) at bridge_channel.c:2587
#16 0x00468610 in bridge_channel_ind_thread (data=0x7f8e28d669f8)
at bridge.c:1690
#17 0x005bdbbb in dummy_start (data=) at
utils.c:1232
#18 0x7f8eae18d9d1 in start_thread () from /lib64/libpthread.so.0
#19 0x7f8ead3488fd in clone () from /lib64/libc.so.6



Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0  0x7f8ead2d9252 in _int_malloc () from /lib64/libc.so.6
#0  0x7f8ead2d9252 in _int_malloc () from /lib64/libc.so.6
No symbol table info available.
#1  0x7f8ead2da6b1 in malloc () from /lib64/libc.so.6
No symbol table info available.
#2  0x7f8ead36f902 in make_request () from /lib64/libc.so.6
No symbol table info available.
#3  0x7f8ead36fa5a in __check_pf () from /lib64/libc.so.6
No symbol table info available.
#4  0x7f8ead332d17 in getaddrinfo () from /lib64/libc.so.6
No symbol table info available.
#5  0x005435f5 in ast_sockaddr_resolve (addrs=0x7f8d6b01fee8,
str=0x7f8d6b01ffb0 "**", flags=768, family=0) at
netsock2.c:304
hints = {ai_flags = 0, ai_family = 0, ai_socktype = 2, ai_protocol
= 0, ai_addrlen = 0, ai_addr = 0x0, ai_canonname = 0x0, ai_next = 0x0}
res = 
ai = 
s = 0x7f8d6b01fe00 "**"
host = 0x7f8d6b01fe00 "**"
port = 0x0
e = 
i = 
res_cnt = 
__PRETTY_FUNCTION__ = "ast_sockaddr_resolve"
#6  0x0043484d in resolve_first (addr=0x7f8d6b022250,
name=0x7f8d6b01ffb0 "**", family=,
flag=768) at acl.c:792
addrs = 
addrs_cnt = 
#7  0x00434ce6 in ast_find_ourip (ourip=0x7f8d6b022250,
bindaddr=, family=0) at acl.c:970
ourhost = "**", '\000' 
root = {ss = {ss_family = 1384, __ss_align = 56, __ss_padding =
"\320\302\000\000\000\000\000\000\060M\177)\000\000\000\000\n\000\000\000\000\000\000\000\002",
'\000' "\300, ", '\000' ,
"\n\000\000\000\062\000\000\000[\000\000\00
0|\000\000\000w\000\000\000n", '\000' ,
"8\000\000\000\000\000\000"}, len = 49872}
res = 1795293104
port = 0
__PRETTY_FUNCTION__ = "ast_find_ourip"
#8  0x7f8e504f4d37 in ast_rtcp_read (instance=0x7f8e293df0f8) at
res_rtp_asterisk.c:4077
i = 
pt = 
length = 
rc = 
message_blob = 
rtcp_report = 0x7f8e2812c198
rtp = 0x7f8e285da140
addr = {ss = {ss_family = 2, __ss_align = 0, __ss_padding =
"0\000\000\000\060\000\000\000\220)\002k\215\177\000\000\320(\002k\215\177\000\000\340\323[\000\000\000\000\000H\360p(\216\177\000\000\021\260Q\000\000\000\000\000\245\006\000\000\216\177\000\000\356O0\255\216\177\000\000\020]\022)\216\177\000\000\000\000\000\000\000\000\000\000\245$\002k\215\177\000\000\000\000\000\000\216\177\000\000H\360p(\216\177\000\000\001\000\000\000\000\000\000"},
len = 16}
rtcpdata = '\000' , "\001", '\000' "\323,
[\000\000\000\000\000\201\310\000\fMm\216\347\334lf\256͑g\205Ms_\205\000\000\003\253\000\002J\340b{\370\363\000\000\000\000\000\000\225\310\000\000\000\017\000\000\000\000\000\000\000\000\201\312\000\aMm\216\347\001\023\065\065\063\067\

Re: [asterisk-users] Asterisk/FFA version upgrade recommendation

[Joshua Colp]

On Sun, Mar 12, 2017, at 02:23 PM, Mike Diehl wrote:
> Hi all,
> 
> I'm needing to upgrade Asterisk from 10.x to whatever the recommended
> version 
> is that will allow me to continue to use Fax For Asterisk.
> 
> I don't have many upgrade windows, I'd like to get the most bang for my
> buck, 
> but I can't afford to be a beta tester on this server.
> 
> The FFA site says that it's supported by Asterisk version 12 and lower,
> but 
> version 12 doesn't seem to be supported.  Perhaps my information is 
> outdated?
> 
> Anyway, I can't go with the spandsp route because my system listens for
> AMA 
> events that spandsp doesn't seem to produce and I can't emulate easily.
> 
> Any recommendations would be very welcome.

Asterisk 11 is the last supported version that would do FFA, but it's
only supported for security fixes. It will go totally EOL October 25th
of this year.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk/FFA version upgrade recommendation

[Mike Diehl]

Hi all,

I'm needing to upgrade Asterisk from 10.x to whatever the recommended version 
is that will allow me to continue to use Fax For Asterisk.

I don't have many upgrade windows, I'd like to get the most bang for my buck, 
but I can't afford to be a beta tester on this server.

The FFA site says that it's supported by Asterisk version 12 and lower, but 
version 12 doesn't seem to be supported.  Perhaps my information is 
outdated?

Anyway, I can't go with the spandsp route because my system listens for AMA 
events that spandsp doesn't seem to produce and I can't emulate easily.

Any recommendations would be very welcome.

-- 
Mike Diehl




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[asterisk-users] Asterisk/FFA version upgrade recommendation

[Mike Diehl]

Hi all,

I'm needing to upgrade Asterisk from 10.x to whatever the recommended version 
is that will allow me to continue to use Fax For Asterisk.

I don't have many upgrade windows, I'd like to get the most bang for my buck, 
but I can't afford to be a beta tester on this server.

The FFA site says that it's supported by Asterisk version 12 and lower, but 
version 12 doesn't seem to be supported.  Perhaps my information is 
outdated?

Anyway, I can't go with the spandsp route because my system listens for AMA 
events that spandsp doesn't seem to produce and I can't emulate easily.

Any recommendations would be very welcome.

-- 
Mike Diehl



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[asterisk-users] asterisk Channel 'IAX2' unable to transfer

[neu pat]

I've upgraded to to asterisk 11.25.1 (from 1.8).  My local asterisk is
showing it is registered with remote asterisk (same version),
But when I try to make a call I get:


iax2 show registry
Host  dnsmgr  UsernamePerceived Refresh  State
192.168.142.1:4569N   home_serve  192.168.142.7:4569
60  Registered

1 IAX2 registrations.
-- Accepted AUTHENTICATED TBD call from 10.0.0.108
-- Accepting DIAL from 10.0.0.108, formats = (ulaw)
-- Executing [4@internal:1] Dial("IAX2/iaxy-322-3730",
"IAX2/home_server:546987@192.168.141.1/4,30,rw") in new stack
-- Called IAX2/home_server:x@192.168.141.1/4
-- Call accepted by 192.168.141.1 (format ulaw)
-- Format for call is (ulaw)
-- IAX2/192.168.141.1:4569-83 answered IAX2/iaxy-322-3730
-- Channel 'IAX2/192.168.141.1:4569-83' unable to transfer
-- Channel 'IAX2/192.168.141.1:4569-83' unable to transfer
-- Hungup 'IAX2/192.168.141.1:4569-83'
  == Spawn extension (internal, 4, 1) exited non-zero on 'IAX2/iaxy-322-3730'
-- Hungup 'IAX2/iaxy-322-3730'

Is there something wrong with my dial plan:
exten => 4,1,Dial(IAX2/home_server:xx@${clinic_server}/${EXTEN},30,rw)

-- 
Regards,
Thelma

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Re: [asterisk-users] asterisk-users Digest, Vol 151, Issue 23

[Saint Michael]

Theory: The carrier is not responding with 100 Trying in the expected time.
Hence, Asterisk is sending the INVITE again.

On Wed, Feb 22, 2017 at 1:00 PM, 
wrote:

> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Looking for Speech Recognition (ASR) suggestions (Dan Cropp)
>2. multiple outbound invites (Jeff LaCoursiere)
>
>
> --
>
> Message: 1
> Date: Wed, 22 Feb 2017 15:43:56 +
> From: Dan Cropp 
> To: "asterisk-users@lists.digium.com"
> 
> Subject: [asterisk-users] Looking for Speech Recognition (ASR)
> suggestions
> Message-ID:
> <41223e927281d842a48cc18032b36cc30118faf...@mail2010c.amtelco.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Is it correct that the unimrcp is the best approach for Asterisk and
> ASR/TTS?
>
> Could anyone provide pros/cons for the various ASR options for Asterisk?
> We need the ability for very large grammars (over 100,000 options).
> Because of this, my initial thought is Nuance or Lumenvox.  Does this sound
> correct?
>
> Have a great day!
>
> Dan
> -- next part --
> An HTML attachment was scrubbed...
> URL:  attachments/20170222/371708a5/attachment-0001.html>
>
> --
>
> Message: 2
> Date: Wed, 22 Feb 2017 11:57:16 -0600
> From: Jeff LaCoursiere 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: [asterisk-users] multiple outbound invites
> Message-ID: 
> Content-Type: text/plain; charset=utf-8; format=flowed
>
>
> Hello,
>
> I have two upstream providers we use for US termination.  The dialplan
> sends calls out the "primary" and if that fails for specific reasons, it
> sends the same call out the "secondary". This has worked well for us
> when we are lazy about keeping balances up, for example.
>
> Starting a few days ago ALL calls sent to the 'primary' were returned as
> busy, though the secondary terminated them fine.  We have a balance, and
> funny enough international calls are going through fine, just not US
> calls.  I opened a ticket.
>
> The response form the carrier is that our asterisk is sending four
> simultaneous invites within one second, and for that reason the call is
> rejected.
>
> I did a packet trace and was able to confirm this is true - only US
> calls sent to this carrier cause our end to send four identical
> simultaneous invites.  When it fails, a single invite for the same call
> is sent to the secondary, which is terminated without issue.
>
> Happy to send the SIP trace if any would care to see it, but is there a
> reason anyone can think of that our asterisk (11.11.0) would suddenly
> start doing this?  It may be that it has been doing it all along, and
> our carrier just started rejected calls that come in this way, I'm not
> sure.
>
> Cheers,
>
> j
>
>
>
>
> --
>
> ___
> --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
>
> End of asterisk-users Digest, Vol 151, Issue 23
> ***
>
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[asterisk-users] Asterisk 14.3.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 14.3.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.3.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
 * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
  (Reported by Richard Mudgett)

Bugs fixed in this release:
---
 * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
  by nappsoft)
 * ASTERISK-26767 - ARI channelvars cause memory leak (Reported by
  Sébastien Duthil)
 * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be
  hung up via ARI (Reported by Tom Pawelek)
 * ASTERISK-26632 - core: Possibility of a frame "imbalance"
  leading to stuck channels. (Reported by Mark Michelson)
 * ASTERISK-25951 - res_agi:  run_agi eats frames it shouldn't
  (Reported by George Joseph)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
  manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26704 - res_odbc.conf contains deprecated
  configuration: 'pooling', 'shared_connections', 'limit', and
  'idlecheck' options were replaced by 'max_connections'.
  (Reported by Anthony Messina)
 * ASTERISK-26765 - res_resolver_unbound: FRACK! Excessive ref
  count trap tripped. (Reported by Richard Mudgett)
 * ASTERISK-21094 - MixMonitorMute mutes through stream if already
  slinear (e.g. Originate) (Reported by David Woolley)
 * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
  (Reported by Dmitriy)
 * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
  request  to endpoint (Reported by Ross Beer)
 * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
  wrong byte order on Intel platform when using slin codec
  (Reported by Frankie Chin)
 * ASTERISK-26754 - build_tools: make_build_h does not handle \ in
  user name  (Reported by Kirill Katsnelson)
 * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite:
  fwrite() returned error: Broken pipe" (Reported by Kirill
  Katsnelson)
 * ASTERISK-26755 - app_queue: Random queues disappear on "core
  reload queue all" (Reported by Kirill Katsnelson)
 * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups"
  after match in .conf has no effect (Reported by Michael Maier)
 * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support
  for SRV (Reported by Joshua Colp)
 * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
  does not work. (Reported by Richard Mudgett)
 * ASTERISK-26740 - voicemail API test: uses varlibdir instead of
  datadir for a sound file (Reported by Tzafrir Cohen)
 * ASTERISK-26739 - voicemail API test: confuses expected and
  actual values (Reported by Tzafrir Cohen)
 * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every
  sorcery memory cache populate (Reported by Ustinov Artem)
 * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
  (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
  (Reported by Aaron An)
 * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
  (Reported by Alexander Traud)
 * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND.
  (Reported by Alexander Traud)
 * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
  function around masquerade (Reported by Joshua Colp)
 * ASTERISK-26684 - res_pjsip: Various issues with compact SIP
  headers (Reported by Joshua Elson)
 * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
  Headers Enabled (Reported by JoshE)
 * ASTERISK-26672 - Crash when setting remote address on RTP
  instance (Reported by Richard Mudgett)
 * ASTERISK-26621 - app_queue: Queue application does not ring
  members with Local interface (Reported by Jonas Kellens)
 * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with
  MWI wasn't registered (Reported by Michael Kuron)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
  bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
  is invalid (Reported by Rusty Newton)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
  with frames leading to spammy log messages (Reported by Jonathan
  Rose)
 * ASTERISK-26653 - pjproject_bundled doesn't verify already
  downloaded tarballs (Reported by George Joseph)
 * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed,
  setting up new calls (Reported by Walter Doekes)
 * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
  line (Reported by Jørgen H)
 * 

