[asterisk-users] Difference in information realtime database table and Asterisk sip show peers

2021-04-20 Thread Jonas Kellens

Hello


can anyone explain to me why (and HOW) there is a difference in data 
between the Asterisk console "sip show peers" and the realtime MySQL 
configuration ?



Using : asterisk-certified-13.21-cert6


Asterisk console data :

/usr/sbin/asterisk -rx 'sip show peers' | grep 660091086
660091086/660091086   11.22.33.44 D  Yes    Yes    
55018    OK (31 ms)   Cached RT



/usr/sbin/asterisk -rx 'sip show peer 660091086'

  * Name   : 660091086
  Description  :
  Realtime peer: Yes, cached
  Secret   : 
   

  DTMFmode : rfc2833
  Timer T1 : 500
  Timer B  : 32000
  ToHost   :
  Addr->IP : 11.22.33.44:55018
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 660091086
  SIP Options  : (none)
  Codecs   : (alaw|g729|gsm)
  Auto-Framing : No
  Status   : OK (31 ms)
  Useragent    : Cisco/SPA508G-7.5.2
  Reg. Contact : sip:660091086@192.168.1.12:5064
  Qualify Freq : 12 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No
  RTCP Mux : Yes


MySQL database table data :

SELECT name, host, nat, type, qualify, fullcontact, ipaddr, port, 
regserver, regseconds, lastms, defaultuser FROM sip_buddies WHERE 
defaultuser='660091086'

+---+-+-++-+-++--+---+++-+
| name  | host    | nat | type   | qualify | 
fullcontact | ipaddr | port | regserver | regseconds | lastms | 
defaultuser |

+---+-+-++-+-++--+---+++-+
| 660091086 | dynamic | force_rport,comedia | friend | yes | 
|    |    0 |   |  0 |  0 | 660091086   |

+---+-+-++-+-++--+---+++-+



According to Asterisk console, the SIP peer is Registered and qualify 
(SIP OPTION) is fine (31 ms).


According to Mysql data, the SIP peer has no "fullcontact", no "ipaddr", 
no "port" and 0 (zero) regseconds and 0 (zero) lastms. While I would 
expect this to be filled in and/or has values higher than 0 (zero).




Is there an explanation for this "difference" in data ?!




Kind regards


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Re: [asterisk-users] dahdi-linux-complete-3.1.0+3.1.0 : issue on CentOS 7.9 but ot on CentOS 6.10

2021-02-22 Thread Jonas Kellens

Hello List


to answer my own question, and for whom it may interest, I no longer 
have the error about libtonezone.so with Dahdi version : 
dahdi-linux-complete-2.11.1+2.11.1


I don't know what the difference is between Dahdi 2.x and Dahdi 3.x but 
I can say that THERE IS somewhere a difference, as I experienced on 
Centos 7.9.




Kind regards.



Op 12-02-21 om 19:11 schreef Jonas Kellens:


Hello list


when installing latest DAHDI (dahdi-linux-complete-3.1.0+3.1.0) for 
usage with asterisk-certified-13.21-cert6 on CentOS 6.10 all works 
well when starting dahdi with "/sbin/service dahdi start".



But when installing the same DAHDI version in CentOS 7.9 I get the 
error :*/usr/sbin/dahdi_cfg: error while loading shared libraries: 
libtonezone.so.2: cannot open shared object file: No such file or 
directory*

when issuing "systemctl start dahdi.service"


Is there something missing on my CentOS 7.9 system to work with the 
latest DAHDI version ?


Or is there a better DAHDI version to be used on CentOS 7.9 ?


libtonezone is present on my CentOS 7.9 system :

[root@server admin]# locate libtonezone
/usr/lib/libtonezone.a
/usr/lib/libtonezone.la
/usr/lib/libtonezone.so
/usr/lib/libtonezone.so.1
/usr/lib/libtonezone.so.1.0
/usr/lib/libtonezone.so.2
/usr/lib/libtonezone.so.2.0
/usr/lib/libtonezone.so.2.0.0




Kind regards.





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[asterisk-users] dahdi-linux-complete-3.1.0+3.1.0 : issue on CentOS 7.9 but ot on CentOS 6.10

2021-02-12 Thread Jonas Kellens

Hello list


when installing latest DAHDI (dahdi-linux-complete-3.1.0+3.1.0) for 
usage with asterisk-certified-13.21-cert6 on CentOS 6.10 all works well 
when starting dahdi with "/sbin/service dahdi start".



But when installing the same DAHDI version in CentOS 7.9 I get the error 
:*/usr/sbin/dahdi_cfg: error while loading shared libraries: 
libtonezone.so.2: cannot open shared object file: No such file or directory*

when issuing "systemctl start dahdi.service"


Is there something missing on my CentOS 7.9 system to work with the 
latest DAHDI version ?


Or is there a better DAHDI version to be used on CentOS 7.9 ?


libtonezone is present on my CentOS 7.9 system :

[root@server admin]# locate libtonezone
/usr/lib/libtonezone.a
/usr/lib/libtonezone.la
/usr/lib/libtonezone.so
/usr/lib/libtonezone.so.1
/usr/lib/libtonezone.so.1.0
/usr/lib/libtonezone.so.2
/usr/lib/libtonezone.so.2.0
/usr/lib/libtonezone.so.2.0.0




Kind regards.

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Re: [asterisk-users] defaultexpiry & maxexpiry on peer level

2019-10-23 Thread Jonas Kellens

Hello

registration time is set to low value because when a network interuption 
occurs, it takes long time for the endpoint (Phone,...) to re-register. 
That is my expercience.



But about my question : is there a "on peer level" setting possible ?




Op 08-10-19 om 19:40 schreef G.Jacobsen:

Why do you want such minimal registration time?

On Tuesday, 8 October 2019, 17:23:03 EEST, Jonas Kellens 
 wrote:



Hello

is it possible to determine the SIP.conf parameters 'defaultexpirty' 
and 'maxexpiry' on a peer basis ?


My default value is 300 seconds, but some specific SIP-clients can 
only send a SIP REGISTER every 3600 seconds. In current configuration 
these SIP peers now become "Unreachable" after 300 seconds.



Or is there another way to differentiate ?


Kind regards.


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[asterisk-users] defaultexpiry & maxexpiry on peer level

2019-10-08 Thread Jonas Kellens

Hello

is it possible to determine the SIP.conf parameters 'defaultexpirty' and 
'maxexpiry' on a peer basis ?


My default value is 300 seconds, but some specific SIP-clients can only 
send a SIP REGISTER every 3600 seconds. In current configuration these 
SIP peers now become "Unreachable" after 300 seconds.



Or is there another way to differentiate ?


Kind regards.


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[asterisk-users] Asterisk manager : core show hints

2019-08-22 Thread Jonas Kellens

Hello

I see on the CLI :

tst*CLI> core show hints
    -= Registered Asterisk Dial Plan Hints =-
 50@blf  : SIP/testacc7 
State:Idle    Watchers  3
   6001@blf   : Custom:q-6001 State:Idle    
Watchers  1
  5@blf  : SIP/testacc6 
State:Unavailable Watchers  1



Is there a way to get this info through the manager API ?



Kind regards

Jonas.

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[asterisk-users] BLF NOTIFY Subscription-State: terminated; reason=timeout

2019-07-08 Thread Jonas Kellens

Hello


I notice that BLF-buttons on my IP-phone are greyed out and again active 
after some time. This goes on and on...


When looking at Asterisk CLI I see in the SIP NOTIFY :

Subscription-State: terminated;reason=timeout


The BLF-buttons turn on again after a new SIP SUBSCRIBE from my 
IP-phone. This SUBSCRIBE happens every 120 seconds.


They fade out after about 85 seconds. Then after 35 seconds they come 
back up.



In sip.conf I have the following settings :
maxexpiry=100
minexpiry=60
defaultexpiry=100

But that does not seem to change much. Except for SIP REGISTER's : they 
now happen every 50 seconds (yes, 50) in stead of every 120 seconds.



Can anyone help me to find the right timer to tune so I can have stable 
BLF buttons ?




Thanks.
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Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video

2019-05-30 Thread Jonas Kellens

Hello

is this mailing list still active ?




Op 10-05-19 om 14:10 schreef Jonas Kellens:


Hello

I am trying to set up webRTC video calls from my Chrome webbrowser 
(Fedora) to my Chrome webbrowser (Windows 10).


There is local video input (I can see myself), but never video on the 
receiving side.


This is the case in both directions (so it makes no difference which 
peer is calling which peer).



Both webRTC SIP peers have opus and H264 codec in their peer definition :

  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (75 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (47 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


In general sip.conf I have :

videosupport=yes
disallow=all
allow=alaw
allow=opus
allow=h264


When one peer makes a SIP INVITE for a video call, it is clear to me 
that the necessary codec information is present (this all looks fine 
to me) :


(calling webRTC client)

SIP Debugging Enabled for IP: 99.99.255.55
[May 10 10:45:24]
[May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 --->
[May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0
[May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692
[May 10 10:45:24] Max-Forwards: 70
[May 10 10:45:24] To: 
[May 10 10:45:24] From: "WC User Chrome" 
;tag=sdmbqkquhe

[May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu
[May 10 10:45:24] CSeq: 4132 INVITE
[May 10 10:45:24] Contact: 
[May 10 10:45:24] Allow: 
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

[May 10 10:45:24] Supported: outbound
[May 10 10:45:24] User-Agent: SIP.js/0.12.0
[May 10 10:45:24] Content-Type: application/sdp
[May 10 10:45:24] Content-Length: 5098
[May 10 10:45:24]
[May 10 10:45:24] v=0
[May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1
[May 10 10:45:24] s=-
[May 10 10:45:24] t=0 0
[May 10 10:45:24] a=group:BUNDLE audio video
[May 10 10:45:24] a=msid-semantic: WMS 
I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 
106 105 13 110 112 113 126

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 
192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 
network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 
192.168.1.110 9 typ host tcptype active generation 0 network-id 1 
network-cost 10

[May 10 10:45:24] a=ice-ufrag:y8md
[May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le
[May 10 10:45:24] a=ice-options:trickle
[May 10 10:45:24] a=fingerprint:sha-256 
C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B

[May 10 10:45:24] a=setup:actpass
[May 10 10:45:24] a=mid:audio
[May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[May 10 10:45:24] a=sendrecv
[May 10 10:45:24] a=rtcp-mux
[May 10 10:45:24] a=rtpmap:111 opus/48000/2
[May 10 10:45:24] a=rtcp-fb:111 transport-cc
[May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1
[May 10 10:45:24] a=rtpmap:103 ISAC/16000
[May 10 10:45:24] a=rtpmap:104 ISAC/32000
[May 10 10:45:24] a=rtpmap:9 G722/8000
[May 10 10:45:24] a=rtpmap:0 PCMU/8000
[May 10 10:45:24] a=rtpmap:8 PCMA/8000
[May 10 10:45:24] a=rtpmap:106 CN/32000
[May 10 10:45:24] a=rtpmap:105 CN/16000
[May 10 10:45:24] a=rtpmap:13 CN/8000
[May 10 10:45:24] a=rtpmap:110 telephone-event/48000
[May 10 10:45:24] a=rtpmap:112 telephone-event/32000
[May 10 10:45:24] a=rtpmap:113 telephone-event/16000
[May 10 10:45:24] a=rtpmap:126 telephone-event/8000
[May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ
[May 10 10:45:24] a=ssrc:401971016 
msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E 
f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] a=ssrc:401971016 
mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] a=ssrc:401971016 
label:f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 
102 123 127 122 125 107 108 109 124

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 
192.168.1.110 48086 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 
network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 
192

Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video

2019-05-28 Thread Jonas Kellens

Hello

is this mailing list still active ?




Op 10-05-19 om 14:10 schreef Jonas Kellens:


Hello

I am trying to set up webRTC video calls from my Chrome webbrowser 
(Fedora) to my Chrome webbrowser (Windows 10).


There is local video input (I can see myself), but never video on the 
receiving side.


This is the case in both directions (so it makes no difference which 
peer is calling which peer).



Both webRTC SIP peers have opus and H264 codec in their peer definition :

  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (75 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (47 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


In general sip.conf I have :

videosupport=yes
disallow=all
allow=alaw
allow=opus
allow=h264


When one peer makes a SIP INVITE for a video call, it is clear to me 
that the necessary codec information is present (this all looks fine 
to me) :


(calling webRTC client)

SIP Debugging Enabled for IP: 99.99.255.55
[May 10 10:45:24]
[May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 --->
[May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0
[May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692
[May 10 10:45:24] Max-Forwards: 70
[May 10 10:45:24] To: 
[May 10 10:45:24] From: "WC User Chrome" 
;tag=sdmbqkquhe

[May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu
[May 10 10:45:24] CSeq: 4132 INVITE
[May 10 10:45:24] Contact: 
[May 10 10:45:24] Allow: 
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

[May 10 10:45:24] Supported: outbound
[May 10 10:45:24] User-Agent: SIP.js/0.12.0
[May 10 10:45:24] Content-Type: application/sdp
[May 10 10:45:24] Content-Length: 5098
[May 10 10:45:24]
[May 10 10:45:24] v=0
[May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1
[May 10 10:45:24] s=-
[May 10 10:45:24] t=0 0
[May 10 10:45:24] a=group:BUNDLE audio video
[May 10 10:45:24] a=msid-semantic: WMS 
I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 
106 105 13 110 112 113 126

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 
192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 
network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 
192.168.1.110 9 typ host tcptype active generation 0 network-id 1 
network-cost 10

[May 10 10:45:24] a=ice-ufrag:y8md
[May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le
[May 10 10:45:24] a=ice-options:trickle
[May 10 10:45:24] a=fingerprint:sha-256 
C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B

[May 10 10:45:24] a=setup:actpass
[May 10 10:45:24] a=mid:audio
[May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[May 10 10:45:24] a=sendrecv
[May 10 10:45:24] a=rtcp-mux
[May 10 10:45:24] a=rtpmap:111 opus/48000/2
[May 10 10:45:24] a=rtcp-fb:111 transport-cc
[May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1
[May 10 10:45:24] a=rtpmap:103 ISAC/16000
[May 10 10:45:24] a=rtpmap:104 ISAC/32000
[May 10 10:45:24] a=rtpmap:9 G722/8000
[May 10 10:45:24] a=rtpmap:0 PCMU/8000
[May 10 10:45:24] a=rtpmap:8 PCMA/8000
[May 10 10:45:24] a=rtpmap:106 CN/32000
[May 10 10:45:24] a=rtpmap:105 CN/16000
[May 10 10:45:24] a=rtpmap:13 CN/8000
[May 10 10:45:24] a=rtpmap:110 telephone-event/48000
[May 10 10:45:24] a=rtpmap:112 telephone-event/32000
[May 10 10:45:24] a=rtpmap:113 telephone-event/16000
[May 10 10:45:24] a=rtpmap:126 telephone-event/8000
[May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ
[May 10 10:45:24] a=ssrc:401971016 
msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E 
f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] a=ssrc:401971016 
mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] a=ssrc:401971016 
label:f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 
102 123 127 122 125 107 108 109 124

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 
192.168.1.110 48086 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 
network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 
192

[asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video

2019-05-10 Thread Jonas Kellens

Hello

I am trying to set up webRTC video calls from my Chrome webbrowser 
(Fedora) to my Chrome webbrowser (Windows 10).


There is local video input (I can see myself), but never video on the 
receiving side.


This is the case in both directions (so it makes no difference which 
peer is calling which peer).



Both webRTC SIP peers have opus and H264 codec in their peer definition :

  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (75 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (47 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


In general sip.conf I have :

videosupport=yes
disallow=all
allow=alaw
allow=opus
allow=h264


When one peer makes a SIP INVITE for a video call, it is clear to me 
that the necessary codec information is present (this all looks fine to 
me) :


(calling webRTC client)

SIP Debugging Enabled for IP: 99.99.255.55
[May 10 10:45:24]
[May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 --->
[May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0
[May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692
[May 10 10:45:24] Max-Forwards: 70
[May 10 10:45:24] To: 
[May 10 10:45:24] From: "WC User Chrome" 
;tag=sdmbqkquhe

[May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu
[May 10 10:45:24] CSeq: 4132 INVITE
[May 10 10:45:24] Contact: 
[May 10 10:45:24] Allow: 
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

[May 10 10:45:24] Supported: outbound
[May 10 10:45:24] User-Agent: SIP.js/0.12.0
[May 10 10:45:24] Content-Type: application/sdp
[May 10 10:45:24] Content-Length: 5098
[May 10 10:45:24]
[May 10 10:45:24] v=0
[May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1
[May 10 10:45:24] s=-
[May 10 10:45:24] t=0 0
[May 10 10:45:24] a=group:BUNDLE audio video
[May 10 10:45:24] a=msid-semantic: WMS I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 
105 13 110 112 113 126

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 
34197 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 network-id 
1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 
9 typ host tcptype active generation 0 network-id 1 network-cost 10

[May 10 10:45:24] a=ice-ufrag:y8md
[May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le
[May 10 10:45:24] a=ice-options:trickle
[May 10 10:45:24] a=fingerprint:sha-256 
C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B

[May 10 10:45:24] a=setup:actpass
[May 10 10:45:24] a=mid:audio
[May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[May 10 10:45:24] a=sendrecv
[May 10 10:45:24] a=rtcp-mux
[May 10 10:45:24] a=rtpmap:111 opus/48000/2
[May 10 10:45:24] a=rtcp-fb:111 transport-cc
[May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1
[May 10 10:45:24] a=rtpmap:103 ISAC/16000
[May 10 10:45:24] a=rtpmap:104 ISAC/32000
[May 10 10:45:24] a=rtpmap:9 G722/8000
[May 10 10:45:24] a=rtpmap:0 PCMU/8000
[May 10 10:45:24] a=rtpmap:8 PCMA/8000
[May 10 10:45:24] a=rtpmap:106 CN/32000
[May 10 10:45:24] a=rtpmap:105 CN/16000
[May 10 10:45:24] a=rtpmap:13 CN/8000
[May 10 10:45:24] a=rtpmap:110 telephone-event/48000
[May 10 10:45:24] a=rtpmap:112 telephone-event/32000
[May 10 10:45:24] a=rtpmap:113 telephone-event/16000
[May 10 10:45:24] a=rtpmap:126 telephone-event/8000
[May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ
[May 10 10:45:24] a=ssrc:401971016 
msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E 
f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] a=ssrc:401971016 
mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] a=ssrc:401971016 
label:f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 
102 123 127 122 125 107 108 109 124

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 
48086 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 network-id 
1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 
9 typ host tcptype active generation 0 network-id 1 network-cost 10

[May 10 10:45:24] 

Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?

2018-10-12 Thread Jonas Kellens

Hello

thank you for your answer.

This does not happen all the time. It happens about once every 4 months. 
I just can not pinpoint WHEN exactly it occurs. I just see in the 
verbose logfile that it occurs after a MYSQL insert/update/delete statement.


If Asterisk 13 handels MYSQL connections in a better way, then indeed I 
should look for upgrade.




Kind regards.



Op 05-10-18 om 01:25 schreef John Novack:
As others have said, clearly it ISN'T "just working" or you would not 
have posted the question


To state again, I am using Version 13, though a few minor revisions 
behind, with MySql, on CentOS 6 and have no rebooting or other MySql 
related issues


Clearly you need to state in more detail what issues remain, once you 
migrate to AT LEAST 13.xx, and state your OS after becoming current 
with Asterisk, MySql and the OS


I use MySql on every incoming call, and also maintain call detail 
records in MySql for every call, and it just simply works, and has for 
some time.


Although I may be using it quite differently that you, it simply works.
Is this a newly developing issue, or has it persisted for some time
What if any changes have been made to the dialplan etc?

Have you considered a strictly hardware issue? Memory? HD? MB??

The crystal ball is very cloudy on this one!

John Novack


Jonas Kellens wrote:


Hello

thank you for your answer.

If I read your (and others) reaction correctly I can conclude that 
this is an Asterisk problem and not a problem of MySQL or dialplan 
logic ?



You should know that the MySQL database is heavily questioned :


mysql> show status like '%onn%';
+--++
| Variable_name    | Value  |
+--++
| Aborted_connects | 469    |
| Connections  | 132762 |
| Max_used_connections | 8  |
| Ssl_client_connects  | 0  |
| Ssl_connect_renegotiates | 0  |
| Ssl_finished_connects    | 0  |
| Threads_connected    | 3  |
+--++
7 rows in set (0.00 sec)



I stick to 1.8 because it just works. I had some issues with version 
11 and 13 in the past.