[asterisk-users] Asterisk 13.14.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 13.14.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
---
 * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
  (Reported by Richard Mudgett)

Bugs fixed in this release:
---
 * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
  by nappsoft)
 * ASTERISK-26704 - res_odbc.conf contains deprecated
  configuration: 'pooling', 'shared_connections', 'limit', and
  'idlecheck' options were replaced by 'max_connections'.
  (Reported by Anthony Messina)
 * ASTERISK-21094 - MixMonitorMute mutes through stream if already
  slinear (e.g. Originate) (Reported by David Woolley)
 * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be
  hung up via ARI (Reported by Tom Pawelek)
 * ASTERISK-26632 - core: Possibility of a frame "imbalance"
  leading to stuck channels. (Reported by Mark Michelson)
 * ASTERISK-25951 - res_agi:  run_agi eats frames it shouldn't
  (Reported by George Joseph)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
  manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
  (Reported by Dmitriy)
 * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
  request  to endpoint (Reported by Ross Beer)
 * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
  wrong byte order on Intel platform when using slin codec
  (Reported by Frankie Chin)
 * ASTERISK-26754 - build_tools: make_build_h does not handle \ in
  user name  (Reported by Kirill Katsnelson)
 * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite:
  fwrite() returned error: Broken pipe" (Reported by Kirill
  Katsnelson)
 * ASTERISK-26755 - app_queue: Random queues disappear on "core
  reload queue all" (Reported by Kirill Katsnelson)
 * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups"
  after match in .conf has no effect (Reported by Michael Maier)
 * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support
  for SRV (Reported by Joshua Colp)
 * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
  does not work. (Reported by Richard Mudgett)
 * ASTERISK-26740 - voicemail API test: uses varlibdir instead of
  datadir for a sound file (Reported by Tzafrir Cohen)
 * ASTERISK-26739 - voicemail API test: confuses expected and
  actual values (Reported by Tzafrir Cohen)
 * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every
  sorcery memory cache populate (Reported by Ustinov Artem)
 * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
  (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
  (Reported by Aaron An)
 * ASTERISK-26672 - Crash when setting remote address on RTP
  instance (Reported by Richard Mudgett)
 * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
  (Reported by Alexander Traud)
 * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND.
  (Reported by Alexander Traud)
 * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
  function around masquerade (Reported by Joshua Colp)
 * ASTERISK-26684 - res_pjsip: Various issues with compact SIP
  headers (Reported by Joshua Elson)
 * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
  Headers Enabled (Reported by JoshE)
 * ASTERISK-26621 - app_queue: Queue application does not ring
  members with Local interface (Reported by Jonas Kellens)
 * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with
  MWI wasn't registered (Reported by Michael Kuron)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
  bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
  is invalid (Reported by Rusty Newton)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
  with frames leading to spammy log messages (Reported by Jonathan
  Rose)
 * ASTERISK-26653 - pjproject_bundled doesn't verify already
  downloaded tarballs (Reported by George Joseph)
 * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed,
  setting up new calls (Reported by Walter Doekes)
 * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
  line (Reported by Jørgen H)
 * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all
  aors (Reported by George Joseph)
 * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not
  Exist when transaction branch parameter 

[asterisk-users] asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)

[Motty Cruz]

Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My
extensions.conf file was mostly copied from server running Asterisk 1.8.
That being said! If I dial a number and get a busy signal I get the
following error: 

 

-- SIP/voipeer-084b redirecting info has changed, passing it to
SIP/1007-084a

-- SIP/voipeer-084b is busy

  == Everyone is busy/congested at this time (1:1/0/0)

-- Timeout on SIP/1007-084a

-- Executing [t@phones:1] Playback("SIP/1007-084a", "goodbye") in
new stack

   > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

--  Playing 'goodbye.slin' (language 'en')

   > 0x7f6a62146640 -- Probation passed - setting RTP source address to
191.96.18.41:62568

-- Executing [t@phones:2] Hangup("SIP/1007-084a", "") in new stack

 

Sip.conf 

[1007]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1007>
disallow=all
allow=ulaw
allow=alaw
username=1007
secret=X
dtmfmode=rfc2833
host=dynamic
mailbox=1007@default
nat=force_rport,comedia

 

Is it a codec issue? Or missed configuration? Asterisk does not know how to
translate busy signal. 

Your help is appreciated!

Thanks,

-- 
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Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

[George Joseph]

On Thu, Feb 2, 2017 at 4:06 AM, Zakir Mahomedy <z...@mayfair2000.com> wrote:

> Yes, from_user was set, removing those entries solved the problem.
>
> Can someone please explain to me the correct use for fromuser field?
>

from_user forces the user portion of the From header to a specific value on
calls that go TO the device represented by the endpoint.  Most often it's
used with a service provider when the service provider requires that all
calls it accepts have some sort of account identifier in the From header
instead of the original caller's info.  I can't think of a scenario where
you'd need to use from_user with a phone.


>
> thanks
> Zakir
>
>
> On Wednesday, February 1, 2017 8:00 PM, "asterisk-users-request@lists.
> digium.com" <asterisk-users-requ...@lists.digium.com> wrote:
>
>
> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>   1. asterisk  callerid issue PJSIP Realtime (Zakir Mahomedy)
>   2. Re: asterisk callerid issue PJSIP Realtime (George Joseph)
>
>
> --
>
> Message: 1
> Date: Wed, 1 Feb 2017 13:50:57 +0000 (UTC)
> From: Zakir Mahomedy <z...@mayfair2000.com>
> To: "asterisk-users@lists.digium.com"
> <asterisk-users@lists.digium.com>
> Subject: [asterisk-users] asterisk  callerid issue PJSIP Realtime
> Message-ID: <1998594554.250932.1485957057...@mail.yahoo.com>
> Content-Type: text/plain; charset="utf-8"
>
> I recently rolled out a new server with asterisk 14. ?On the Called user
> phone, the caller ID is the same as the Called User.
> eg) ext ?406 ?with callerid 406 ? calls ext 405 ,??on the caller id on the
> ext 405 phone displaying 405.
>
>
> We are using realtime PJSIP, I set the callerid field in the database but
> no luck.?
> - Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP
> CLID"") in new stack
> - Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID = ?"ross"
> <406>") in new stack- Executing [405@common:3] Dial("PJSIP/406-000f",
> "PJSIP/405") in new stack
> In the above dialplan, the callerid is been taken from the database PJSIP
> endpoints.?
> Here is the sip debugger files
> INVITE sip:405@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP 192.168.1.82:5060
> ;branch=z9hG4bK714210067;rportFrom: "zak" 
> <sip:406@192.168.1.27>;tag=2071662084To:
> <sip:405@192.168.1.27>Call-ID: 50172054-506...@bjc.bgi.B.ICCSeq: 21
> INVITEContact: "zak" <sip:406@192.168.1.82:5060>Authorization: Digest
> username="406", realm="asterisk", nonce="1485956409/
> e852b2a5e081f01421212d9a6ca954fa", uri="sip:405@192.168.1.27", response="
> ef94bae123f16dc5d9314a43922c949d", algorithm=md5, cnonce="13226017",
> opaque="50d490d233efd03e", qop=auth, nc=0003
>
> INVITE sip:405@192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP
> 197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
> <sip:405@192.168.1.27>;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To:  405@192.168.1.209;ob>Contact: <sip:405@197.245.99.113:5060>Call-ID:
> b4a83465-9105-4c70-9da1-11f410c37657
>
> <--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767
> --->SIP/2.0 180 RingingVia: SIP/2.0/UDP 197.245.99.113:5060;rport=
> 5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
> f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom: <sip:405@192.168.1.27>;tag=
> 77ea8869-273a-4f65-8128-e334b445f970To: <sip:405@192.168.1.209;ob>;
> tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221 INVITEContact:  405@192.168.1.209:36767;ob>Allow: PRACK, INVITE, ACK, B
>
>
> ?ParameterName ? ? ? ? ? ? ? ? ? ? ?: ParameterValue?===
> ==?callerid ? ? ? ? ? ? ? ? ? ? ?
> ? ? : "john doe" <405>?callerid_privacy ? ? ? ? ? ? : allowed?callerid_tag
> ? ? ? ? ? ? ? ? ? ?:
> Zakir
>
> -- next part --
> An HTML attachment was scrubbed...
> URL: <http://lists.digium.com/pipermail/asterisk-users/
> attachments/20170201/ede9ff18/attachment-0001.html>
>
> --
>
> Message: 2
> Date: Wed, 1 Feb 2017 08:52:59 -0700
> From: George Joseph <gjos...@digium.com>
> To: Zakir Mahomedy <z...@mayfair2000.com>,  Asterisk Users Mailing List
> - Non-Commercial Discussion <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] asterisk callerid issue PJSIP Realtime
> Message-ID:
> 

Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

[Zakir Mahomedy]

Yes, from_user was set, removing those entries solved the problem.
Can someone please explain to me the correct use for fromuser field?
thanksZakir 

On Wednesday, February 1, 2017 8:00 PM, 
"asterisk-users-requ...@lists.digium.com" 
<asterisk-users-requ...@lists.digium.com> wrote:
 

 Send asterisk-users mailing list submissions to
    asterisk-users@lists.digium.com

To subscribe or unsubscribe via the World Wide Web, visit
    http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
    asterisk-users-requ...@lists.digium.com

You can reach the person managing the list at
    asterisk-users-ow...@lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."


Today's Topics:

  1. asterisk  callerid issue PJSIP Realtime (Zakir Mahomedy)
  2. Re: asterisk callerid issue PJSIP Realtime (George Joseph)


--

Message: 1
Date: Wed, 1 Feb 2017 13:50:57 + (UTC)
From: Zakir Mahomedy <z...@mayfair2000.com>
To: "asterisk-users@lists.digium.com"
    <asterisk-users@lists.digium.com>
Subject: [asterisk-users] asterisk  callerid issue PJSIP Realtime
Message-ID: <1998594554.250932.1485957057...@mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"

I recently rolled out a new server with asterisk 14. ?On the Called user phone, 
the caller ID is the same as the Called User.
eg) ext ?406 ?with callerid 406 ? calls ext 405 ,??on the caller id on the ext 
405 phone displaying 405.


We are using realtime PJSIP, I set the callerid field in the database but no 
luck.?
- Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP CLID"") 
in new stack
- Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID = ?"ross" 
<406>") in new stack- Executing [405@common:3] Dial("PJSIP/406-000f", 
"PJSIP/405") in new stack
In the above dialplan, the callerid is been taken from the database PJSIP 
endpoints.?
Here is the sip debugger files
INVITE sip:405@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP 
192.168.1.82:5060;branch=z9hG4bK714210067;rportFrom: "zak" 
<sip:406@192.168.1.27>;tag=2071662084To: <sip:405@192.168.1.27>Call-ID: 
50172054-506...@bjc.bgi.B.ICCSeq: 21 INVITEContact: "zak" 
<sip:406@192.168.1.82:5060>Authorization: Digest username="406", 
realm="asterisk", nonce="1485956409/e852b2a5e081f01421212d9a6ca954fa", 
uri="sip:405@192.168.1.27", response="ef94bae123f16dc5d9314a43922c949d", 
algorithm=md5, cnonce="13226017", opaque="50d490d233efd03e", qop=auth, 
nc=0003

INVITE sip:405@192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP 
197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
 <sip:405@192.168.1.27>;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To: 
<sip:405@192.168.1.209;ob>Contact: <sip:405@197.245.99.113:5060>Call-ID: 
b4a83465-9105-4c70-9da1-11f410c37657

<--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767 --->SIP/2.0 
180 RingingVia: SIP/2.0/UDP 
197.245.99.113:5060;rport=5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
 f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom: 
<sip:405@192.168.1.27>;tag=77ea8869-273a-4f65-8128-e334b445f970To: 
<sip:405@192.168.1.209;ob>;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221 
INVITEContact: <sip:405@192.168.1.209:36767;ob>Allow: PRACK, INVITE, ACK, B


?ParameterName ? ? ? ? ? ? ? ? ? ? ?: 
ParameterValue?=?callerid
 ? ? ? ? ? ? ? ? ? ? ? ? ? : "john doe" <405>?callerid_privacy ? ? ? ? ? ? : 
allowed?callerid_tag ? ? ? ? ? ? ? ? ? ?:
Zakir
 
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Message: 2
Date: Wed, 1 Feb 2017 08:52:59 -0700
From: George Joseph <gjos...@digium.com>
To: Zakir Mahomedy <z...@mayfair2000.com>,  Asterisk Users Mailing List
    - Non-Commercial Discussion <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] asterisk callerid issue PJSIP Realtime
Message-ID:
    

Re: [asterisk-users] asterisk callerid issue PJSIP Realtime

[George Joseph]

On Wed, Feb 1, 2017 at 6:50 AM, Zakir Mahomedy  wrote:

> I recently rolled out a new server with asterisk 14.
> On the Called user phone, the caller ID is the same as the Called User.
>
> eg) ext  406  with callerid 406   calls ext 405 ,
>
> on the caller id on the ext 405 phone displaying 405.
>
>
>
> We are using realtime PJSIP, I set the callerid field in the database but
> no luck.
>
> - Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP
> CLID"") in new stack
> - Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID =  "ross"
> <406>") in new stack
> - Executing [405@common:3] Dial("PJSIP/406-000f", "PJSIP/405") in new
> stack
>
> In the above dialplan, the callerid is been taken from the database PJSIP
> endpoints.
>
> Here is the sip debugger files
>
> INVITE sip:405@192.168.1.27 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.82:5060;branch=z9hG4bK714210067;rport
> From: "zak" ;tag=2071662084
> To: 
> Call-ID: 50172054-506...@bjc.bgi.b.ic
> CSeq: 21 INVITE
> Contact: "zak" 
> Authorization: Digest username="406", realm="asterisk", nonce="1485956409/
> e852b2a5e081f01421212d9a6ca954fa", uri="sip:405@192.168.1.27", response="
> ef94bae123f16dc5d9314a43922c949d", algorithm=md5, cnonce="13226017",
> opaque="50d490d233efd03e", qop=auth, nc=0003
>
>
> INVITE sip:405@192.168.1.209:36767;ob SIP/2.0
> Via: SIP/2.0/UDP 197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-
> 49e1-b92d-7b4091b3138b
> From: ;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328
>


On 405's endpoiint, you're not forcing from_user to 405 are you?




> To: 
> Contact: 
> Call-ID: b4a83465-9105-4c70-9da1-11f410c37657
>
>
> <--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 197.245.99.113:5060;rport=5060;received=192.168.1.27;
> branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682
> Call-ID: f0b31a86-0ac3-47f0-8b13-487d54982e9b
> From: ;tag=77ea8869-273a-4f65-8128-e334b445f970
> To: ;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1d
> CSeq: 12221 INVITE
> Contact: 
> Allow: PRACK, INVITE, ACK, B
>
>
>
>  ParameterName  : ParameterValue
>  =
>  callerid   : "john doe" <405>
>  callerid_privacy : allowed
>  callerid_tag:
>
> Zakir
>
>
>
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> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
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>



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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[asterisk-users] asterisk callerid issue PJSIP Realtime

[Zakir Mahomedy]

I recently rolled out a new server with asterisk 14.  On the Called user phone, 
the caller ID is the same as the Called User.
eg) ext  406  with callerid 406   calls ext 405 ,  on the caller id on the ext 
405 phone displaying 405.


We are using realtime PJSIP, I set the callerid field in the database but no 
luck. 
- Executing [405@common:1] NoOp("PJSIP/406-000f", ""DEBUGGING PJSIP CLID"") 
in new stack
- Executing [405@common:2] NoOp("PJSIP/406-000f", "CALLERID =  "ross" 
<406>") in new stack- Executing [405@common:3] Dial("PJSIP/406-000f", 
"PJSIP/405") in new stack
In the above dialplan, the callerid is been taken from the database PJSIP 
endpoints. 
Here is the sip debugger files
INVITE sip:405@192.168.1.27 SIP/2.0Via: SIP/2.0/UDP 
192.168.1.82:5060;branch=z9hG4bK714210067;rportFrom: "zak" 
;tag=2071662084To: Call-ID: 
50172054-506...@bjc.bgi.B.ICCSeq: 21 INVITEContact: "zak" 
Authorization: Digest username="406", 
realm="asterisk", nonce="1485956409/e852b2a5e081f01421212d9a6ca954fa", 
uri="sip:405@192.168.1.27", response="ef94bae123f16dc5d9314a43922c949d", 
algorithm=md5, cnonce="13226017", opaque="50d490d233efd03e", qop=auth, 
nc=0003

INVITE sip:405@192.168.1.209:36767;ob SIP/2.0Via: SIP/2.0/UDP 
197.245.99.113:5060;rport;branch=z9hG4bKPj2f9d3dde-5ec4-49e1-b92d-7b4091b3138bFrom:
 ;tag=e4a0ecf6-c74e-4ab5-8438-bac5c073e328To: 
Contact: Call-ID: 
b4a83465-9105-4c70-9da1-11f410c37657

<--- Received SIP response (515 bytes) from UDP:192.168.1.209:36767 --->SIP/2.0 
180 RingingVia: SIP/2.0/UDP 
197.245.99.113:5060;rport=5060;received=192.168.1.27;branch=z9hG4bKPj70fb8ef9-d99c-4e5b-88a5-eecbf7dd7682Call-ID:
 f0b31a86-0ac3-47f0-8b13-487d54982e9bFrom: 
;tag=77ea8869-273a-4f65-8128-e334b445f970To: 
;tag=jurMewPN-95CgNyoQbhRCFpbH90hKw1dCSeq: 12221 
INVITEContact: Allow: PRACK, INVITE, ACK, B


 ParameterName                      : ParameterValue 
= callerid              
             : "john doe" <405> callerid_privacy             : allowed 
callerid_tag                    :
Zakir
 
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Re: [asterisk-users] Asterisk 13.13.1

[Olivier]

SIP packet loss is one thing, RTP packet loss is another one.
One does not necessarily imply the other though, of course, both may happen
for a common reason.

What about audio codecs ?
Is it possible to configure things so that you only have a single codec
enabled all over your system (trunks, phones, ...) ?
Do you still have audio issues with a single codec ?

2017-01-30 17:55 GMT+01:00 Motty Cruz <motty.c...@gmail.com>:

> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from
> here: http://downloads.asterisk.org/pub/telephony/asterisk/
> asterisk-13-current.tar.gz
>
>
>
> I continue to see errors like this:
>
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
> Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@
> 192.168.125.173 for seqno 109 (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
> Retransmission timeout reached on transmission 6e3dd238-911e2ac3-f1260152@
> 192.168.125.152 for seqno 103 (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
> Retransmission timeout reached on transmission
> ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical
> Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+
> Retransmissions
>
> Packet timed out after 32000ms with no response
>
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
> Retransmission timeout reached on transmission ef497d11-a81b1c00-8bfbd3bf@
> 192.168.1.244 for seqno 103 (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
>
>
> Before upgrading to this new server, Asterisk version 1.8 on  CentOS 5.9
> hardware on both servers were similar in CPU, Memory
>
>
>
> Any support on this matter is appreciated!
>
>
>
> Thanks,
> Motty
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *kambiz sharifi
> *Sent:* Saturday, January 28, 2017 5:13 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk 13.13.1
>
>
>
>
>
> On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4...@gmail.com> wrote:
>
> What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
> How did you installed Asterisk 1.8 and 13 ? From source or from package ?
>
> I would be curious to see what would happen after downgrading back to 1.8.
>
>
>
> 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.c...@gmail.com>:
>
> Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
> starting to complaint about packets loss, conversations are choppy!
>
>
>
>
> PkEI don’t even know where to start looking! Choppy conversations happened
> within users. I am using sip.conf
>
>
>
> [1091]
>
> type=friend
>
> context=sip-phone
>
> call-limit=2
>
> trustrpid=no
>
> callerid="dev1" <1091>
>
> disallow=all
>
> allow=ulaw
>
> allow=alaw
>
> username=1091
>
> secret=X
>
> dtmfmode=rfc2833
>
> host=dynamic
>
> mailbox=10091@default
>
> nat=force_rport,comedia
>
> canreinvite=no
>
>
>
> extensions.conf
>
> exten => 1091,hint,SIP/${EXTEN}
>
> exten => 1091,1,Dial(SIP/${EXTEN},15,t)
>
> exten => 1091,2,Voicemail(${EXTEN}@default,u)
>
> exten => 1091,102,Voicemail(${EXTEN}@default,b)
>
> exten => 1091,103,Hangup
>
>
>
> [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:
>
> Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@
> 192.168.0.191 for seqno 156 (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
>
>
> any ideas?
>
>
>
> Thanks!
>
> Motty
>
>
> --
>
>
> _
>
>
> -- 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 UNSUBSCR

Re: [asterisk-users] Asterisk 13.13.1

[Ron Wheeler]

CentOS 7 uses firewalld to control TCP amd UDP access.

The iptables configuration will be overwritten and dynamically changed 
by Firewalld so don't count on the old practice of manipulating iptables 
directly.


I recently moved our Asterisk from an old CentOS to CentOS 7 running 
FreePBX 14.0.1.beta2.


You can add a firewalld service yp /etc/firewalld/services like mine.
[root@firewall0 services]# cat Asterisk.xml


  asterisk
  Asterisk PBX
  
  
  
  
  


You then permit this service in your interface (zones) as a service
 

I also added a rule to get some logging on the Asterisk ports while 
getting things up and running.

  



  
  


I did this all on my exterior firewall which is also a CentOS 7 system.
On the Asterisk server, I do not block anything which is not a best 
practice but the entire internal network is very small and I consider it 
to be secure.


You (and I) should control the interface using Firewalld with the same 
service and zone specifications.






On 30/01/2017 12:13 PM, Motty Cruz wrote:

I thought it was a firewall issues. I disabled IP Tables & Selinux, but the
problem persist! I have not made changes on our firewall since the upgrade!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1


On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote:
Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from

here:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar
.gz


I continue to see errors like this:
[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:

Retransmission timeout reached on transmission
56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request)
-- See >>> >>>

Firewall?

Doug




--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102


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Re: [asterisk-users] Asterisk 13.13.1

[Michael Maier]

On 01/30/2017 at 05:55 PM Motty Cruz wrote:
> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: 
> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz
>
> 
>  
> 
> I continue to see errors like this: 
> 
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
> Retransmission timeout reached on transmission 
> 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) 
> -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> 
> Packet timed out after 32000ms with no response
> 
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
> Retransmission timeout reached on transmission 
> 6e3dd238-911e2ac3-f1260152@192.168.125.152 for seqno 103 (Critical Request) 
> -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> 
> Packet timed out after 32000ms with no response
> 
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
> Retransmission timeout reached on transmission 
> ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical Request) -- 
> See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> 
> Packet timed out after 32000ms with no response
> 
> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
> Retransmission timeout reached on transmission 
> ef497d11-a81b1c00-8bfbd3bf@192.168.1.244 for seqno 103 (Critical Request) -- 
> See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> 
>  
> 
> Before upgrading to this new server, Asterisk version 1.8 on  CentOS 5.9 
> hardware on both servers were similar in CPU, Memory 
> 
>  
> 
> Any support on this matter is appreciated!

Did you setup tcpdump (behind the machine) to see, if the packets really
leave the machine? Can you see any answer?


Regards,
Michael

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Re: [asterisk-users] Asterisk 13.13.1

[Motty Cruz]

I thought it was a firewall issues. I disabled IP Tables & Selinux, but the
problem persist! I have not made changes on our firewall since the upgrade!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 30, 2017 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1

>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote:
>>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from
here:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar
.gz 

>>> I continue to see errors like this: 
>>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:
Retransmission timeout reached on transmission
56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request)
-- See >>> >>> 

Firewall?

Doug

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Re: [asterisk-users] Asterisk 13.13.1

[Doug Lytle]

>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote:
>>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from 
>>> here: 
>>> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz
>>>  

>>> I continue to see errors like this: 
>>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
>>> Retransmission timeout reached on transmission 
>>> 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) 
>>> -- See >>> >>> 

Firewall?

Doug

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Re: [asterisk-users] Asterisk 13.13.1

[Motty Cruz]

Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: 
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz 
  

 

I continue to see errors like this: 

[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
Retransmission timeout reached on transmission 
56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
Retransmission timeout reached on transmission 
6e3dd238-911e2ac3-f1260152@192.168.125.152 for seqno 103 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
Retransmission timeout reached on transmission 
ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

[2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: 
Retransmission timeout reached on transmission 
ef497d11-a81b1c00-8bfbd3bf@192.168.1.244 for seqno 103 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

 

Before upgrading to this new server, Asterisk version 1.8 on  CentOS 5.9 
hardware on both servers were similar in CPU, Memory 

 

Any support on this matter is appreciated!

 

Thanks, 
Motty   

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kambiz sharifi
Sent: Saturday, January 28, 2017 5:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.13.1

 

 

On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4...@gmail.com> wrote:

What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
How did you installed Asterisk 1.8 and 13 ? From source or from package ?

I would be curious to see what would happen after downgrading back to 1.8.

 

2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.c...@gmail.com>:

Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are 
starting to complaint about packets loss, conversations are choppy!  