Regards

Jonas.


Op 04-10-18 om 17:49 schreef John Novack:

Woefully out of date.
You really need to put your efforts into at least a modest upgrade
I use version 13 with MySql queries built into the dialplan on 
CentOs 6 and have NO such issues, either performance or any restart 
or reboot. It simply works


I never used either 1.6 or 1.8, going from 1.4 to version 11, which 
did require some syntax changes to the dialplan.


Given that even version 11 is EOL, you really need to put your 
efforts into doing the migration rather than tracking this one down


JMO

John Novack



Jonas Kellens wrote:


Hello

using Asterisk 1.8.32.

I notice that there is a spontaneous reboot of the Asterisk system 
from time to time.


When I look in the logs (verbose file) I noticed that every time 
this occurs it's at a moment that there is a MySQL action, be it a 
lookup or an insert/update/delete.


I must say I do have some MySQL queries that occur in my dialplan 
when a call comes in, to look up different actions to perform on 
this call.



An idea how to overcome this problem ? Seems a "performance" issue, 
no ?!


Is it better to have these MySQL queries to be done by an external 
script (like a php script that I call with the System()-command or 
a SHELL()-command) ?



Here are some examples from the verbose file.



[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- 
Executing [s@sub-GetAlertInfo:3] MYSQL("SIP/SipAgenT01-317d", 
"Connect connid localhost myuser mypwd myDB") in new stack
[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- 
Executing [s@sub-GetAlertInfo:5] MYSQL("SIP/SipAgenT01-317d", 
"Query resultid 1 SELECT uri, callinfo FROM distringtone WHERE 
onoff='1'") in new stack
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/logger.conf': [Aug 22 15:19:18] 
VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] 
VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] manager.c: [Aug 22 15:19:18]   == 
Manager registered action DataGet
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/codecs.conf': [Aug 22 15:19:18] 
VERBOSE[3306] config.c: [Aug 22 15:19:18] == Found
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18]  
Asterisk Dynamic Loader Starting:
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/modules.conf': [Aug 22 15:19:18] 
VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/res_config_mysql.conf': [Aug 22 15:19:18

Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?

2018-10-04 Thread Jonas Kellens

Hello

thank you for your answer.

If I read your (and others) reaction correctly I can conclude that this 
is an Asterisk problem and not a problem of MySQL or dialplan logic ?



You should know that the MySQL database is heavily questioned :


mysql> show status like '%onn%';
+--++
| Variable_name    | Value  |
+--++
| Aborted_connects | 469    |
| Connections  | 132762 |
| Max_used_connections | 8  |
| Ssl_client_connects  | 0  |
| Ssl_connect_renegotiates | 0  |
| Ssl_finished_connects    | 0  |
| Threads_connected    | 3  |
+--++
7 rows in set (0.00 sec)



I stick to 1.8 because it just works. I had some issues with version 11 
and 13 in the past.



Regards

Jonas.


Op 04-10-18 om 17:49 schreef John Novack:

Woefully out of date.
You really need to put your efforts into at least a modest upgrade
I use version 13 with MySql queries built into the dialplan on CentOs 
6 and have NO such issues, either performance or any restart or 
reboot. It simply works


I never used either 1.6 or 1.8, going from 1.4 to version 11, which 
did require some syntax changes to the dialplan.


Given that even version 11 is EOL, you really need to put your efforts 
into doing the migration rather than tracking this one down


JMO

John Novack



Jonas Kellens wrote:


Hello

using Asterisk 1.8.32.

I notice that there is a spontaneous reboot of the Asterisk system 
from time to time.


When I look in the logs (verbose file) I noticed that every time this 
occurs it's at a moment that there is a MySQL action, be it a lookup 
or an insert/update/delete.


I must say I do have some MySQL queries that occur in my dialplan 
when a call comes in, to look up different actions to perform on this 
call.



An idea how to overcome this problem ? Seems a "performance" issue, no ?!

Is it better to have these MySQL queries to be done by an external 
script (like a php script that I call with the System()-command or a 
SHELL()-command) ?



Here are some examples from the verbose file.



[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing 
[s@sub-GetAlertInfo:3] MYSQL("SIP/SipAgenT01-317d", "Connect 
connid localhost myuser mypwd myDB") in new stack
[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing 
[s@sub-GetAlertInfo:5] MYSQL("SIP/SipAgenT01-317d", "Query 
resultid 1 SELECT uri, callinfo FROM distringtone WHERE onoff='1'") 
in new stack
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/logger.conf': [Aug 22 15:19:18] VERBOSE[3306] 
config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] 
VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] manager.c: [Aug 22 15:19:18]   == 
Manager registered action DataGet
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/codecs.conf': [Aug 22 15:19:18] VERBOSE[3306] 
config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] Asterisk 
Dynamic Loader Starting:
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/modules.conf': [Aug 22 15:19:18] VERBOSE[3306] 
config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == 
Parsing '/etc/asterisk/res_config_mysql.conf': [Aug 22 15:19:18] 
VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] res_config_mysql.c: [Aug 22 
15:19:18]   == MySQL RealTime driver loaded.
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] 
res_config_mysql.so => (MySQL RealTime Configuration Driver)




[Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- 
Executing [s@sub-GetSipAccountdetails:3] 
MYSQL("SIP/SipAgenT01-4184", "Connect connid localhost myuser 
mypwd myDB") in new stack
[Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- 
Executing [s@sub-GetSipAccountdetails:4] 
MYSQL("SIP/SipAgenT01-4184", "Query resultid 1 SELECT 
SIPusername, currstatus, available FROM tbl_SIP WHERE ID="800"") in 
new stack
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32]   == 
Parsing '/etc/asterisk/logger.conf': [Aug 22 16:23:32] VERBOSE[24309] 
config.c: [Aug 22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32]   == 
Parsing '/etc/asterisk/asterisk.conf': [Aug 22 16:23:32] 
VERBOSE[24309] config.c: [Aug 22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] manager.c: [Aug 22 16:23:32]   == 
Manager registered action DataGet
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32]   == 
Parsing '/etc/asterisk/codecs.conf': [Aug 22 

[asterisk-users] Spontaneous reboot due to MySQL lookups ?

2018-10-04 Thread Jonas Kellens

Hello

using Asterisk 1.8.32.

I notice that there is a spontaneous reboot of the Asterisk system from 
time to time.


When I look in the logs (verbose file) I noticed that every time this 
occurs it's at a moment that there is a MySQL action, be it a lookup or 
an insert/update/delete.


I must say I do have some MySQL queries that occur in my dialplan when a 
call comes in, to look up different actions to perform on this call.



An idea how to overcome this problem ? Seems a "performance" issue, no ?!

Is it better to have these MySQL queries to be done by an external 
script (like a php script that I call with the System()-command or a 
SHELL()-command) ?



Here are some examples from the verbose file.



[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- 
Executing [s@sub-GetAlertInfo:3] MYSQL("SIP/SipAgenT01-317d", 
"Connect connid localhost myuser mypwd myDB") in new stack
[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- 
Executing [s@sub-GetAlertInfo:5] MYSQL("SIP/SipAgenT01-317d", "Query 
resultid 1 SELECT uri, callinfo FROM distringtone WHERE onoff='1'") in 
new stack
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/logger.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: 
[Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: 
[Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] manager.c: [Aug 22 15:19:18] == Manager 
registered action DataGet
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/codecs.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: 
[Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] Asterisk 
Dynamic Loader Starting:
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/modules.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: 
[Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/res_config_mysql.conf': [Aug 22 15:19:18] VERBOSE[3306] 
config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] res_config_mysql.c: [Aug 22 15:19:18]   
== MySQL RealTime driver loaded.
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] 
res_config_mysql.so => (MySQL RealTime Configuration Driver)




[Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- 
Executing [s@sub-GetSipAccountdetails:3] 
MYSQL("SIP/SipAgenT01-4184", "Connect connid localhost myuser mypwd 
myDB") in new stack
[Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- 
Executing [s@sub-GetSipAccountdetails:4] 
MYSQL("SIP/SipAgenT01-4184", "Query resultid 1 SELECT SIPusername, 
currstatus, available FROM tbl_SIP WHERE ID="800"") in new stack
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/logger.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: 
[Aug 22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/asterisk.conf': [Aug 22 16:23:32] VERBOSE[24309] 
config.c: [Aug 22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] manager.c: [Aug 22 16:23:32] == Manager 
registered action DataGet
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/codecs.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: 
[Aug 22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] loader.c: [Aug 22 16:23:32] Asterisk 
Dynamic Loader Starting:
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/modules.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: 
[Aug 22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/res_config_mysql.conf': [Aug 22 16:23:32] VERBOSE[24309] 
config.c: [Aug 22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] res_config_mysql.c: [Aug 22 16:23:32]   
== MySQL RealTime driver loaded.
[Aug 22 16:23:32] VERBOSE[24309] loader.c: [Aug 22 16:23:32] 
res_config_mysql.so => (MySQL RealTime Configuration Driver)




[Oct  4 10:11:25] VERBOSE[4944] pbx.c: [Oct  4 10:11:25] -- 
Executing [s@sub-settings:16] MYSQL("SIP/SipAgenT01-08cb", "Connect 
connid localhost myuser mypwd myDB") in new stack
[Oct  4 10:11:25] VERBOSE[4944] pbx.c: [Oct  4 10:11:25] -- 
Executing [s@sub-settings:17] MYSQL("SIP/SipAgenT01-08cb", "Query 
resultid 1 SELECT blockID from DID where DID=987654321") in new stack
[Oct  4 10:11:29] VERBOSE[4961] config.c: [Oct  4 10:11:29]   == Parsing 
'/etc/asterisk/asterisk.conf': [Oct  4 10:11:29] VERBOSE[4961] config.c: 
[Oct  4 10:11:29]   == Found
[Oct  4 10:11:29] VERBOSE[4961] manager.c: [Oct  4 10:11:29] == Manager 
registered action DataGet
[Oct  4 10:11:29] VERBOSE[4961] config.c: [Oct  4 10:11:29]   == Parsing 
'/etc/asterisk/codecs.conf': [Oct  4 

[asterisk-users] doing dnsmgr_lookup for

2018-05-31 Thread Jonas Kellens

Hello list

is there a way to limit the number of dns lookup's for 1 and the same host ?