 

PkEI don’t even know where to start looking! Choppy conversations happened 
within users. I am using sip.conf

 

[1091]

type=friend

context=sip-phone

call-limit=2

trustrpid=no

callerid="dev1" <1091>

disallow=all

allow=ulaw

allow=alaw

username=1091

secret=X

dtmfmode=rfc2833

host=dynamic

mailbox=10091@default

nat=force_rport,comedia

canreinvite=no

 

extensions.conf

exten => 1091,hint,SIP/${EXTEN}

exten => 1091,1,Dial(SIP/${EXTEN},15,t)

exten => 1091,2,Voicemail(${EXTEN}@default,u)

exten => 1091,102,Voicemail(${EXTEN}@default,b)

exten => 1091,103,Hangup

 

[2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: 

Retransmission timeout reached on transmission 
7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) -- 
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

 

any ideas? 

 

Thanks!

Motty


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Re: [asterisk-users] Asterisk 13.13.1

[kambiz sharifi]

On Wed, Jan 25, 2017 at 16:00 Olivier  wrote:

> What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
> How did you installed Asterisk 1.8 and 13 ? From source or from package ?
>
> I would be curious to see what would happen after downgrading back to 1.8.
>
> 2017-01-24 21:03 GMT+01:00 Motty Cruz :
>
> Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
> starting to complaint about packets loss, conversations are choppy!
>
>
>
>
> PkEI don’t even know where to start looking! Choppy conversations happened
> within users. I am using sip.conf
>
>
>
> [1091]
>
> type=friend
>
> context=sip-phone
>
> call-limit=2
>
> trustrpid=no
>
> callerid="dev1" <1091>
>
> disallow=all
>
> allow=ulaw
>
> allow=alaw
>
> username=1091
>
> secret=X
>
> dtmfmode=rfc2833
>
> host=dynamic
>
> mailbox=10091@default
>
> nat=force_rport,comedia
>
> canreinvite=no
>
>
>
> extensions.conf
>
> exten => 1091,hint,SIP/${EXTEN}
>
> exten => 1091,1,Dial(SIP/${EXTEN},15,t)
>
> exten => 1091,2,Voicemail(${EXTEN}@default,u)
>
> exten => 1091,102,Voicemail(${EXTEN}@default,b)
>
> exten => 1091,103,Hangup
>
>
>
> [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:
>
> Retransmission timeout reached on transmission
> 7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request)
> -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
>
>
> any ideas?
>
>
>
> Thanks!
>
> Motty
>
> --
>
>
> _
>
>
> -- 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
>
>
>
>
> --
>
> _
>
> -- 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
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Re: [asterisk-users] asterisk-users Digest, Vol 150, Issue 17

[Henrique L.]

hi,

Do you edit your

voicemail.conf?
[default]
1091=(number to access your voicemail in your phone ex: 1234)





2017-01-25 16:00 GMT-02:00 <asterisk-users-requ...@lists.digium.com>:

> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
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> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1.  Asterisk 13.13.1 (Motty Cruz)
>2. Re: Asterisk 13.13.1 (Olivier)
>
>
> --
>
> Message: 1
> Date: Tue, 24 Jan 2017 12:03:05 -0800
> From: "Motty Cruz" <motty.c...@gmail.com>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users@lists.digium.com>
> Subject: [asterisk-users]  Asterisk 13.13.1
> Message-ID: <5887b2fb.86c5620a.8e94a.d...@mx.google.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
> starting to complaint about packets loss, conversations are choppy!
>
>
>
>
> I don't even know where to start looking! Choppy conversations happened
> within users. I am using sip.conf
>
>
>
> [1091]
>
> type=friend
>
> context=sip-phone
>
> call-limit=2
>
> trustrpid=no
>
> callerid="dev1" <1091>
>
> disallow=all
>
> allow=ulaw
>
> allow=alaw
>
> username=1091
>
> secret=X
>
> dtmfmode=rfc2833
>
> host=dynamic
>
> mailbox=10091@default
>
> nat=force_rport,comedia
>
> canreinvite=no
>
>
>
> extensions.conf
>
> exten => 1091,hint,SIP/${EXTEN}
>
> exten => 1091,1,Dial(SIP/${EXTEN},15,t)
>
> exten => 1091,2,Voicemail(${EXTEN}@default,u)
>
> exten => 1091,102,Voicemail(${EXTEN}@default,b)
>
> exten => 1091,103,Hangup
>
>
>
> [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:
>
> Retransmission timeout reached on transmission
> 7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request)
> --
> See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
>
>
> any ideas?
>
>
>
> Thanks!
>
> Motty
>
> -- next part --
> An HTML attachment was scrubbed...
> URL: <http://lists.digium.com/pipermail/asterisk-users/
> attachments/20170124/e9731841/attachment-0001.html>
>
> --
>
> Message: 2
> Date: Wed, 25 Jan 2017 13:30:00 +0100
> From: Olivier <oza.4...@gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Asterisk 13.13.1
> Message-ID:
> <CAPeT9jjqkD3nWJG5j9vq8sDgsKmiQpGi0ENDKXYhwfGaVzTv2w@mail.
> gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
> How did you installed Asterisk 1.8 and 13 ? From source or from package ?
>
> I would be curious to see what would happen after downgrading back to 1.8.
>
> 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.c...@gmail.com>:
>
> > Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users
> are
> > starting to complaint about packets loss, conversations are choppy!
> >
> >
> >
> >
> > I don?t even know where to start looking! Choppy conversations happened
> > within users. I am using sip.conf
> >
> >
> >
> > [1091]
> >
> > type=friend
> >
> > context=sip-phone
> >
> > call-limit=2
> >
> > trustrpid=no
> >
> > callerid="dev1" <1091>
> >
> > disallow=all
> >
> > allow=ulaw
> >
> > allow=alaw
> >
> > username=1091
> >
> > secret=X
> >
> > dtmfmode=rfc2833
> >
> > host=dynamic
> >
> > mailbox=10091@default
> >
> > nat=force_rport,comedia
> >
> > canreinvite=no
> >
> >
> >
> > extensions.conf
> >
> > exten => 1091,hint,SIP/${EXTEN}
> >
> > exten => 1091,1,Dia

Re: [asterisk-users] Asterisk 13.13.1

[Olivier]

What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
How did you installed Asterisk 1.8 and 13 ? From source or from package ?

I would be curious to see what would happen after downgrading back to 1.8.

2017-01-24 21:03 GMT+01:00 Motty Cruz :

> Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
> starting to complaint about packets loss, conversations are choppy!
>
>
>
>
> I don’t even know where to start looking! Choppy conversations happened
> within users. I am using sip.conf
>
>
>
> [1091]
>
> type=friend
>
> context=sip-phone
>
> call-limit=2
>
> trustrpid=no
>
> callerid="dev1" <1091>
>
> disallow=all
>
> allow=ulaw
>
> allow=alaw
>
> username=1091
>
> secret=X
>
> dtmfmode=rfc2833
>
> host=dynamic
>
> mailbox=10091@default
>
> nat=force_rport,comedia
>
> canreinvite=no
>
>
>
> extensions.conf
>
> exten => 1091,hint,SIP/${EXTEN}
>
> exten => 1091,1,Dial(SIP/${EXTEN},15,t)
>
> exten => 1091,2,Voicemail(${EXTEN}@default,u)
>
> exten => 1091,102,Voicemail(${EXTEN}@default,b)
>
> exten => 1091,103,Hangup
>
>
>
> [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt:
>
> Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@
> 192.168.0.191 for seqno 156 (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
> Packet timed out after 32000ms with no response
>
>
>
> any ideas?
>
>
>
> Thanks!
>
> Motty
>
> --
> _
> -- 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
>
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New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 13.13.1

[Motty Cruz]

Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
starting to complaint about packets loss, conversations are choppy!


 

I don't even know where to start looking! Choppy conversations happened
within users. I am using sip.conf

 

[1091]

type=friend

context=sip-phone

call-limit=2

trustrpid=no

callerid="dev1" <1091>

disallow=all

allow=ulaw

allow=alaw

username=1091

secret=X

dtmfmode=rfc2833

host=dynamic

mailbox=10091@default

nat=force_rport,comedia

canreinvite=no

 

extensions.conf

exten => 1091,hint,SIP/${EXTEN}

exten => 1091,1,Dial(SIP/${EXTEN},15,t)

exten => 1091,2,Voicemail(${EXTEN}@default,u)

exten => 1091,102,Voicemail(${EXTEN}@default,b)

exten => 1091,103,Hangup

 

[2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: 

Retransmission timeout reached on transmission
7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) --
See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response

 

any ideas? 

 

Thanks!

Motty

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

[Joshua Colp]

On Tue, Jan 24, 2017, at 01:41 PM, Dan Cropp wrote:
> Thank you Joshua.
> 
> So there is no way to retrieve header information which may come in on
> subsequent packages?
> 
> If not, is there any way to make an Attended Transfer following the
> RFC5589?
> https://tools.ietf.org/html/rfc5589
> 
> Asking because we have a hospital with a Cisco switch.  Hospital has two
> calls from their Cisco switch into an Asterisk box.  Operator handling
> the two calls and needs to transfer Call A to be connected to call B. 
> Can obviously be patched inside of Asterisk.  However, the hospital wants
> the call to be Attended Transferred. Basically, we need to send the
> Transfer (REFER) with the Replaces containing the call ID, From tag, and
> the To Tag.
> 
> I am able to gather everything needed for the REFER field and pass that
> to the Transfer command (via AMI), except the To tag. 

There isn't that I can think of. Even if you are able to construct such
a REFER I'm not sure what exactly will happen inside of Asterisk with
the legs.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

[Dan Cropp]

Thank you Joshua.

So there is no way to retrieve header information which may come in on 
subsequent packages?

If not, is there any way to make an Attended Transfer following the RFC5589?
https://tools.ietf.org/html/rfc5589

Asking because we have a hospital with a Cisco switch.  Hospital has two calls 
from their Cisco switch into an Asterisk box.  Operator handling the two calls 
and needs to transfer Call A to be connected to call B.  Can obviously be 
patched inside of Asterisk.  However, the hospital wants the call to be 
Attended Transferred. Basically, we need to send the Transfer (REFER) with the 
Replaces containing the call ID, From tag, and the To Tag.

I am able to gather everything needed for the REFER field and pass that to the 
Transfer command (via AMI), except the To tag. 

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Tuesday, January 24, 2017 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to 
retrieve a PJSIP header To field for the SIP OK response to Trying?

On Tue, Jan 24, 2017, at 01:25 PM, Dan Cropp wrote:
> I place a call into Asterisk (from SIP phone) and the To header does 
> not have a tag.  Asterisk then sends it's Trying response, still no 
> tag in the To header.  The phone then replies with OK, this time the 
> To header includes a tag.
> 
> Is there any way to retrieve this response To header (including the 
> tag
> field) from the dial plan?
> I have tried the PJSIP-HEADER read of the To header, but it seems to 
> only have access to the initial To header.
> I even tried reading multiple layers of the To header, but it still 
> didn't retrieve the newer dialog To headers.

The PJSIP_HEADER dialplan function currently only looks at the initial message. 
It does not allow access to subsequent ones.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
www.digium.com & www.asterisk.org

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

[Joshua Colp]

On Tue, Jan 24, 2017, at 01:25 PM, Dan Cropp wrote:
> I place a call into Asterisk (from SIP phone) and the To header does not
> have a tag.  Asterisk then sends it's Trying response, still no tag in
> the To header.  The phone then replies with OK, this time the To header
> includes a tag.
> 
> Is there any way to retrieve this response To header (including the tag
> field) from the dial plan?
> I have tried the PJSIP-HEADER read of the To header, but it seems to only
> have access to the initial To header.
> I even tried reading multiple layers of the To header, but it still
> didn't retrieve the newer dialog To headers.

The PJSIP_HEADER dialplan function currently only looks at the initial
message. It does not allow access to subsequent ones.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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

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[asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

[Dan Cropp]

I place a call into Asterisk (from SIP phone) and the To header does not have a 
tag.  Asterisk then sends it's Trying response, still no tag in the To header.  
The phone then replies with OK, this time the To header includes a tag.

Is there any way to retrieve this response To header (including the tag field) 
from the dial plan?
I have tried the PJSIP-HEADER read of the To header, but it seems to only have 
access to the initial To header.
I even tried reading multiple layers of the To header, but it still didn't 
retrieve the newer dialog To headers.

I am including the SIP messages reported by Asterisk for the call coming in...