I see on Asterisk CLI a flood of :

[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'
[May 31 15:45:37]    > doing dnsmgr_lookup for 'proxy1.sip.x2reg.be'


I have several sip peer definitions (sip trunks) pointing at this same host.


Kind regards.


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

2017-07-25 Thread Jonas Kellens
For those never getting a decend answer by the community on this 
mailinglist, I share my solution to my video problem :


preferred_codec_only=no

(I had this on 'yes')



Op 26-06-17 om 14:43 schreef Jonas Kellens:


Hello

this is the debug output of a test video call. You see codec 
negotiation but at the end only alaw is chosen and gone is the video !



[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our native formats are 
0x8 (alaw)
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Joint capabilities are 
0x8 (alaw)
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our capabilities are 
0x20010a (gsm|alaw|g729|h264)
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** AST_CODEC_CHOOSE 
formats are 0x8 (alaw)
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our preferred formats 
from the incoming channel are 0x8 (alaw)
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: This channel can handle 
video! HOLLYWOOD next!


[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: This call needs video offers!
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: ** Our capability: 0x2a 
(gsm|alaw|h264) Video flag: False Text flag: False

[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: ** Our prefcodec: 0x8 (alaw)
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: -- Done with adding codecs 
to SDP
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Done building SDP. Settling 
with this capability: 0x2a (gsm|alaw|h264)


[Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Initializing initreq for 
method INVITE - callid 6ca7f3fb70b56f6f6f9373b776cd495d@11.22.33.44:5060
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  0 [ 54]: INVITE 
sip:sipaccount12@192.168.1.111:50104;ob SIP/2.0
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  1 [ 65]: Via: 
SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK54e24150;rport
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  2 [ 16]: 
Max-Forwards: 70
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  3 [ 62]: From: "My 
Account" <sip:71@11.22.33.44>;tag=as130bc3f0
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  4 [ 45]: To: 
<sip:sipaccount12@192.168.1.111:50104;ob>
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  5 [ 37]: Contact: 
<sip:71@11.22.33.44:5060>
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  6 [ 61]: Call-ID: 
6ca7f3fb70b56f6f6f9373b776cd495d@11.22.33.44:5060
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  7 [ 16]: CSeq: 102 
INVITE
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  8 [ 21]: 
User-Agent: mydomain
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header  9 [ 35]: Date: 
Mon, 26 Jun 2017 12:20:55 GMT
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header 10 [ 90]: Allow: 
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header 11 [ 19]: 
Supported: replaces
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header 12 [ 42]: 
Alert-Info: <http://127.0.0.1>;info=intern
[Jun 26 14:20:55] DEBUG[28609] chan_sip.c:  Header 14 [ 29]: 
Content-Type: application/sdp


...

[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  0 [ 14]: SIP/2.0 200 OK
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  1 [ 94]: Via: 
SIP/2.0/UDP 
11.22.33.44:5060;rport=5060;received=11.22.33.44;branch=z9hG4bK54e24150
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  2 [ 61]: Call-ID: 
6ca7f3fb70b56f6f6f9373b776cd495d@11.22.33.44:5060
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  3 [ 62]: From: "My 
Account" <sip:71@11.22.33.44>;tag=as130bc3f0
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  4 [ 76]: To: 
<sip:sipaccount12@192.168.1.111;ob>;tag=8a1f91570e9f434c9da9aca27ded7fb9
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  5 [ 16]: CSeq: 102 
INVITE
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  6 [ 96]: Allow: 
PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, 
REFER, MESSAGE, OPTIONS
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  7 [ 50]: Contact: 
<sip:sipaccount12@192.168.1.111:50104;ob>
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  8 [ 46]: Supported: 
replaces, 100rel, timer, norefersub
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header  9 [ 29]: 
Content-Type: application/sdp
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header 10 [ 19]: 
Content-Length: 469

[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:  Header 11 [  0]:
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body  0 [  3]: v=0
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body  1 [ 46]: o=- 
3707475663 3707475664 IN IP4 192.168.1.111

[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body  2 [  9]: s=pjmedia
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body  3 [  8]: b=AS:352
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body  4 [  5]: t=0 0
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body  5 [  9]: a=X-nat:0
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body  6 [ 26]: m=audio 
4000 RTP/AVP 8 101
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body  7 [ 22]: c=IN IP4 
192.168.1.111

[Jun 26 14:20:57] DEBUG[1932] chan

Re: [asterisk-users] BLF and Call Queues

2017-07-14 Thread Jonas Kellens

Hello

concerning this question of aug 2012, I am now using 1.8.32.2 and it 
seems that the code of app_queue.c has changed.


The function ast_devstate_changed() is no longer used. Can anyone tell 
me what it is replaced with ?




Kind regards


Op 18-08-12 om 12:42 schreef Alec Davis:
  


-Original Message-
From: Alec Davis [mailto:siva...@paradise.net.nz]
Sent: Saturday, 18 August 2012 10:36 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] BLF and Call Queues



I've seen this post. That's why I thought it was possible.
I'm using 1.8.11

What is the difference between this post and asterisk 1.8.11 ?



The patch hasn't been accepted by the community, thus isn't
in asterisk trunk or any asterisk branches.

Alec

Jonas;
In case you had't seen it, the patch is available from review board
https://reviewboard.asterisk.org/r/1619/ using the 'Download Diff' link at
the top right of the review.
Or directly form here https://reviewboard.asterisk.org/r/1619/diff/raw/

Alec


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

2017-06-26 Thread Jonas Kellens
57] DEBUG[1932] chan_sip.c:Body 11 [ 20]: a=rtpmap:8 
PCMA/8000
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 12 [ 33]: a=rtpmap:101 
telephone-event/8000

[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 13 [ 15]: a=fmtp:101 0-16
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 14 [ 23]: m=video 4002 
RTP/AVP 99
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 15 [ 22]: c=IN IP4 
192.168.1.111

[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 16 [ 13]: b=TIAS:256000
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 17 [ 32]: a=rtcp:4003 
IN IP4 192.168.1.111

[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 18 [ 10]: a=sendrecv
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 19 [ 22]: a=rtpmap:99 
H264/9
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c:Body 20 [ 55]: a=fmtp:99 
profile-level-id=42000a; packetization-mode=0


[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: SIP response 200 to standard 
invite
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP 
v=0... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP 
o=- 3707475663 3707475664 IN IP4 192.168.1.111... OK.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP 
s=pjmedia... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP 
b=AS:352... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP 
t=0 0... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP 
a=X-nat:0... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 8 based on m 
type on 0x7efde80a5930
[Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 101 based on 
m type on 0x7efde80a5930


[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) 
SDP c=IN IP4 192.168.1.111... OK.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) 
SDP b=TIAS:64000... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) 
SDP a=rtcp:4001 IN IP4 192.168.1.111... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) 
SDP a=sendrecv... OK.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) 
SDP a=rtpmap:8 PCMA/8000... OK.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) 
SDP a=rtpmap:101 telephone-event/8000... OK.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) 
SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 99 based on 
m type on 0x7efde80a47f0
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) 
SDP c=IN IP4 192.168.1.111... OK.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) 
SDP b=TIAS:256000... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) 
SDP a=rtcp:4003 IN IP4 192.168.1.111... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) 
SDP a=sendrecv... UNSUPPORTED OR FAILED.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) 
SDP a=rtpmap:99 H264/9... OK.
[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) 
SDP a=fmtp:99 profile-level-id=42000a; packetization-mode=0... 
UNSUPPORTED OR FAILED.


[Jun 26 14:20:57] DEBUG[1932] chan_sip.c: We're settling with these 
formats: 0x8 (alaw)






Op 21-04-17 om 16:33 schreef Derek Bolichowski:


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Friday, April 21, 2017 10:18 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>

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

Hello


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





Kind regards.

J.




Why not try removing all codecs from the SIP Peer (deny all, allow 
only H264), unregister the peer, and try a video call again?  If it 
works, try adding G711 back but keep H264 at the top of the priority.






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Re: [asterisk-users] Let's encrypt privkey : Specified certificate file could not be used

2017-06-03 Thread Jonas Kellens

Hello James

I am running asterisk as root, just to 'disable' all issues related to 
file rights. So this should not be the problem.



Kind regards.


Op 03-06-17 om 08:09 schreef James Cloos:

"JK" == Jonas Kellens <jonas.kell...@telenet.be> writes:

JK> [Jun  2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441
JK> ast_rtp_dtls_set_configuration: Specified certificate file
JK> '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance
JK> '0x7f920c538a78' could not be used

That error means that openssl's SSL_CTX_use_certificate_file() returned
an error.

The later error is just a result of that one.

Does the uid/gid used for asterisk have access to the key?

If the uid you use for asterisk is called asterisk, run this as root:

su -c 'cat /etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' - asterisk

If it fails, then the problem is permissions.

You may need to alter the permissions on /etc/letsencrypt to allow
non-root uids to access the symlinks and their targets.

-JimC


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[asterisk-users] Let's encrypt privkey : Specified certificate file could not be used

2017-06-02 Thread Jonas Kellens

Hello

I get the following error when using our Let's Encrypt ssl certificate 
for webRTC calls :


[Jun  2 14:29:28]   == DTLS ECDH initialized (secp256r1), faster PFS enabled
[Jun  2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441 
ast_rtp_dtls_set_configuration: Specified certificate file 
'/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance 
'0x7f920c538a78' could not be used
[Jun  2 14:29:28] ERROR[27360][C-0ae5]: chan_sip.c:5941 
dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP 
configuration on RTP instance '0x7f920c538a78'


(ws.mydomain.tld is of course masked)


Any idea why Asterisk has a problem with the certificate ?


Kind regards.


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Re: [asterisk-users] Best way to know a call is being transfered

2017-05-29 Thread Jonas Kellens

Hello

thank you for your answer.

However this does not help me to know when a call is being transfered.

My question is simple : if A calls B, and then B tranfers (unattened or 
attended) the call to C, how can I know this happens ?? I see it 
happening on the CLI, but how can I "catch" this, for example in the 
dialplan logic ? Or through AMI perhaps ?




Kind regards.

J.



Op 29-05-17 om 10:16 schreef Jonathan H:

Well, once you've upgraded to a version of Asterisk which didn't
become "EOL - DO NOT USE - NO FIXES" (!) almost 2 years ago, then you
might be able use logging which was introduced 5 years ago in Asterisk
11. Although the "transfers" section in the info below says it "can be
a little tricky...". Read on!

https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging



Call ID Logging (which has nothing to do with caller ID) is a new
feature of Asterisk 11 intended to help administrators and support
givers to more quickly understand problems that occur during the
course of calls. Channels are now bound to call identifiers which can
be shared among a number of channels, threads, and other consumers.

Transfers

Transfers can be a little tricky to follow with the call ID logging
feature. As a general rule, an attended transfer will always result in
a new call ID being made because a separate call must occur between
the party that initiates the transfer and whatever extension is going
to receive it. Once the attended transfer is completed, the channel
that was transferred will use the Call ID created when the transferrer
called the recipient.

Blind transfers are slightly more variable. If a SIP peer 'peer1'
calls another SIP peer 'peer2' via the dial application and peer2
blind transfers peer1 elsewhere, the call ID will persist. If on the
other hand, peer1 blind transfers peer2 at this point a new call ID
will be created. When peer1 transfers peer2, peer2 has a new channel
created which enters the PBX for the first time, so it creates a new
call ID. When peer1 is transferred, it simply resumes running PBX, so
the call is still considered the same call. By setting the debug level
to 3 for the channel internal API (channel_internal_api.c), all call
ID settings for every channel will be logged and this may be able to
help when trying to keep track of calls through multiple transfers.


On 29 May 2017 at 08:17, Jonas Kellens <jonas.kell...@telenet.be> wrote:

Hello

using Asterisk 1.8.32.3.

What is the best way of knowing a call is being transfered (attended and
unattended) ? And also knowing whereto (sip user) the call is being
transfered and who is the transferer ?

So I can log this information.



Kind regards.

J.


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[asterisk-users] Best way to know a call is being transfered

2017-05-29 Thread Jonas Kellens

Hello

using Asterisk 1.8.32.3.

What is the best way of knowing a call is being transfered (attended and 
unattended) ? And also knowing whereto (sip user) the call is being 
transfered and who is the transferer ?


So I can log this information.



Kind regards.

J.

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

2017-04-21 Thread 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 <mhter...@gmail.com>
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 <jonas.kell...@telenet.be> 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 <mhter...@gmail.com>
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 <jonas.kell...@telenet.be> 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)   (

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

2017-04-20 Thread 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 <mhter...@gmail.com>
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 <jonas.kell...@telenet.be> 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) 

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

2017-04-19 Thread 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] Define SIP fromuser field in Dial()-command

2017-04-14 Thread Jonas Kellens

Hello


function sip_header is read-only.




Kind regards.

J.



On 14-04-17 11:28, registrator wrote:

In this case you will help function SIP_HEADER(from)


Sent from: Lenovo P70-A

On Apr 14, 2017 12:04 PM, Jonas Kellens <jonas.kell...@telenet.be> wrote:

Hello


this does not set user field in From-header.

I get :

From: "user762" <sip:9876543...@myprovider.biz>;tag=as7f44c043

What I want is :

From: "9876543210" <sip:user...@myprovider.biz>;tag=as7f44c043


I need this part : <sip:user...@myprovider.biz>

you see the user part ? I need to set the value 'user762'




Kind regards

J.




On 14-04-17 10:46, registrator wrote:

Hello!



May be you help CALLERID(name) function?



exten => _X.,1,Set(CALLERID(name)=$name)



Then you well see INVITE

SIP : FROM "$name" .



Sent from: Lenovo P70-A



On Apr 14, 2017 10:54 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote:


Hello





any input on this ? How to set user-field in From-header with the 
Dial()-command in dialplan ?







Kind regards



J.





On 03-04-17 10:25, Jonas Kellens wrote:


Hello



how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?



None of the below Dial() command give the correct result :



exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})



The From part of the SIP INVITE always has the EXTEN in it in stead of the user 
(user762) :



From: "the_extension" <sip:the_extens...@my.ip.add.ress>;tag=as224453ac



How can I get :



From: "the_extension" <sip:user...@my.ip.add.ress>;tag=as224453ac



??







I know about sip.conf. That is not the question. My question is clear : how to 
set this in dialplan ?







Thank you for the feedback.





Kind regards.







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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

2017-04-14 Thread Jonas Kellens

Hello


this does not set user field in From-header.

I get :

From: "user762" <sip:9876543...@myprovider.biz>;tag=as7f44c043

What I want is :

From: "9876543210" <sip:user...@myprovider.biz>;tag=as7f44c043


I need this part : <sip:user...@myprovider.biz>

you see the user part ? I need to set the value 'user762'




Kind regards

J.




On 14-04-17 10:46, registrator wrote:

Hello!

May be you help CALLERID(name) function?

exten => _X.,1,Set(CALLERID(name)=$name)

Then you well see INVITE
SIP : FROM "$name" .

Sent from: Lenovo P70-A

On Apr 14, 2017 10:54 AM, Jonas Kellens<jonas.kell...@telenet.be>  wrote:

Hello


any input on this ? How to set user-field in From-header with the 
Dial()-command in dialplan ?



Kind regards

J.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?

None of the below Dial() command give the correct result :

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})

The From part of the SIP INVITE always has the EXTEN in it in stead of the user 
(user762) :

From: "the_extension"<sip:the_extens...@my.ip.add.ress>;tag=as224453ac

How can I get :

From: "the_extension"<sip:user...@my.ip.add.ress>;tag=as224453ac

??



I know about sip.conf. That is not the question. My question is clear : how to 
set this in dialplan ?



Thank you for the feedback.


Kind regards.




-- 
_
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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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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

2017-04-14 Thread Jonas Kellens

Hello


any input on this ? How to set user-field in From-header with the 
Dial()-command in dialplan ?




Kind regards

J.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?


None of the below Dial() command give the correct result :

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten => 
_XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => 
_XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})

The From part of the SIP INVITE always has the EXTEN in it in stead of 
the user (user762) :


From: "the_extension" <sip:the_extens...@my.ip.add.ress>;tag=as224453ac

How can I get :

From: "the_extension" <sip:user...@my.ip.add.ress>;tag=as224453ac

??



I know about sip.conf. That is not the question. My question is clear 
: how to set this in dialplan ?




Thank you for the feedback.


Kind regards.




-- 
_
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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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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

2017-04-06 Thread Jonas Kellens

Hello


in what way does this set the 'fromuser' field in the SIP INVITE ?



Kind regards.


J.

On 05-04-17 22:05, Pete Mundy wrote:

Hi Jonas

Does the information at this link help?

http://the-asterisk-book.com/1.6/funktionen-callerid.html

Pete


On 5/04/2017, at 8:11 pm, Jonas Kellens <jonas.kell...@telenet.be 
<mailto:jonas.kell...@telenet.be>> wrote:


Hello

anyone have some useful input on this ?



Thanks.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?





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

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Re: [asterisk-users] Define SIP fromuser field in Dial()-command

2017-04-05 Thread Jonas Kellens

Hello

anyone have some useful input on this ?



Thanks.


On 03-04-17 10:25, Jonas Kellens wrote:

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?


None of the below Dial() command give the correct result :

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten => 
_XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => 
_XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})

The From part of the SIP INVITE always has the EXTEN in it in stead of 
the user (user762) :


From: "the_extension" <sip:the_extens...@my.ip.add.ress>;tag=as224453ac

How can I get :

From: "the_extension" <sip:user...@my.ip.add.ress>;tag=as224453ac

??



I know about sip.conf. That is not the question. My question is clear 
: how to set this in dialplan ?




Thank you for the feedback.


Kind regards.




-- 
_
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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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[asterisk-users] Define SIP fromuser field in Dial()-command

2017-04-03 Thread Jonas Kellens

Hello

how can I set the fromuser field of the SIP INVITE when using the 
Dial()-command in the dialplan ?


None of the below Dial() command give the correct result :

exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz)
exten => 
_XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user...@myprovider.biz/${EXTEN})
exten => 
_XX.,n,Dial(SIP/user762:passwdk5j6::user...@myprovider.biz/${EXTEN})

exten => _XX.,n,Dial(SIP/user762:passwdk...@myprovider.biz/${EXTEN})

The From part of the SIP INVITE always has the EXTEN in it in stead of 
the user (user762) :


From: "the_extension" ;tag=as224453ac

How can I get :

From: "the_extension" ;tag=as224453ac

??



I know about sip.conf. That is not the question. My question is clear : 
how to set this in dialplan ?




Thank you for the feedback.


Kind regards.
-- 
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Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

2017-03-30 Thread Jonas Kellens

Hello

I can confirm that touch-ing /etc/asterisk/musiconhold.conf (just open 
with vi and close again) and then issuing a 'module reload 
res_musiconhold.so' on the Asterisk CLI makes the new files load into 
Asterisk.


Very strange !!

I would not know how to automate this through script...



Kind regards.


On 24-03-17 12:29, Daniel Journo wrote:


> Hello
> as you can read in my original post "moh reload" and "module reload 
res_musiconhold.so" does nothing.

> Only at restart the new files are available.
> Is this a bug ?? How can I get more debugging for this problem ??

I think there is currently a bug with MOH. For now, if you add a file 
to a moh folder, ‘touch musiconhold.conf’ and then reload moh.


Please let me know how it goes.

Kind regards

Dan Journo





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Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

2017-03-24 Thread Jonas Kellens

Hello

as you can read in my original post "moh reload" and "module reload 
res_musiconhold.so" does nothing.


Only at restart the new files are available.

Is this a bug ?? How can I get more debugging for this problem ??



Kind regards.


On 23-03-17 22:54, Administrator TOOTAI wrote:

Le 23/03/2017 à 20:17, Jonas Kellens a écrit :

Hello


is there any more information on how to reload/read musiconhold files ?


CLI> module reload res_musiconhold

--
Daniel


On 07-03-17 10:46, Jonas Kellens wrote:

Hello

I did not mention it but of course the MOH directory is listed in
/etc/asterisk/musiconhold.conf :

[default]
mode=files
directory=/var/lib/asterisk/moh

[myfolder_1]
mode=files
directory=/var/lib/asterisk/moh/myfolder/1
sort=alpha

[myfolder_2]
mode=files
directory=/var/lib/asterisk/moh/myfolder/2
sort=alpha

[myfolder_3]
mode=files
directory=/var/lib/asterisk/moh/myfolder/3
sort=alpha


No mather where I put the new file, it is never listed.