*** Phone sends INVITE to Asterisk ***

INVITE sip:3...@xxx.xxx.xxx.xxx SIP/2.0^M
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5063;branch=z9hG4bK-18e552c3^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ^M
Call-ID: 3162d378-ea2b2...@yyy.yyy.yyy.yyy^M
CSeq: 102 INVITE^M
Max-Forwards: 70^M
Authorization: Digest 
username="1004",realm="asterisk",nonce="1485271992/b1bebde5cb4a763ed85b1d8e52c8e30d",uri="sip:3...@xxx.xxx.xxx.xxx",algorithm=MD5,response="8dd827e9910c2446fb0b8497f5944b81",opaque="66e52\
68a2111e777",qop=auth,nc=0001,cnonce="9dda9e0d"^M
Contact: "1004" ^M
Expires: 240^M
User-Agent: Cisco/SPA504G-7.4.8a^M
Content-Length: 401^M
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE^M
Supported: replaces^M
Content-Type: application/sdp^M
^M
v=0^M
o=- 32730859 32730859 IN IP4 yyy.yyy.yyy.yyy^M
s=-^M
c=IN IP4 yyy.yyy.yyy.yyy^M
t=0 0^M
m=audio 16436 RTP/AVP 0 2 8 9 18 96 97 98 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:2 G726-32/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:9 G722/8000^M
a=rtpmap:18 G729a/8000^M
a=rtpmap:96 G726-40/8000^M
a=rtpmap:97 G726-24/8000^M
a=rtpmap:98 G726-16/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
a=ptime:30^M
a=sendrecv^M

*** reply from Asterisk to phone ***

SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5063;received=yyy.yyy.yyy.yyy;branch=z9hG4bK-18e552c3^M
Call-ID: 3162d378-ea2b2...@yyy.yyy.yyy.yyy^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 14.2.1^M
Content-Length:  0^M
^M


**
Asterisk receives this packet in response to the Trying.
Is it possible to retrieve this To header via the dial plan?  Specifically, I 
need the tag portion of the From
**

SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 
yyy.yyy.yyy.yyy:5063;received=yyy.yyy.yyy.yyy;branch=z9hG4bK-18e552c3^M
Call-ID: 3162d378-ea2b2...@yyy.yyy.yyy.yyy^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ;tag=96156bd7-9e8e-4077-b6e4-f3eb12e39069^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 14.2.1^M
Contact: ^M
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, REGISTER, MESSAGE, REFER^M
Supported: 100rel, timer, replaces, norefersub^M
Content-Type: application/sdp^M
Content-Length:   179^M
^M
v=0^M
o=- 32730859 32730861 IN IP4 xxx.xxx.xxx.xxx^M
s=Asterisk^M
c=IN IP4 xxx.xxx.xxx.xxx^M
t=0 0^M
m=audio 19384 RTP/AVP 0^M
a=rtpmap:0 PCMU/8000^M
a=ptime:20^M
a=maxptime:150^M
a=sendrecv^M


ACK sip:xxx.xxx.xxx.xxx:5060 SIP/2.0^M
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5063;branch=z9hG4bK-c38362b^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ;tag=96156bd7-9e8e-4077-b6e4-f3eb12e39069^M
Call-ID: 3162d378-ea2b2...@yyy.yyy.yyy.yyy^M
CSeq: 102 ACK^M
Max-Forwards: 70^M
Authorization: Digest 
username="1004",realm="asterisk",nonce="1485271992/b1bebde5cb4a763ed85b1d8e52c8e30d",uri="sip:3...@xxx.xxx.xxx.xxx",algorithm=MD5,response="8dd827e9910c2446fb0b8497f5944b81",opaque="66e52\
68a2111e777",qop=auth,nc=0001,cnonce="9dda9e0d"^M
Contact: "1004" ^M
User-Agent: Cisco/SPA504G-7.4.8a^M
Content-Length: 0^M
^M

SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 
192.168.35.91:5063;received=192.168.35.91;branch=z9hG4bK-18e552c3^M
Call-ID: 3162d378-ea2b2452@192.168.35.91^M
From: "1004" ;tag=79e7940882a792ao2^M
To: ;tag=96156bd7-9e8e-4077-b6e4-f3eb12e39069^M
CSeq: 102 INVITE^M
Server: Asterisk PBX 14.2.1^M
Contact: ^M
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, REGISTER, MESSAGE, REFER^M
Supported: 100rel, timer, replaces, norefersub^M
Content-Type: application/sdp^M
Content-Length:   179^M
^M
v=0^M
o=- 32730859 32730861 IN IP4 192.168.33.30^M
s=Asterisk^M
c=IN IP4 192.168.33.30^M
t=0 0^M
m=audio 19384 RTP/AVP 0^M
a=rtpmap:0 PCMU/8000^M
a=ptime:20^M
a=maxptime:150^M
a=sendrecv^M
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Re: [asterisk-users] Asterisk - Vtiger integration

[Victor Villarreal]

Hi Alejandro,

The documentation about your question is here:
https://wiki.vtiger.com/vtiger6/index.php/PBX_Manager

After a few seconds of read, I think that VTigerAsteriskConnector can run
on a separate server than Asterisk PBX.

VTigerAsteriskConnector connects to Asterisk via Asterisk Manager Interface
(AMI), so you need to edit your /etc/asterisk/manager_custom.conf (because
you use Elastix distro) and create a user for the VTigerConnector. Then go
to CRM Settings -- > Integration --> PBXManager and complete all the info.

Note that seems that VTigerCoonector needs Java 1.7 onwards.

Please, follows the steps on the links. Cheers.

2017-01-13 16:04 GMT-03:00 Alejandro Cabrera Obed :

> Dear, I have Asterisk 1.8 (installed with Elastix 2.4) and I want to
> integrate a Vtiger 6.5 server.
>
> In my PBX I have Asterisk 1.8, Java 1.4 and I have not Java Jetty.
>
> What are the requirements in the Asterisk server in order to install the
> VtigerAsteriskConnector package and then integrate the services.
>
> Thanks a lot.
>
>
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> org/
>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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[asterisk-users] Asterisk - Vtiger integration

[Alejandro Cabrera Obed]

Dear, I have Asterisk 1.8 (installed with Elastix 2.4) and I want to
integrate a Vtiger 6.5 server.

In my PBX I have Asterisk 1.8, Java 1.4 and I have not Java Jetty.

What are the requirements in the Asterisk server in order to install the
VtigerAsteriskConnector package and then integrate the services.

Thanks a lot.
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Re: [asterisk-users] Asterisk hep.conf

[Olivier]

Hello,

Reading this thread, may I ask if you could get this to work ?

Regards

2016-06-29 6:29 GMT+02:00 Annus Fictus :

> hello,
>
> I'm trying to use Asterisk 13.9.1 with Homer SIP Capture Server.
>
> My hep.conf Asterisk configuration is:
>
> [general]
> enabled = yes
> capture_address=107.170.151.154:9060
> ;capture_password = foo
> capture_id = 2464
>
> SIP Signaling work correctly but no RTCP STATS arrive to Homer Server. On
> the Asterisk Console, many messages like this:
>
> NOTICE[3739] res_hep.c: Unable to send packet: Address Family mismatch
> between source/destination
>
> Any hint?
>
> Regards
>
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Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]

[Tzafrir Cohen]

On Mon, Dec 19, 2016 at 05:10:42PM +0100, Olivier wrote:

> Thanks for the tip:
> changing to permissive mode made it !
> 
> Using methods suggested in [1], do you think its possible and worth the
> effort to configure SELinux to work with Asterisk/Systemd in Enforcing mode
> ?
> 
> [1] https://wiki.centos.org/HowTos/SELinux

I think it should be possible. IIRC I once gave it a shot and was mildly
successful, but eventually gave up due to issues related to interaction
with Apache. If you do run into a problem, I wonder what it is.

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Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd

[Tzafrir Cohen]

On Mon, Dec 19, 2016 at 03:54:47PM +0100, Olivier wrote:
> Hello,
> 
> For a new project, I'm adapting existing installation script to CentOS 7.
> I must admit I don't understand how to adapt things to systemd.
> 
> Here are my questions:
> 
> 1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib.
> Do you think such directory and matching Makefile target could be useful ?
> 
> 2. Should /run/asterisk directory creation be left to systemd or done by
> installation script before running "systemctl start asterisk"  ?
> 
> 3. I edited the following /etc/systemd/system:asterisk.service file:
> [Unit]
> Description=Asterisk PBX and telephony daemon.
> After=network.target
> 
> [Service]
> Type=forking
> PIDFile=/var/run/asterisk/asterisk.pid

Remove those two (or get latest version with sd_notify support, make
sure it works, and use 'Type=notify')

> Environment=HOME=/var/lib/asterisk
> WorkingDirectory=/var/lib/asterisk
> ExecStart=/usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C

Drop -F as well

> /etc/asterisk/asterisk.conf
> #ExecStart=/usr/sbin/asterisk -vvvgF -C /etc/asterisk/asterisk.conf
> ExecStop=/usr/sbin/asterisk -rx 'core stop now'

I'm trying to think if this is needed. Anything wrong with just letting
systemd kill asterisk and all of its child precesses?

> ExecReload=/usr/sbin/asterisk -rx 'core reload'

Also, IIRC:

User=asterisk

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Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]

[Jean Aunis]

Le 19/12/2016 à 17:10, Olivier a écrit :



2016-12-19 16:11 GMT+01:00 Jean Aunis >:


Le 19/12/2016 à 15:54, Olivier a écrit :



Running systemctl start asterisk fails with :
Dec 19 15:43:08 foobar systemd: PID file
/var/run/asterisk/asterisk.pid not readable (yet?) after start.
Dec 19 15:43:09 foobar systemd: asterisk.service: main process
exited, code=exited, status=1/FAILURE
Dec 19 15:43:09 foobar asterisk: Unable to connect to remote
asterisk (does /var/run/asterisk/asterisk.ctl exist?)
Dec 19 15:43:09 foobar systemd: asterisk.service: control process
exited, code=exited status=1
Dec 19 15:43:09 foobar systemd: Unit asterisk.service entered
failed state.
Dec 19 15:43:09 foobar systemd: asterisk.service failed.


But /usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C
/etc/asterisk/asterisk.conf succeeds:
# rasterisk
Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
...
=
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 13.13.1 currently running on ...

Any hint or help on how to debug this ?
(I tried with and without any /run/asterisk directory owned by
asterisk.asterisk)


Best regards




Hello,

Make sure that selinux is disabled, or in "permissive" mode.
Otherwise it will prevent asterisk from starting.


Thanks for the tip:
changing to permissive mode made it !

Using methods suggested in [1], do you think its possible and worth 
the effort to configure SELinux to work with Asterisk/Systemd in 
Enforcing mode ?

A quick look in various tuto all disable SELinux.



[1] https://wiki.centos.org/HowTos/SELinux



I never spent time to figure out how selinux should be configured for 
Asterisk, but it is certainly possible to do something clean about that. 
I noticed that, when I install Asterisk with a custom-made RPM package, 
SELinux will stop blocking it. I guess RPM has some magic embedded into 
it to configure SELinux with the proper rules.


Still, is it worth the effort ? Probably not if you consider Asterisk 
alone : as it is running with the unprivileged user asterisk, the 
standard Linux permissions will protect your system if Asterisk is attacked.
But considering your system as a whole, disabling selinux may not be a 
good idea : other processes may required to be secured with the selinux 
stuff.


I'm not an IT security expert, so please consider what I wrote above 
with caution.
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Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]

[Olivier]

2016-12-19 16:11 GMT+01:00 Jean Aunis :

> Le 19/12/2016 à 15:54, Olivier a écrit :
>
> Hello,
>
> For a new project, I'm adapting existing installation script to CentOS 7.
> I must admit I don't understand how to adapt things to systemd.
>
> Here are my questions:
>
> 1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib.
> Do you think such directory and matching Makefile target could be useful ?
>
> 2. Should /run/asterisk directory creation be left to systemd or done by
> installation script before running "systemctl start asterisk"  ?
>
> 3. I edited the following /etc/systemd/system:asterisk.service file:
> [Unit]
> Description=Asterisk PBX and telephony daemon.
> After=network.target
>
> [Service]
> Type=forking
> PIDFile=/var/run/asterisk/asterisk.pid
> Environment=HOME=/var/lib/asterisk
> WorkingDirectory=/var/lib/asterisk
> ExecStart=/usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C
> /etc/asterisk/asterisk.conf
> #ExecStart=/usr/sbin/asterisk -vvvgF -C /etc/asterisk/asterisk.conf
> ExecStop=/usr/sbin/asterisk -rx 'core stop now'
> ExecReload=/usr/sbin/asterisk -rx 'core reload'
>
>
> [Install]
> WantedBy=multi-user.target
>
> Running systemctl start asterisk fails with :
> Dec 19 15:43:08 foobar systemd: PID file /var/run/asterisk/asterisk.pid
> not readable (yet?) after start.
> Dec 19 15:43:09 foobar systemd: asterisk.service: main process exited,
> code=exited, status=1/FAILURE
> Dec 19 15:43:09 foobar asterisk: Unable to connect to remote asterisk
> (does /var/run/asterisk/asterisk.ctl exist?)
> Dec 19 15:43:09 foobar systemd: asterisk.service: control process exited,
> code=exited status=1
> Dec 19 15:43:09 foobar systemd: Unit asterisk.service entered failed state.
> Dec 19 15:43:09 foobar systemd: asterisk.service failed.
>
>
> But /usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C
> /etc/asterisk/asterisk.conf succeeds:
> # rasterisk
> Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
> ...
> =
> Running as user 'asterisk'
> Running under group 'asterisk'
> Connected to Asterisk 13.13.1 currently running on ...
>
> Any hint or help on how to debug this ?
> (I tried with and without any /run/asterisk directory owned by
> asterisk.asterisk)
>
>
> Best regards
>
>
>
> Hello,
>
> Make sure that selinux is disabled, or in "permissive" mode. Otherwise it
> will prevent asterisk from starting.
>

Thanks for the tip:
changing to permissive mode made it !

Using methods suggested in [1], do you think its possible and worth the
effort to configure SELinux to work with Asterisk/Systemd in Enforcing mode
?
A quick look in various tuto all disable SELinux.