Untill a full restart of Asterisk ! Then it is listed. But is there no
other way to load/read a new MOH file than to completely restart
Asterisk ??


After Asterisk restart :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/macroform-the_simplicity
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
Class: myfolder_1
File: /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity



Kind regards.




On 03-03-17 18:26, John Kiniston wrote:

Your new file is in the 'myfolder/1'' subdirectory of the MOH
directory.

Either move the file into the MOH directory or define a new class in
musiconhold.conf that is for your directory.


On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:

Hello

using Asterisk 1.8.32.3

Current music on hold :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity

New musiconhold file :

[root@myserver ]# file
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz

I issue a reload of the moh :

myserver*CLI> moh reload
myserver*CLI> module reload res_musiconhold.so
[Mar  3 15:04:53] -- Reloading module 'res_musiconhold.so'
(Music On Hold Resource)

Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity


Even a complete 'module reload' on Asterisk CLI does nothing :

myserver*CLI> module reload
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/extconfig.conf':
[Mar  3 15:13:54]   == Found
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/logger.conf': [Mar
3 15:13:54]   == Found
...
[Mar  3 15:13:54] -- Reloading module 'res_musiconhold.so'
(Music On Hold Resource)
...


Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity



So, reloading musiconhold does not reload/read musiconhold files.


How to read/load new musiconhold files into asterisk ??


Kind regards.



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Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

2017-03-23 Thread Jonas Kellens

Hello


is there any more information on how to reload/read musiconhold files ?


Kind regards.



On 07-03-17 10:46, Jonas Kellens wrote:

Hello

I did not mention it but of course the MOH directory is listed in 
/etc/asterisk/musiconhold.conf :


[default]
mode=files
directory=/var/lib/asterisk/moh

[myfolder_1]
mode=files
directory=/var/lib/asterisk/moh/myfolder/1
sort=alpha

[myfolder_2]
mode=files
directory=/var/lib/asterisk/moh/myfolder/2
sort=alpha

[myfolder_3]
mode=files
directory=/var/lib/asterisk/moh/myfolder/3
sort=alpha


No mather where I put the new file, it is never listed.

Untill a full restart of Asterisk ! Then it is listed. But is there no 
other way to load/read a new MOH file than to completely restart 
Asterisk ??



After Asterisk restart :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/macroform-the_simplicity
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
Class: myfolder_1
File: /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity



Kind regards.




On 03-03-17 18:26, John Kiniston wrote:

Your new file is in the 'myfolder/1'' subdirectory of the MOH directory.

Either move the file into the MOH directory or define a new class in 
musiconhold.conf that is for your directory.



On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


Hello

using Asterisk 1.8.32.3

Current music on hold :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity

New musiconhold file :

[root@myserver ]# file
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz

I issue a reload of the moh :

myserver*CLI> moh reload
myserver*CLI> module reload res_musiconhold.so
[Mar  3 15:04:53] -- Reloading module 'res_musiconhold.so'
(Music On Hold Resource)

Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity


Even a complete 'module reload' on Asterisk CLI does nothing :

myserver*CLI> module reload
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/extconfig.conf':
[Mar  3 15:13:54]   == Found
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/logger.conf': [Mar 
3 15:13:54]   == Found

...
[Mar  3 15:13:54] -- Reloading module 'res_musiconhold.so'
(Music On Hold Resource)
...


Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity



So, reloading musiconhold does not reload/read musiconhold files.


How to read/load new musiconhold files into asterisk ??


Kind regards.



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butcher a hog, conn a ship, design a building, write a sonnet, 
balance accounts, build a wall, set a bone, comfort the dying, take 
orders, give orders, cooperate, act alone, solve equations, analyze a 
new problem, pitch manure, program a computer, cook a tasty meal, 
fight efficiently, die gallantly. Specialization is for insects.

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Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

2017-03-07 Thread Jonas Kellens

Hello

I did not mention it but of course the MOH directory is listed in 
/etc/asterisk/musiconhold.conf :


[default]
mode=files
directory=/var/lib/asterisk/moh

[myfolder_1]
mode=files
directory=/var/lib/asterisk/moh/myfolder/1
sort=alpha

[myfolder_2]
mode=files
directory=/var/lib/asterisk/moh/myfolder/2
sort=alpha

[myfolder_3]
mode=files
directory=/var/lib/asterisk/moh/myfolder/3
sort=alpha


No mather where I put the new file, it is never listed.

Untill a full restart of Asterisk ! Then it is listed. But is there no 
other way to load/read a new MOH file than to completely restart Asterisk ??



After Asterisk restart :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/macroform-the_simplicity
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
Class: myfolder_1
File: /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity



Kind regards.




On 03-03-17 18:26, John Kiniston wrote:

Your new file is in the 'myfolder/1'' subdirectory of the MOH directory.

Either move the file into the MOH directory or define a new class in 
musiconhold.conf that is for your directory.



On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


Hello

using Asterisk 1.8.32.3

Current music on hold :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity

New musiconhold file :

[root@myserver ]# file
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono
8000 Hz

I issue a reload of the moh :

myserver*CLI> moh reload
myserver*CLI> module reload res_musiconhold.so
[Mar  3 15:04:53] -- Reloading module 'res_musiconhold.so'
(Music On Hold Resource)

Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity


Even a complete 'module reload' on Asterisk CLI does nothing :

myserver*CLI> module reload
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/extconfig.conf':
[Mar  3 15:13:54] == Found
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/logger.conf': [Mar 
3 15:13:54]   == Found

...
[Mar  3 15:13:54] -- Reloading module 'res_musiconhold.so'
(Music On Hold Resource)
...


Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity



So, reloading musiconhold does not reload/read musiconhold files.


How to read/load new musiconhold files into asterisk ??


Kind regards.



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

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butcher a hog, conn a ship, design a building, write a sonnet, balance 
accounts, build a wall, set a bone, comfort the dying, take orders, 
give orders, cooperate, act alone, solve equations, analyze a new 
problem, pitch manure, program a computer, cook a tasty meal, fight 
efficiently, die gallantly. Specialization is for insects.

---Heinlein




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New to Asterisk? Start her

[asterisk-users] moh reload not reloading/reading new musiconhold files

2017-03-03 Thread Jonas Kellens

Hello

using Asterisk 1.8.32.3

Current music on hold :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity

New musiconhold file :

[root@myserver ]# file 
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav: RIFF 
(little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz


I issue a reload of the moh :

myserver*CLI> moh reload
myserver*CLI> module reload res_musiconhold.so
[Mar  3 15:04:53] -- Reloading module 'res_musiconhold.so' (Music On 
Hold Resource)


Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity


Even a complete 'module reload' on Asterisk CLI does nothing :

myserver*CLI> module reload
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/extconfig.conf': [Mar  3 
15:13:54]   == Found
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/logger.conf': [Mar 3 
15:13:54]   == Found

...
[Mar  3 15:13:54] -- Reloading module 'res_musiconhold.so' (Music On 
Hold Resource)

...


Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity



So, reloading musiconhold does not reload/read musiconhold files.


How to read/load new musiconhold files into asterisk ??


Kind regards.


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Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

2016-11-22 Thread Jonas Kellens

On 21-11-16 17:20, Matthew Jordan wrote:



On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


On 21-11-16 15:17, Matthew Jordan wrote:


On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:

Hello

when using Asterisk version 13.12.2 I notice that it takes up
to 30 seconds (sometimes even longer) for a call queue to
call its members.

Example 1 :

[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack
[Nov 21 08:17:57] res_musiconhold.c: Started music on hold,
class 'default', on channel 'SIP/incoming-0246'

[Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack
[Nov 21 08:18:26] app_queue.c: Called
Local/mysip692@CallFromQueue
[Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-003c;2",
"SIP/mysip692") in new stack
[Nov 21 08:18:26] app_dial.c: Called SIP/mysip692


Example 2 :

[Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack
[Nov 21 08:20:11] res_musiconhold.c: Started music on hold,
class 'default', on channel 'SIP/incoming-0255'

[Nov 21 08:20:45] app_queue.c: Called
Local/mysip692@CallFromQueue
[Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack
[Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-0040;2",
"SIP/mysip692") in new stack
[Nov 21 08:20:45] app_dial.c: Called SIP/mysip692


I did not see this behaviour in previous Asterisk versions.

Could this be a bug ?


There's not enough information here to know what is preventing
the call from occurring.

I'd look at a debug log between the caller entering the Queue and
the outbound call being made. That should illustrate what is
causing the delay.

-- 
Matthew Jordan



Hello


and what exactly am I looking for in the debug logs ?

I have generated debug output and re-produced the issue.


Again 23 seconds before calling the queue member :

[Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack
[Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class
'default', on channel 'SIP/incoming-4e6e'

[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack
[Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2]
NoOp("Local/mysip692@CallFromQueue-081a;2", "exten =
mysip692") in new stack
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692") in
new stack
[Nov 21 16:23:56] app_dial.c: Called SIP/mysip692
[Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing
[Nov 21 16:23:56] app_queue.c:
Local/mysip692@CallFromQueue-081a;1 is ringing



Could it be that it is because my Queue member 'mysip692' is
occupied in another bridge (call) ?

This I see in the logs just before the Call Queue starts calling
the queue member :

[Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged
'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63
left 'native_rtp' basic-bridge 
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a
left 'native_rtp' basic-bridge 


A bit too coincidal, no ?

So then it has something to do with the bridging ?



I did not have this behaviour in previous Asterisk versions.


Those aren't debug logs. Instructions for generating debug information 
can be found on the wiki:


https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

That being said, if the Queue Member is currently busy (which will be 
denoted by their device state), and you have not configured the Queue 
to ring the Queue Member while they are busy, then I would expect any 
new caller to hang out in the Queue until that Member is available.


--
Matthew Jordan


Hello


I did a call with my queue member mysip517 and no

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

2016-11-22 Thread Jonas Kellens

On 21-11-16 19:14, Jonas Kellens wrote:


On 21-11-16 17:20, Matthew Jordan wrote:



On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


On 21-11-16 15:17, Matthew Jordan wrote:


On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:

Hello

when using Asterisk version 13.12.2 I notice that it takes
up to 30 seconds (sometimes even longer) for a call queue to
call its members.