[1] https://wiki.centos.org/HowTos/SELinux



> Best regards
>
> Jean Aunis
>
> --
> _
> -- 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
>
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Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd

[Jean Aunis]

Le 19/12/2016 à 15:54, Olivier a écrit :

Hello,

For a new project, I'm adapting existing installation script to CentOS 7.
I must admit I don't understand how to adapt things to systemd.

Here are my questions:

1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib.
Do you think such directory and matching Makefile target could be useful ?

2. Should /run/asterisk directory creation be left to systemd or done 
by installation script before running "systemctl start asterisk"  ?


3. I edited the following /etc/systemd/system:asterisk.service file:
[Unit]
Description=Asterisk PBX and telephony daemon.
After=network.target

[Service]
Type=forking
PIDFile=/var/run/asterisk/asterisk.pid
Environment=HOME=/var/lib/asterisk
WorkingDirectory=/var/lib/asterisk
ExecStart=/usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C 
/etc/asterisk/asterisk.conf

#ExecStart=/usr/sbin/asterisk -vvvgF -C /etc/asterisk/asterisk.conf
ExecStop=/usr/sbin/asterisk -rx 'core stop now'
ExecReload=/usr/sbin/asterisk -rx 'core reload'


[Install]
WantedBy=multi-user.target

Running systemctl start asterisk fails with :
Dec 19 15:43:08 foobar systemd: PID file 
/var/run/asterisk/asterisk.pid not readable (yet?) after start.
Dec 19 15:43:09 foobar systemd: asterisk.service: main process exited, 
code=exited, status=1/FAILURE
Dec 19 15:43:09 foobar asterisk: Unable to connect to remote asterisk 
(does /var/run/asterisk/asterisk.ctl exist?)
Dec 19 15:43:09 foobar systemd: asterisk.service: control process 
exited, code=exited status=1
Dec 19 15:43:09 foobar systemd: Unit asterisk.service entered failed 
state.

Dec 19 15:43:09 foobar systemd: asterisk.service failed.


But /usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C 
/etc/asterisk/asterisk.conf succeeds:

# rasterisk
Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
...
=
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 13.13.1 currently running on ...

Any hint or help on how to debug this ?
(I tried with and without any /run/asterisk directory owned by 
asterisk.asterisk)



Best regards




Hello,

Make sure that selinux is disabled, or in "permissive" mode. Otherwise 
it will prevent asterisk from starting.


Best regards

Jean Aunis

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[asterisk-users] Asterisk installation script on CentOS7 with systemd

[Olivier]

Hello,

For a new project, I'm adapting existing installation script to CentOS 7.
I must admit I don't understand how to adapt things to systemd.

Here are my questions:

1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib.
Do you think such directory and matching Makefile target could be useful ?

2. Should /run/asterisk directory creation be left to systemd or done by
installation script before running "systemctl start asterisk"  ?

3. I edited the following /etc/systemd/system:asterisk.service file:
[Unit]
Description=Asterisk PBX and telephony daemon.
After=network.target

[Service]
Type=forking
PIDFile=/var/run/asterisk/asterisk.pid
Environment=HOME=/var/lib/asterisk
WorkingDirectory=/var/lib/asterisk
ExecStart=/usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C
/etc/asterisk/asterisk.conf
#ExecStart=/usr/sbin/asterisk -vvvgF -C /etc/asterisk/asterisk.conf
ExecStop=/usr/sbin/asterisk -rx 'core stop now'
ExecReload=/usr/sbin/asterisk -rx 'core reload'


[Install]
WantedBy=multi-user.target

Running systemctl start asterisk fails with :
Dec 19 15:43:08 foobar systemd: PID file /var/run/asterisk/asterisk.pid not
readable (yet?) after start.
Dec 19 15:43:09 foobar systemd: asterisk.service: main process exited,
code=exited, status=1/FAILURE
Dec 19 15:43:09 foobar asterisk: Unable to connect to remote asterisk (does
/var/run/asterisk/asterisk.ctl exist?)
Dec 19 15:43:09 foobar systemd: asterisk.service: control process exited,
code=exited status=1
Dec 19 15:43:09 foobar systemd: Unit asterisk.service entered failed state.
Dec 19 15:43:09 foobar systemd: asterisk.service failed.


But /usr/sbin/asterisk -vvvgF -U asterisk -G asterisk -C
/etc/asterisk/asterisk.conf succeeds:
# rasterisk
Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
...
=
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 13.13.1 currently running on ...

Any hint or help on how to debug this ?
(I tried with and without any /run/asterisk directory owned by
asterisk.asterisk)


Best regards
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Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Yves]

ok,

thank you... then I´ll take it as it is

cheers,
yves

Am 18.12.2016 um 13:15 schrieb Larry Moore:

Hi,

I haven't found anything definitive however I expect the TSI that is 
sent during initial fax call establishment is stored by the receiving 
terminal, see pages 28 & 29 of the English version of the document at 
https://www.itu.int/rec/T-REC-T.30-200509-I/en , I expect the header, 
which will include the TSI, is all part of the image (Tagline in 
HylaFAX) and not stored separately on the receiving terminal.


Cheers,

Larry.

On 18/12/2016 6:20 PM, Yves wrote:

Hi,

thanks for your answer. Unfortunately this is, what I already know. I 
was wondering, why it is possible to set ID and Header for an 
outgoing fax (which will then in turn
be inserted via asterisk on top of the transferred "image") , while 
it seems to not be possible to get the Header from a received fax 
(only the id), although it is present in the faxdocument.
The ID is also present in the faxdocument and there does a 
Faxopt(remotestationid) exist... so I thought, this info must be 
transferred not only binary within the "image", but
also within the "meta-data" / protocol-data of the fax (within the 
TSI) otherwise asterisk must do some kind of ocr to get the ID, 
what it definitely does not...


btw... when using sendfax, asterisk inserts the date, the id, the 
header and the pagenum on top of each faxpage... someone knows how to 
modify some settings like font, position, and so on?


thanks,
yves


Am 18.12.2016 um 00:02 schrieb Larry Moore:
The list of options available are listed here 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT


It doesn't appear that a received header is available unless it is 
written into the 'headerinfo' variable after it is received, I 
haven't checked for this.


From my days working with fax machines, the header could be inserted 
in the line the TSI is on or in the image being transmitted, if you 
receive a fax that has been sent to you with the latter set, then 
the 'headerinfo' will not be of any use. Perhaps someone with more 
knowledge may be able to explain this better.


A quick Google search for 'fax header outside of tsi' will provide a 
list of manuals, here's one - 
http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 



Expand the line for


To specify the position of Header Position printed on a sent
fax ([Header Position])

Larry.

On 18/12/2016 4:30 AM, Yves wrote:

Hi,

I am using asterisk 11.8 in combination with spandsp to send and 
receive T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and 
the headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo 
besides the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves
















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Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Larry Moore]

Hi,

I haven't found anything definitive however I expect the TSI that is 
sent during initial fax call establishment is stored by the receiving 
terminal, see pages 28 & 29 of the English version of the document at 
https://www.itu.int/rec/T-REC-T.30-200509-I/en , I expect the header, 
which will include the TSI, is all part of the image (Tagline in 
HylaFAX) and not stored separately on the receiving terminal.


Cheers,

Larry.

On 18/12/2016 6:20 PM, Yves wrote:

Hi,

thanks for your answer. Unfortunately this is, what I already know. I 
was wondering, why it is possible to set ID and Header for an outgoing 
fax (which will then in turn
be inserted via asterisk on top of the transferred "image") , while it 
seems to not be possible to get the Header from a received fax (only 
the id), although it is present in the faxdocument.
The ID is also present in the faxdocument and there does a 
Faxopt(remotestationid) exist... so I thought, this info must be 
transferred not only binary within the "image", but
also within the "meta-data" / protocol-data of the fax (within the 
TSI) otherwise asterisk must do some kind of ocr to get the ID, 
what it definitely does not...


btw... when using sendfax, asterisk inserts the date, the id, the 
header and the pagenum on top of each faxpage... someone knows how to 
modify some settings like font, position, and so on?


thanks,
yves


Am 18.12.2016 um 00:02 schrieb Larry Moore:
The list of options available are listed here 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT


It doesn't appear that a received header is available unless it is 
written into the 'headerinfo' variable after it is received, I 
haven't checked for this.


From my days working with fax machines, the header could be inserted 
in the line the TSI is on or in the image being transmitted, if you 
receive a fax that has been sent to you with the latter set, then the 
'headerinfo' will not be of any use. Perhaps someone with more 
knowledge may be able to explain this better.


A quick Google search for 'fax header outside of tsi' will provide a 
list of manuals, here's one - 
http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 



Expand the line for


To specify the position of Header Position printed on a sent
fax ([Header Position])

Larry.

On 18/12/2016 4:30 AM, Yves wrote:

Hi,

I am using asterisk 11.8 in combination with spandsp to send and 
receive T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and 
the headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo 
besides the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves












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Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Yves]

Hi,

thanks for your answer. Unfortunately this is, what I already know. I 
was wondering, why it is possible to set ID and Header for an outgoing 
fax (which will then in turn
be inserted via asterisk on top of the transferred "image") , while it 
seems to not be possible to get the Header from a received fax (only the 
id), although it is present in the faxdocument.
The ID is also present in the faxdocument and there does a 
Faxopt(remotestationid) exist... so I thought, this info must be 
transferred not only binary within the "image", but
also within the "meta-data" / protocol-data of the fax (within the 
TSI) otherwise asterisk must do some kind of ocr to get the ID, what 
it definitely does not...


btw... when using sendfax, asterisk inserts the date, the id, the header 
and the pagenum on top of each faxpage... someone knows how to modify 
some settings like font, position, and so on?


thanks,
yves


Am 18.12.2016 um 00:02 schrieb Larry Moore:
The list of options available are listed here 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT


It doesn't appear that a received header is available unless it is 
written into the 'headerinfo' variable after it is received, I haven't 
checked for this.


From my days working with fax machines, the header could be inserted 
in the line the TSI is on or in the image being transmitted, if you 
receive a fax that has been sent to you with the latter set, then the 
'headerinfo' will not be of any use. Perhaps someone with more 
knowledge may be able to explain this better.


A quick Google search for 'fax header outside of tsi' will provide a 
list of manuals, here's one - 
http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 



Expand the line for


To specify the position of Header Position printed on a sent
fax ([Header Position])

Larry.

On 18/12/2016 4:30 AM, Yves wrote:

Hi,

I am using asterisk 11.8 in combination with spandsp to send and 
receive T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and 
the headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo besides 
the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves








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Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Larry Moore]

The list of options available are listed here 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT


It doesn't appear that a received header is available unless it is 
written into the 'headerinfo' variable after it is received, I haven't 
checked for this.


From my days working with fax machines, the header could be inserted in 
the line the TSI is on or in the image being transmitted, if you receive 
a fax that has been sent to you with the latter set, then the 
'headerinfo' will not be of any use. Perhaps someone with more knowledge 
may be able to explain this better.


A quick Google search for 'fax header outside of tsi' will provide a 
list of manuals, here's one - 
http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 



Expand the line for


   To specify the position of Header Position printed on a sent fax
   ([Header Position])

Larry.

On 18/12/2016 4:30 AM, Yves wrote:

Hi,

I am using asterisk 11.8 in combination with spandsp to send and 
receive T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and the 
headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo besides 
the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves




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[asterisk-users] Asterisk Fax Receive - how to get the remoteheader?

[Yves]

Hi,

I am using asterisk 11.8 in combination with spandsp to send and receive 
T38 Faxes. All works fine, but I do not know

how to get the remoteheader from the fax I receive.

When I send a fax, there are Faxopts to set the localstationid and the 
headerinfo, but for receiving, there seems to only exist

the Faxopts remotestationid

but for sure on any fax I receive there is a remoteheaderinfo besides 
the remotestationid... it is on the tiff-file, but I need this

info in a channel-variable...

Does anybody know how to get the remoteheaderinfo for a received fax?

thanks

yves


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[asterisk-users] Asterisk 11 with Dovecot IMAP Email

[Tammy Firefly]

Hi all,
Has anyone gotten dovecot 2.2 working with the asterisk 11 imap
voicemail system?  I can login via thunderbird or telnet to the imap
server just fine.

The below is logged:

[Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:3176 mm_log: IMAP Error:
Can't open mailbox
{localhost:993/imap/authuser=asterisk/tls,novalidate-cert/user=test}INBOX:
invalid remote specification
[Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:2906 init_mailstream:
Can't connect to imap server
{localhost:993/imap/authuser=asterisk/tls,novalidate-cert/user=test}INBOX
[Dec 16 13:35:47] ERROR[17285]: app_voicemail.c:2424 __messagecount:
Houston we have a problem - IMAP mailstream is NULL

As of now its not even sending a imap request to dovecot.

I am using a master user which I can successfully login as.  Ive tried
specifying individual usernames/passwords in the voicemail.conf file.
This doesnt work either.

Thanks
--Tammy



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[asterisk-users] Asterisk versions supporting Path header?

[Daniel Pocock]

The bug tracker includes several issues relating to Path (RFC 3327)
support.  It is not clear which version actually included the patch and
which versions are working.