Example 1 :

[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0246", "myqueue1300,,,") in new
stack
[Nov 21 08:17:57] res_musiconhold.c: Started music on hold,
class 'default', on channel 'SIP/incoming-0246'

[Nov 21 08:18:26] pbx.c: Executing
[mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack
[Nov 21 08:18:26] app_queue.c: Called
Local/mysip692@CallFromQueue
[Nov 21 08:18:26] pbx.c: Executing
[mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-003c;2",
"SIP/mysip692") in new stack
[Nov 21 08:18:26] app_dial.c: Called SIP/mysip692


Example 2 :

[Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0255", "myqueue1300,,,") in new
stack
[Nov 21 08:20:11] res_musiconhold.c: Started music on hold,
class 'default', on channel 'SIP/incoming-0255'

[Nov 21 08:20:45] app_queue.c: Called
Local/mysip692@CallFromQueue
[Nov 21 08:20:45] pbx.c: Executing
[mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack
[Nov 21 08:20:45] pbx.c: Executing
[mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-0040;2",
"SIP/mysip692") in new stack
[Nov 21 08:20:45] app_dial.c: Called SIP/mysip692


I did not see this behaviour in previous Asterisk versions.

Could this be a bug ?


There's not enough information here to know what is preventing
the call from occurring.

I'd look at a debug log between the caller entering the Queue
and the outbound call being made. That should illustrate what is
causing the delay.

-- 
Matthew Jordan



Hello


and what exactly am I looking for in the debug logs ?

I have generated debug output and re-produced the issue.


Again 23 seconds before calling the queue member :

[Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack
[Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class
'default', on channel 'SIP/incoming-4e6e'

[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack
[Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2]
NoOp("Local/mysip692@CallFromQueue-081a;2", "exten =
mysip692") in new stack
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692")
in new stack
[Nov 21 16:23:56] app_dial.c: Called SIP/mysip692
[Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing
[Nov 21 16:23:56] app_queue.c:
Local/mysip692@CallFromQueue-081a;1 is ringing



Could it be that it is because my Queue member 'mysip692' is
occupied in another bridge (call) ?

This I see in the logs just before the Call Queue starts calling
the queue member :

[Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged
'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63
left 'native_rtp' basic-bridge 
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a
left 'native_rtp' basic-bridge 


A bit too coincidal, no ?

So then it has something to do with the bridging ?



I did not have this behaviour in previous Asterisk versions.


Those aren't debug logs. Instructions for generating debug 
information can be found on the wiki:


https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

That being said, if the Queue Member is currently busy (which will be 
denoted by their device state), and you have not configured the Queue 
to ring the Queue Member while they are busy, then I would expect any 
new caller to hang out in the Queue until that Member is

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

2016-11-21 Thread Jonas Kellens


On 21-11-16 17:20, Matthew Jordan wrote:



On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


On 21-11-16 15:17, Matthew Jordan wrote:


On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:

Hello

when using Asterisk version 13.12.2 I notice that it takes up
to 30 seconds (sometimes even longer) for a call queue to
call its members.

Example 1 :

[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack
[Nov 21 08:17:57] res_musiconhold.c: Started music on hold,
class 'default', on channel 'SIP/incoming-0246'

[Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack
[Nov 21 08:18:26] app_queue.c: Called
Local/mysip692@CallFromQueue
[Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-003c;2",
"SIP/mysip692") in new stack
[Nov 21 08:18:26] app_dial.c: Called SIP/mysip692


Example 2 :

[Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack
[Nov 21 08:20:11] res_musiconhold.c: Started music on hold,
class 'default', on channel 'SIP/incoming-0255'

[Nov 21 08:20:45] app_queue.c: Called
Local/mysip692@CallFromQueue
[Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack
[Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-0040;2",
"SIP/mysip692") in new stack
[Nov 21 08:20:45] app_dial.c: Called SIP/mysip692


I did not see this behaviour in previous Asterisk versions.

Could this be a bug ?


There's not enough information here to know what is preventing
the call from occurring.

I'd look at a debug log between the caller entering the Queue and
the outbound call being made. That should illustrate what is
causing the delay.

-- 
Matthew Jordan



Hello


and what exactly am I looking for in the debug logs ?

I have generated debug output and re-produced the issue.


Again 23 seconds before calling the queue member :

[Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack
[Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class
'default', on channel 'SIP/incoming-4e6e'

[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack
[Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2]
NoOp("Local/mysip692@CallFromQueue-081a;2", "exten =
mysip692") in new stack
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692") in
new stack
[Nov 21 16:23:56] app_dial.c: Called SIP/mysip692
[Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing
[Nov 21 16:23:56] app_queue.c:
Local/mysip692@CallFromQueue-081a;1 is ringing



Could it be that it is because my Queue member 'mysip692' is
occupied in another bridge (call) ?

This I see in the logs just before the Call Queue starts calling
the queue member :

[Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged
'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63
left 'native_rtp' basic-bridge 
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a
left 'native_rtp' basic-bridge 


A bit too coincidal, no ?

So then it has something to do with the bridging ?



I did not have this behaviour in previous Asterisk versions.


Those aren't debug logs. Instructions for generating debug information 
can be found on the wiki:


https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

That being said, if the Queue Member is currently busy (which will be 
denoted by their device state), and you have not configured the Queue 
to ring the Queue Member while they are busy, then I would expect any 
new caller to hang out in the Queue until that Member is available.


--
Matthew Jordan


Hello

indeed no debug log output. Therefore I need to 

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

2016-11-21 Thread Jonas Kellens

On 21-11-16 15:17, Matthew Jordan wrote:


On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


Hello

when using Asterisk version 13.12.2 I notice that it takes up to
30 seconds (sometimes even longer) for a call queue to call its
members.

Example 1 :

[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack
[Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class
'default', on channel 'SIP/incoming-0246'

[Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack
[Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue
[Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in
new stack
[Nov 21 08:18:26] app_dial.c: Called SIP/mysip692


Example 2 :

[Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15]
Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack
[Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class
'default', on channel 'SIP/incoming-0255'

[Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue
[Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1]
NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack
[Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3]
Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in
new stack
[Nov 21 08:20:45] app_dial.c: Called SIP/mysip692


I did not see this behaviour in previous Asterisk versions.

Could this be a bug ?


There's not enough information here to know what is preventing the 
call from occurring.


I'd look at a debug log between the caller entering the Queue and the 
outbound call being made. That should illustrate what is causing the 
delay.


--
Matthew Jordan



Hello


and what exactly am I looking for in the debug logs ?

I have generated debug output and re-produced the issue.


Again 23 seconds before calling the queue member :

[Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15] 
Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack
[Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class 
'default', on channel 'SIP/incoming-4e6e'


[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1] 
NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack

[Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2] 
NoOp("Local/mysip692@CallFromQueue-081a;2", "exten = mysip692") in 
new stack
[Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3] 
Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692") in new stack

[Nov 21 16:23:56] app_dial.c: Called SIP/mysip692
[Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing
[Nov 21 16:23:56] app_queue.c: Local/mysip692@CallFromQueue-081a;1 
is ringing




Could it be that it is because my Queue member 'mysip692' is occupied in 
another bridge (call) ?


This I see in the logs just before the Call Queue starts calling the 
queue member :


[Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged 
'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63 left 
'native_rtp' basic-bridge 
[Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a left 
'native_rtp' basic-bridge 



A bit too coincidal, no ?

So then it has something to do with the bridging ?



I did not have this behaviour in previous Asterisk versions.


Kind regards.

J.

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[asterisk-users] Asterisk 13.12.2 : strange queue behaviour

2016-11-21 Thread Jonas Kellens

Hello

when using Asterisk version 13.12.2 I notice that it takes up to 30 
seconds (sometimes even longer) for a call queue to call its members.


Example 1 :

[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] 
Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack
[Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 
'default', on channel 'SIP/incoming-0246'


[Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1] 
NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack

[Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue
[Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3] 
Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in new stack

[Nov 21 08:18:26] app_dial.c: Called SIP/mysip692


Example 2 :

[Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15] 
Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack
[Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class 
'default', on channel 'SIP/incoming-0255'


[Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue
[Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1] 
NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack
[Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3] 
Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in new stack

[Nov 21 08:20:45] app_dial.c: Called SIP/mysip692


I did not see this behaviour in previous Asterisk versions.

Could this be a bug ?





Kind regards.

Jonas.
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Re: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'

2016-11-12 Thread Jonas Kellens


On 11-10-16 14:44, Joshua Colp wrote:

Jonas Kellens wrote:

Hello

I am experiencing a freeze of the Asterisk proces when issuing a 'sip
reload'.

I have this issue every time on asterisk versions : 13.11.2, 13.11.1,
13.10.0 and certified-13.8-cert3.

I do not have this on versions certified-13.8-cert2,
certified-13.8-cert1 and asterisk 1.8.32.3.

The only solution is a cold restart of Asterisk.

I can execute any command on CLI except 'sip reload'.


This doesn't ring a bell on any issues filed or any posts anywhere. 
Getting a backtrace[1] would show precisely where it is hanging 
though. Is it possible a host is having DNS issues? That can cause 
chan_sip to lock up for a period of time.


[1] 
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock




Hello

backtrace was generated and issue is reported : 
https://issues.asterisk.org/jira/browse/ASTERISK-26585




Kind regards.

J.


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Re: [asterisk-users] Realtime queue & agent groups

2016-11-02 Thread Jonas Kellens

Hello

any one have some input on this ?

I've already tried changing the membername to : testacc77000/@1
Is completely ignored.

I've already tried changing the interface to : testacc77000/@1
Is completely ignored.



Or is it just not possible to group queue members ??



Thanks.

J.


On 27-10-16 15:53, Jonas Kellens wrote:

Hello

I'm a bit confused on how to group agents (give agents a group number) 
when using realtime queues.


I read on the wiki :

  * If you include groups in your queue definition the calls get
routed in the order of the group regardless of the specified
strategy. So I just have a member= line for each agent.

member => Agent/@1 ; a group
member => Agent/501 ; a single agent
member => Agent/:1,1 ; Any agent in group 1, wait for first available, 
but consider with penalty



In my realtime database I have table queue_members :

+--++-++-+-++
| uniqueid | membername | queue_name  | 
interface  | state_interface | penalty 
| paused |

+--++-++-+-++
| 2916 | testacc77000   | queue7700q4 | testacc77000 
| |   0 |   NULL |
| 2917 | testacc77001   | queue7700q4 | testacc77001 
| |   3 |   NULL |
| 2843 | testacc77000   | queue7700q4 | testacc77000 
| |   0 |   NULL |
| 2905 | testacc7700905 | queue7700q5 | testacc7700905 
| |   0 |   NULL |
| 2888 | testacc77000   | queue7700q5 | testacc77000 
| |   0 |   NULL |
| 2900 | testacc77000   | queue7700q5 | testacc77000 
| |   0 |   NULL |
| 2901 | testacc77001   | queue7700q5 | testacc77001 
| |   0 |   NULL |




How do I define a group to a certain agent/member in this case ?