Could anybody update these issues in Jira with a brief comment about the
supported versions?


https://issues.asterisk.org/jira/browse/ASTERISK-16884
   original patch against chan_sip / Asterisk 1.8
   Status is "Fixed", but not version is recorded,
   which version was this merged in?

https://issues.asterisk.org/jira/browse/ASTERISK-21084
   chan_pjsip Path support
   Satus is Fixed for v12.1.0
   - is that only for chan_pjsip, or is Path also
 supported in chan_sip in any versions up to 12.1.0?

https://issues.asterisk.org/jira/browse/ASTERISK-25666
   Path header ignored (looks like a regression?)
   reported for 13.6.0 - which is the last version where it did work?


https://jira.digium.com/browse/SWP-2484
   "add Path header support to chan_sip"
   Internal Jira link - does this issue contain any further
   details about the versions supported?

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Re: [asterisk-users] Asterisk dahai install centos 7

[Alex Villacís Lasso]

El 10/12/16 a las 10:15, christopher kamutumwa escribió:
Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and 
latest version but i still receive the same error


[root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make
make -C linux all
make[1]: Entering directory 
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux'

make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory 
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
make[2]: Leaving directory 
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
You do not appear to have the sources for the 3.10.0-327.el7.x86_64 
kernel installed.

make[1]: *** [modules] Error 1


When compiling kernel modules (such as DAHDI) using the system kernel 
and kernel-devel, they must be compiled under a specific kernel version. 
If no kernel version is specified, it defaults to the currently running 
kernel. In your case, the currently running kernel is 
3.10.0-327.el7.x86_64 .


For the selected kernel version, the kernel development files (supplied 
by kernel-devel) must exist under the exact same version. The installed 
kernel-devel version is 3.10.0-327.36.3.el7.x86_64 . Note carefully - 
this is NOT the same kernel version as the kernel selected for compiling -


3.10.0-327.36.3.el7.x86_64 (supplied by kernel-devel)
3.10.0-327.el7.x86_64 (selected by default)

Notice the extra ".36.3" between the two versions.

To solve this, you should do EXACTLY ONE of the following two options:
1) install the exact kernel-devel version for your currently-running 
kernel. In this case, kernel-devel-3.10.0-327.el7.x86_64 . Then run make 
as before.
2) install kernel-3.10.0-327.36.3.el7.x86_64 if not already installed, 
then specify this version as the kernel to compile DAHDI against:


make KVERS="3.10.0-327.36.3.el7.x86_64"
make install KVERS="3.10.0-327.36.3.el7.x86_64"


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Re: [asterisk-users] Asterisk dahai install centos 7

[Tony Mountifield]

In article ,
christopher kamutumwa  wrote:
> 
> Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and
> latest version but i still receive the same error
> 
> [root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make
> make -C linux all
> make[1]: Entering directory `/usr/src/dahdi-linux-
> complete-2.11.1+2.11.1/linux'
> make -C drivers/dahdi/firmware firmware-loaders
> make[2]: Entering directory `/usr/src/dahdi-linux-
> complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
> make[2]: Leaving directory `/usr/src/dahdi-linux-
> complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
> You do not appear to have the sources for the 3.10.0-327.el7.x86_64 kernel
> installed.
> make[1]: *** [modules] Error 1

You need to make sure the kernel and kernel-devel packages are the same
version, and that you are running from the same kernel too (make sure
you rebooted after any update of the kernel).

The version of kernel-devel you have installed is: 3.10.0-327.36.3
The version of kernel-devel the makefile wants is: 3.10.0-327

The difference is significant, and suggests that you are actually
running an older kernel than the latest.

Cheers
Tony
-- 
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] Asterisk dahai install centos 7

[Markus]

There are inofficial RPMs for CentOS 7 available if you don't want to 
mess with compiling: https://www.tucny.com/telephony/asterisk-rpms



Am 10.12.2016 um 15:47 schrieb christopher kamutumwa:

Hello support

am trying to install dahdi on centos 7 and am doing the make ommand and
below is result any way out this You do not appear to have the sources
for the 3.10.0-327.el7.x86_64 kernel installed.

uname -a gives me below;

Linux localhost.localdomain 3.10.0-327.el7.x86_64 #1 SMP Thu Nov 19
22:10:57 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux



[root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make
make -C linux all
make[1]: Entering directory
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
make[2]: Leaving directory
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
You do not appear to have the sources for the 3.10.0-327.el7.x86_64
kernel installed.
make[1]: *** [modules] Error 1
make[1]: Leaving directory
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux'
make: *** [all] Error 2






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Re: [asterisk-users] Asterisk dahai install centos 7

[Greg Woods]

On Sat, Dec 10, 2016 at 8:15 AM, christopher kamutumwa <
chriskamutu...@gmail.com> wrote:

> Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and
> latest version but i still receive the same error
>

How about kernel-headers? That's a separate package on my Fedora system,
not sure about CentOS.

--Greg
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Re: [asterisk-users] Asterisk dahai install centos 7

[christopher kamutumwa]

Package kernel-devel-3.10.0-327.36.3.el7.x86_64 already installed and
latest version but i still receive the same error

[root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make
make -C linux all
make[1]: Entering directory `/usr/src/dahdi-linux-
complete-2.11.1+2.11.1/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory `/usr/src/dahdi-linux-
complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
make[2]: Leaving directory `/usr/src/dahdi-linux-
complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
You do not appear to have the sources for the 3.10.0-327.el7.x86_64 kernel
installed.
make[1]: *** [modules] Error 1


On Sat, Dec 10, 2016 at 4:56 PM, Greg Woods  wrote:

>
> On Sat, Dec 10, 2016 at 7:47 AM, christopher kamutumwa <
> chriskamutu...@gmail.com> wrote:
>
>> You do not appear to have the sources for the 3.10.0-327.el7.x86_64
>> kernel installed.
>
>
> You need to install the kernel-devel package to compile kernel modules.
>
> --Greg
>
>
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> 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
>
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Re: [asterisk-users] Asterisk dahai install centos 7

[Greg Woods]

On Sat, Dec 10, 2016 at 7:47 AM, christopher kamutumwa <
chriskamutu...@gmail.com> wrote:

> You do not appear to have the sources for the 3.10.0-327.el7.x86_64 kernel
> installed.


You need to install the kernel-devel package to compile kernel modules.

--Greg
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[asterisk-users] Asterisk dahai install centos 7

[christopher kamutumwa]

Hello support

am trying to install dahdi on centos 7 and am doing the make ommand and
below is result any way out this You do not appear to have the sources for
the 3.10.0-327.el7.x86_64 kernel installed.

uname -a gives me below;

Linux localhost.localdomain 3.10.0-327.el7.x86_64 #1 SMP Thu Nov 19
22:10:57 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux



[root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make
make -C linux all
make[1]: Entering directory
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
make[2]: Leaving directory
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware'
You do not appear to have the sources for the 3.10.0-327.el7.x86_64 kernel
installed.
make[1]: *** [modules] Error 1
make[1]: Leaving directory
`/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux'
make: *** [all] Error 2
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[asterisk-users] Asterisk 11.25.1, 13.13.1, 14.2.1, 11.6-cert16, and 13.8-cert4 Now Available (Security Release)

[Asterisk Development Team]

The Asterisk Development Team has announced security releases for Asterisk
11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available
security releases are released as versions 11.25.1, 13.13.1, 14.2.1,
11.6-cert16, and 13.8-cert4.

These releases are available for immediate download at:

http://downloads.asterisk.org/pub/telephony/asterisk/releases
http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/

The release of versions 13.13.1 and 14.2.1 resolve the following security
vulnerability:

* AST-2016-008: Crash on SDP offer or answer from endpoint using Opus

  If an SDP offer or answer is received with the Opus codec and with the
format
  parameters separated using a space the code responsible for parsing will
  recursively call itself until it crashes. This occurs as the code does not
  properly handle spaces separating the parameters.

  This does NOT require the endpoint to have Opus configured in Asterisk.
This
  also does not require the endpoint to be authenticated. If guest is
enabled
  for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
  processed and the crash occurs.

The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
resolve the following security vulnerability:

* AST-2016-009: Remote unauthenticated sessions in chan_sip

  The chan_sip channel driver has a liberal definition for whitespace when
  attempting to strip the content between a SIP header name and a colon
  character. Rather than following RFC 3261 and stripping only spaces and
  horizontal tabs, Asterisk treats any non-printable ASCII character as if
it
  were whitespace. This means that headers such as

 Contact\x01:

  will be seen as a valid Contact header.

  This mostly does not pose a problem until Asterisk is placed in tandem
with
  an authenticating SIP proxy. In such a case, a crafty combination of valid
  and invalid To headers can cause a proxy to allow an INVITE request into
  Asterisk without authentication since it believes the request is an
in-dialog
  request. However, because of the bug described above, the request will
look
  like an out-of-dialog request to Asterisk. Asterisk will then process the
  request as a new call. The result is that Asterisk can process calls from
  unvetted sources without any authentication.

  If you do not use a proxy for authentication, then this issue does not
affect
  you. If your proxy is dialog-aware (meaning that the proxy keeps track of
what
  dialogs are currently valid), then this issue does not affect you. If you
use
  chan_pjsip instead of chan_sip, then this issue does not affect you.

For a full list of changes in the current releases, please see the
ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/release
s/ChangeLog-11.25.1
http://downloads.asterisk.org/pub/telephony/asterisk/release
s/ChangeLog-13.13.1
http://downloads.asterisk.org/pub/telephony/asterisk/release
s/ChangeLog-14.2.1
http://downloads.asterisk.org/pub/telephony/certified-asteri
sk/releases/ChangeLog-certified-11.6-cert16
http://downloads.asterisk.org/pub/telephony/certified-asteri
sk/releases/ChangeLog-certified-13.8-cert4

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf

Thank you for your continued support of Asterisk!
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[asterisk-users] Asterisk 11.25.1, 13.13.1, 14.2.1, 11.6-cert16, and 13.8-cert4 Now Available (Security Release)

[Asterisk Development Team]

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Re: [asterisk-users] Asterisk Can't start with the default configs

[Tzafrir Cohen]

On Sun, Dec 04, 2016 at 08:00:45PM +0100, Mr Dini wrote:
> Hi,
> 
> I tried to run the make progdocs, but the first time, it said, I have no
> doxygen installed. So I compiled the latest release and reconfigure the
> asterisk. And after it, ut sucessfully started to build the docs. But it
> took a lot of time, So finally I aborted the process...
> 
> Is there a way to disable doc creating? The --disable-xmldoc is enough?

I must be missing something here. Isn't 'progdoc' an optional target?

The relevant XML documentation is doc/core-en_US.xml, which is generated
unconditionaly in the build and using awk, shell and make.

Another note: it is installed to the astdatadir. Which is indeed by
default var/lib/asterisk, but may have different values (build time or
set at run time in asterisk.conf).

-- 
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Re: [asterisk-users] Asterisk install challenge

[Tzafrir Cohen]

On Tue, Dec 06, 2016 at 06:35:07PM +0200, christopher kamutumwa wrote:
> am new to asterisk and am trying to "make all". or dahdi install but in
> only reach this stage below on centos 6.8 . Any idea how to resolve or
> bypass this
> 
> 
> configure: creating ./config.status
> ./configure: line 18858: cannot create temp file for here-document: No such
> file or directory
> configure: error: write failure creating ./config.status
> make: *** [all] Error 1


Not really sure. But maybe you're out of disk space? Can you create a
file?

  df -h .

Failing that: permission issues?

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[asterisk-users] Asterisk install challenge

[christopher kamutumwa]

am new to asterisk and am trying to "make all". or dahdi install but in
only reach this stage below on centos 6.8 . Any idea how to resolve or
bypass this


configure: creating ./config.status
./configure: line 18858: cannot create temp file for here-document: No such
file or directory
configure: error: write failure creating ./config.status
make: *** [all] Error 1


Thanks,

Chris
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Re: [asterisk-users] Asterisk Can't start with the default configs

[Mr Dini]

Okay, in this case will create a ticket tomorrow.

Thanks for Your help!
On Dec 4, 2016 10:25 PM, "Joshua Colp"  wrote:

> On Sun, Dec 4, 2016, at 05:13 PM, Mr Dini wrote:
> > No, the disable-xmldoc doesn't disable the whole doc creating procedure.
> >
> > Is there a way to disable it completely?
> >
> > Regarding the issue... Of course, I Can open a ticket, just I don't know
> > about what exactly. I want to compile it without doc generate to make the
> > asterisk module loads up fine.
>
> The configure option should disable the creation and requirement for the
> documentation. That's the problem.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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>
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> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Asterisk Can't start with the default configs

[Joshua Colp]

On Sun, Dec 4, 2016, at 05:13 PM, Mr Dini wrote:
> No, the disable-xmldoc doesn't disable the whole doc creating procedure.
> 
> Is there a way to disable it completely?
> 
> Regarding the issue... Of course, I Can open a ticket, just I don't know
> about what exactly. I want to compile it without doc generate to make the
> asterisk module loads up fine.

The configure option should disable the creation and requirement for the
documentation. That's the problem.

-- 
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk Can't start with the default configs

[Mr Dini]

No, the disable-xmldoc doesn't disable the whole doc creating procedure.

Is there a way to disable it completely?

Regarding the issue... Of course, I Can open a ticket, just I don't know
about what exactly. I want to compile it without doc generate to make the
asterisk module loads up fine.
On Dec 4, 2016 8:41 PM, "Joshua Colp"  wrote:

> On Sun, Dec 4, 2016, at 03:00 PM, Mr Dini wrote:
> > Hi,
> >
> > I tried to run the make progdocs, but the first time, it said, I have no
> > doxygen installed. So I compiled the latest release and reconfigure the
> > asterisk. And after it, ut sucessfully started to build the docs. But it
> > took a lot of time, So finally I aborted the process...
> >
> > Is there a way to disable doc creating? The --disable-xmldoc is enough?
>
> The --disable-xmldoc option should work. From my own experimentation it
> doesn't appear to be working as expected though. Can you file an
> issue[1] to look into it?
>
> [1] https://issues.asterisk.org/jira
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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>
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> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Asterisk Can't start with the default configs

[Joshua Colp]

On Sun, Dec 4, 2016, at 03:00 PM, Mr Dini wrote:
> Hi,
> 
> I tried to run the make progdocs, but the first time, it said, I have no
> doxygen installed. So I compiled the latest release and reconfigure the
> asterisk. And after it, ut sucessfully started to build the docs. But it
> took a lot of time, So finally I aborted the process...
> 
> Is there a way to disable doc creating? The --disable-xmldoc is enough?

The --disable-xmldoc option should work. From my own experimentation it
doesn't appear to be working as expected though. Can you file an
issue[1] to look into it?

[1] https://issues.asterisk.org/jira

-- 
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk Can't start with the default configs

[Mr Dini]

Hi,

I tried to run the make progdocs, but the first time, it said, I have no
doxygen installed. So I compiled the latest release and reconfigure the
asterisk. And after it, ut sucessfully started to build the docs. But it
took a lot of time, So finally I aborted the process...

Is there a way to disable doc creating? The --disable-xmldoc is enough?

Thanks!
On Dec 2, 2016 3:36 PM, "Joshua Colp"  wrote:

> On Fri, Dec 2, 2016, at 10:27 AM, Mr Dini wrote:
> > Hi,
> >
> > I compiled the asterisk 14.0.2 to my ARMv5 NAS, however I just have
> > enough
> > time to test it now.
> >
> > But with the default config (I only edited the http.conf), it won't
> > start,
> > but gives the following:
> >
> > Sorcery registered wizard 'bucket'
> > Sorcery registered wizard 'bucket_file'
> > Parsing /ffp/etc/asterisk/sorcery.conf
> > Parsing '/ffp/etc/asterisk/sorcery.conf': Found
> > Cannot update type 'bucket' in module 'core' because it has no existing
> > documentation!
> > Failed to register 'bucket' object type in Bucket sorcery
> > Failed: ast_bucket_init
> >
> > What is the problem?
>
> This means that Asterisk was built with documentation support but the
> documentation is not installed. Many things will fail to load as a
> result of this. It is commonly located in the
> /var/lib/asterisk/documentation directory as core-en_US.xml but in your
> embedded environment it may be different.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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>
> 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
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk Can't start with the default configs

[Mr Dini]

Okay, thanks for the info! Will recompile it with corrected doc support and
see what happens.
On Dec 2, 2016 3:36 PM, "Joshua Colp"  wrote:

> On Fri, Dec 2, 2016, at 10:27 AM, Mr Dini wrote:
> > Hi,
> >
> > I compiled the asterisk 14.0.2 to my ARMv5 NAS, however I just have
> > enough
> > time to test it now.
> >
> > But with the default config (I only edited the http.conf), it won't
> > start,
> > but gives the following:
> >
> > Sorcery registered wizard 'bucket'
> > Sorcery registered wizard 'bucket_file'
> > Parsing /ffp/etc/asterisk/sorcery.conf
> > Parsing '/ffp/etc/asterisk/sorcery.conf': Found
> > Cannot update type 'bucket' in module 'core' because it has no existing
> > documentation!
> > Failed to register 'bucket' object type in Bucket sorcery
> > Failed: ast_bucket_init
> >
> > What is the problem?
>
> This means that Asterisk was built with documentation support but the
> documentation is not installed. Many things will fail to load as a
> result of this. It is commonly located in the
> /var/lib/asterisk/documentation directory as core-en_US.xml but in your
> embedded environment it may be different.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
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> 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
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Re: [asterisk-users] Asterisk Can't start with the default configs

[Joshua Colp]

On Fri, Dec 2, 2016, at 10:27 AM, Mr Dini wrote:
> Hi,
> 
> I compiled the asterisk 14.0.2 to my ARMv5 NAS, however I just have
> enough
> time to test it now.
> 
> But with the default config (I only edited the http.conf), it won't
> start,
> but gives the following:
> 
> Sorcery registered wizard 'bucket'
> Sorcery registered wizard 'bucket_file'
> Parsing /ffp/etc/asterisk/sorcery.conf
> Parsing '/ffp/etc/asterisk/sorcery.conf': Found
> Cannot update type 'bucket' in module 'core' because it has no existing
> documentation!
> Failed to register 'bucket' object type in Bucket sorcery
> Failed: ast_bucket_init
> 
> What is the problem?

This means that Asterisk was built with documentation support but the
documentation is not installed. Many things will fail to load as a
result of this. It is commonly located in the
/var/lib/asterisk/documentation directory as core-en_US.xml but in your
embedded environment it may be different.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk Can't start with the default configs

[Mr Dini]

Hi,

I compiled the asterisk 14.0.2 to my ARMv5 NAS, however I just have enough
time to test it now.

But with the default config (I only edited the http.conf), it won't start,
but gives the following:

Sorcery registered wizard 'bucket'
Sorcery registered wizard 'bucket_file'
Parsing /ffp/etc/asterisk/sorcery.conf
Parsing '/ffp/etc/asterisk/sorcery.conf': Found
Cannot update type 'bucket' in module 'core' because it has no existing
documentation!
Failed to register 'bucket' object type in Bucket sorcery
Failed: ast_bucket_init

What is the problem?

Many thanks!

Ps: if any addition information is needed, certainly I will attach it, just
please tell me, what is needed.
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[asterisk-users] Asterisk 14 web broadcast

[Matt Fredrickson]

Hey All,

Slight interlude from your regularly scheduled programming.

For any interested, I will be giving a web broadcast today about
Asterisk 14 and what's new with Asterisk since the 13 release.  For
those of you that aren't aware, I'm responsible for day to day
management of the Asterisk project now that Matt Jordan has been moved
into the CTO role at Digium.

You can get info about it at:

http://bit.ly/2gDkFrh

It will be live today at 8AM, 2PM, and 9PM CDT.

Hope to see many of you there!

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] Asterisk compatibility with SMS services

[A J Stiles]

On Wednesday 30 Nov 2016, Emiliano Vazquez wrote:
> i'm using gammu[1] with a 3g dongle and my own chip with my preffer
> provider. It can send over 700 every our and receive to. I don't know if
> you need asterisk and sms in the same way but with this tool you can make
> everything. It has python tools to.
> [1] https://wammu.eu/gammu/

I've used that, with an old pre-smartphone mobile as a GSM modem.  It has the 
usual GUI frontend and scriptable backend.  But it is separate from Asterisk.


I have also used the OpenVox G400P/E in the past, which integrates beautifully 
with Asterisk  (incoming text messages, successfully sent outgoing messages 
and failed sent outgoing messages trigger extensions in the dialplan);  but 
this card has been "no longer recommended for new designs" for some time now  
(chan_extra won't even build against recent kernel or Asterisk versions).


Be aware that if you send too many text messages in too short a timespan, your 
telco might deem that to be in excess of their "fair use policy" -- although 
the exact dividing line between fair and unfair use seems to be a jealously-
guarded secret.  The only way to find out for sure is to cross it .


-- 
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] Asterisk compatibility with SMS services

[Emiliano Vazquez]

i'm using gammu[1] with a 3g dongle and my own chip with my preffer
provider. It can send over 700 every our and receive to. I don't know if
you need asterisk and sms in the same way but with this tool you can make
everything. It has python tools to.


Best regards.

Emiliano.

[1] https://wammu.eu/gammu/

On Tue, Nov 29, 2016 at 3:07 PM, Brandon B.  wrote:

>
> Can anyone comment on using SMS in conjunction with VoIP service using one
> of these three VoIP providers: voip.ms, vitelity.com, flowroute.com? Are
> some SMS services more compatible with Asterisk (i.e. SMS over SIP works
> perfectly or not)? Is it best to use a different data channel for SMS
> messages (i.e. SMS via HTTP, SMS via XMPP) instead of Asterisk's built in SMS
> application MessageSend
> ? In order to develop
> a web application for sending and receives SMS messages for business users,
> are there any pitfalls in using Asterisk to handle the message exchanges?
>
>
> On 2016-11-29 09:01 AM, Sebastian Nielsen wrote:
>
> Im using SMS successfully over VoIP. No problems at all. You however need
> to use a good codec.
>
>
>
> However, I don’t use the MessageSend application, instead I use the raw
> SMS() application.
>
> This works by the SMS centre calling my fixed landline from a specific
> number, I detect the callerid, initiate a SMS reception and then the SMS is
> in the spool files.
>
> If I want to send a outgoing SMS, I push a SMS file in the spool folder,
> then initate a call to the SMS centre.
>
>
> That's pretty cool, thank you for the details. You are using the builtin
> SMS application that exchanges SMS data over SIP / PSTN connections. I
> don't believe I can get service like that in Canada. Does anyone use the
> SMS applications to send and receive SMS messages in North America? Who
> provides that service?
>
>
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> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>



-- 
Emiliano Vazquez | PcCentro S.R.L.
White 1611 | C.P. C1407IJG | C.A.B.A.
Office: +54 (11) 4635-7764
Celular: 15.6253.7165
Mail: emilianovazq...@gmail.com 
Web: http://www.pccentro.com.ar
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Re: [asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

[Michele Pinassi]

Yes, it works !

Thanks :-)

Michele

On 30/11/2016 10:19, Jonathan H wrote:
> I think it might be related to this?
> https://issues.asterisk.org/jira/browse/ASTERISK-26391
>
> I think I remember having to edit logger.conf - this is what mine
> looks like now:
> console => notice,warning,error
> messages => notice,warning,error
>
> Try that, restart asterisk and see if it works :)

-- 
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di 
Ateneo, http://www.faq.unisi.it 




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Re: [asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

[Jonathan H]

I think it might be related to this?
https://issues.asterisk.org/jira/browse/ASTERISK-26391

I think I remember having to edit logger.conf - this is what mine
looks like now:
console => notice,warning,error
messages => notice,warning,error

Try that, restart asterisk and see if it works :)

On 30 November 2016 at 09:09, Michele Pinassi  wrote:
> Hi all,
>
> after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show
> what's happens. I've trying setting debug and verbose to 100 but
> nothing, no show. All commands works as expected but i can't what's
> happens on my asterisk server.
>
> asterisk*CLI> core show settings
>
> PBX Core settings
> -
>   Version: 14.2.0
>   Build Options:   LOADABLE_MODULES, BUILD_NATIVE, OPTIONAL_API
>   Maximum calls:   30 (Current 0)
>   Maximum open file handles:   1024
>   Root console verbosity:  100
>   Current console verbosity:   100
>   Debug level: 100
>   Maximum load average:0.90
>   Minimum free memory: 1 MB
>   Startup time:09:07:33
>   Last reload time:09:07:33
>   System:  Linux/3.16.0-4-686-pae built by root on
> i686 2016-11-28 14:50:24 UTC
>   System name:
>   Default language:en
>   Language prefix: Enabled
>   User name and group: /
>   Executable includes: Disabled
>   Transcode via SLIN:  Enabled
>   Transmit silence during rec: Disabled
>   Generic PLC: Enabled
>   Min DTMF duration::  80
>   RTP dynamic payload types:   96-127
>
> * Subsystems
>   -
>   Manager (AMI):   Disabled
>   Web Manager (AMI/HTTP):  Disabled
>   Call data records:   Disabled
>   Realtime Architecture (ARA): Disabled
>
> * Directories
>   -
>   Configuration file:  /etc/asterisk/asterisk.conf
>   Configuration directory: /etc/asterisk
>   Module directory:/usr/lib/asterisk/modules
>   Spool directory: /var/spool/asterisk
>   Log directory:   /var/log/asterisk
>   Run/Sockets directory:   /var/run/asterisk
>   PID file:/var/run/asterisk/asterisk.pid
>   VarLib directory:/var/lib/asterisk
>   Data directory:  /var/lib/asterisk
>   ASTDB:   /var/lib/asterisk/astdb
>   IAX2 Keys directory: /var/lib/asterisk/keys
>   AGI Scripts directory:   /var/lib/asterisk/agi-bin
>
> Any hint ?
>
> Michele
>
> --
> Michele Pinassi
> Responsabile Telefonia di Ateneo
> Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di 
> Siena
> tel: 0577.(23)5000 - central...@unisi.it
>
> Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di 
> Ateneo, http://www.faq.unisi.it
>
>
>
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>
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>
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