Kind regards

J.






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Re: [asterisk-users] Problem setting up ssl connection

2016-10-28 Thread Jonas Kellens

On 26-10-16 23:24, Stefan Tichy wrote:

On Wed, Oct 26, 2016 at 04:57:15PM +0200, Jonas Kellens wrote:

if it is indeed manager.conf that I need to edit then the problem is
that I see no param : tlsdontverifyserver=yes

A comment copied from sip.conf.sample:
"If set to yes, don't verify the servers certificate when acting as a client."

With AMI connections asterisk is allways the server.



I don't know how to make the AMI ignore the self-signed certificate.

The client fails to verfify the certificate. Do you use PHP 5.6? The
default behavior has changed.


Hello


I use PHP 5.6.27.

So I should be looking inside php.ini ?




Kind regards

J.


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[asterisk-users] Realtime queue & agent groups

2016-10-27 Thread Jonas Kellens

Hello

I'm a bit confused on how to group agents (give agents a group number) 
when using realtime queues.


I read on the wiki :

 * If you include groups in your queue definition the calls get routed
   in the order of the group regardless of the specified strategy. So I
   just have a member= line for each agent.

member => Agent/@1 ; a group
member => Agent/501 ; a single agent
member => Agent/:1,1 ; Any agent in group 1, wait for first available, 
but consider with penalty



In my realtime database I have table queue_members :

+--++-++-+-++
| uniqueid | membername | queue_name  | 
interface  | state_interface | penalty | 
paused |

+--++-++-+-++
| 2916 | testacc77000   | queue7700q4 | testacc77000 
| |   0 |   NULL |
| 2917 | testacc77001   | queue7700q4 | testacc77001 
| |   3 |   NULL |
| 2843 | testacc77000   | queue7700q4 | testacc77000 
| |   0 |   NULL |
| 2905 | testacc7700905 | queue7700q5 | testacc7700905 
| |   0 |   NULL |
| 2888 | testacc77000   | queue7700q5 | testacc77000 
| |   0 |   NULL |
| 2900 | testacc77000   | queue7700q5 | testacc77000 
| |   0 |   NULL |
| 2901 | testacc77001   | queue7700q5 | testacc77001 
| |   0 |   NULL |




How do I define a group to a certain agent/member in this case ?





Kind regards

J.


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Re: [asterisk-users] Problem setting up ssl connection

2016-10-26 Thread Jonas Kellens


On 26-10-16 15:03, Dan Jenkins wrote:



On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


Hello


I keep getting the following error when trying to connect to the
Asterisk server using AMI :

$socket = fsockopen("tls://11.22.33.44
<http://11.22.33.44>","5039", $errno, $errstr, 5);

Erorr on CLI :

[Oct 26 14:38:19] ERROR[2992]: tcptls.c:609
handle_tcptls_connection: Problem setting up ssl connection:
error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
[Oct 26 14:38:19] WARNING[2992]: tcptls.c:684
handle_tcptls_connection: FILE * open failed!

I have in sip.conf :

tlsenable=yes
tlsbindaddr=0.0.0.0

tlscertfile=/etc/asterisk/keys/asterisk.pem
tlsdontverifyserver=yes
tlscipher=ALL
;tlsclientmethod=tlsv2

/etc/asterisk/keys :

-rw--- 1 root root 1,2K okt 26 14:25 asterisk.crt
-rw--- 1 root root  574 okt 26 14:24 asterisk.csr
-rw--- 1 root root  887 okt 26 14:24 asterisk.key
-rw--- 1 root root 2,1K okt 26 14:25 asterisk.pem
-rw--- 1 root root  160 okt 26 14:24 ca.cfg
-rw--- 1 root root 1,8K okt 26 14:24 ca.crt
-rw--- 1 root root 3,3K okt 26 14:24 ca.key
-rw--- 1 root root  123 okt 26 14:24 tmp.cfg


The webserver ( A ) from where I open the socket to
tls://11.22.33.44 <http://11.22.33.44> also has a self-signed
certificate.

This problem started when creating a new self-signed cert on
webserver A.




Any thoughts ?


Thanks !


Kind regards.


J.

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Jonas,

You talk about sip.conf and setting your TLS cert there - but you're 
trying to connect to the AMI over TLS - so you need to set this stuff 
in manager.conf 
(https://github.com/asterisk/asterisk/blob/master/configs/samples/manager.conf.sample) 
- did you mean manager.conf ?


The error says that it doesn't understand the Certificate Authority in 
the cert. The box you're connecting from shouldn't affect anything so 
the issue will be with the CA of the cert - usually you need to add 
the CA to the cert to complete the chain.


If this is a public box then I'd recommend just using LetsEncrypt - 
many things don't like Self Signed Certs now


Dan



Hello Dan

if it is indeed manager.conf that I need to edit then the problem is 
that I see no param : tlsdontverifyserver=yes


I don't know how to make the AMI ignore the self-signed certificate.




Kind regards

J.

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[asterisk-users] Problem setting up ssl connection

2016-10-26 Thread Jonas Kellens

Hello


I keep getting the following error when trying to connect to the 
Asterisk server using AMI :


$socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5);

Erorr on CLI :

[Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: 
Problem setting up ssl connection: error:14094418:SSL 
routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
[Oct 26 14:38:19] WARNING[2992]: tcptls.c:684 handle_tcptls_connection: 
FILE * open failed!


I have in sip.conf :

tlsenable=yes
tlsbindaddr=0.0.0.0

tlscertfile=/etc/asterisk/keys/asterisk.pem
tlsdontverifyserver=yes
tlscipher=ALL
;tlsclientmethod=tlsv2

/etc/asterisk/keys :

-rw--- 1 root root 1,2K okt 26 14:25 asterisk.crt
-rw--- 1 root root  574 okt 26 14:24 asterisk.csr
-rw--- 1 root root  887 okt 26 14:24 asterisk.key
-rw--- 1 root root 2,1K okt 26 14:25 asterisk.pem
-rw--- 1 root root  160 okt 26 14:24 ca.cfg
-rw--- 1 root root 1,8K okt 26 14:24 ca.crt
-rw--- 1 root root 3,3K okt 26 14:24 ca.key
-rw--- 1 root root  123 okt 26 14:24 tmp.cfg


The webserver ( A ) from where I open the socket to tls://11.22.33.44 
also has a self-signed certificate.


This problem started when creating a new self-signed cert on webserver A.




Any thoughts ?


Thanks !


Kind regards.


J.
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Re: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'

2016-10-12 Thread Jonas Kellens

Hello Edilson


I have removed all sip peer definition in sip.conf.

I only have rtcachefriends=yes for my mysql realtime sip peers (IP-phones).

But what I don't understand is why I have no issue when for example 
using asterisk 1.8.32.3 again.





Kind regards.



On 11-10-16 14:50, Edilson Amaral wrote:

Hi

This happens to me when one peer (provider) is bad !
Try to remove all peers from your sip.conf and gradually add them back!







*From:* Jonas Kellens <jonas.kell...@telenet.be>
*To:* Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>

*Sent:* Tuesday, October 11, 2016 8:41 AM
*Subject:* [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and 
certified-13.8-cert3 : freeze on 'sip reload'


Hello

I am experiencing a freeze of the Asterisk proces when issuing a 'sip 
reload'.


I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 
13.10.0 and certified-13.8-cert3.


I do not have this on versions certified-13.8-cert2, 
certified-13.8-cert1 and asterisk 1.8.32.3.


The only solution is a cold restart of Asterisk.

I can execute any command on CLI except 'sip reload'.

This is what I have on CLI :

sip5*CLI> sip reload
[Oct  7 23:58:40]  Reloading SIP
[Oct  7 23:58:40]   == Parsing '/etc/asterisk/sip.conf': Found
[Oct  7 23:58:40]   == Parsing '/etc/asterisk/sipTemplates.conf': Found
[Oct  7 23:58:40]   == Parsing '/etc/asterisk/users.conf': Found
[Oct  7 23:58:40]   == Using SIP TOS bits 96
[Oct  7 23:58:40]   == Using SIP CoS mark 3
[Oct  7 23:58:40]   == TLS/SSL ECDH initialized (secp256r1), faster 
PFS cipher-suites enabled

[Oct  7 23:58:40]   == TLS/SSL certificate ok

--> no more output on CLI. Asterisk has gone completely !

Another 'sip reload' gives :

sip5*CLI> sip reload
[Oct  8 00:01:10] Previous SIP reload not yet done

sip5*CLI> sip reload
sip5*CLI>


Other commands are no problem on the CLI (while the freeze occurs ! ) :

sip5*CLI> core show  version
Asterisk certified/13.8-cert3 built by root @ sip5.mydomain.tld on a 
x86_64 running Linux on 2016-10-07 21:27:15 UTC



sip5*CLI> sip show channelstats
Peer Call ID  Duration Recv: Pack Lost   ( %) 
Jitter Send: Pack  Lost ( %) Jitter

0 active SIP channels


sip5*CLI> core show threads
0x7f97ff0fb700 2849 netconsole   started at [ 1639] asterisk.c 
listener()
0x7f97fe843700 2760 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97ff367700 2759 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97fe8bf700 2758 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97fe93b700 2173 monitor_sig_flagsstarted at [ 4768] asterisk.c 
asterisk_daemon()
0x7f97fe9b7700 2172 default_tps_processing_function started at [  200] 
taskprocessor.c default_listener_start()
0x7f97fea33700 2171 default_tps_processing_function started at [  200] 
taskprocessor.c default_listener_start()
0x7f97feaaf700 2170 default_tps_processing_function started at [  200] 
taskprocessor.c default_listener_start()
0x7f97feb2b700 2169 scan_thread  started at [  920] 
pbx_spool.c load_module()
0x7f97feba7700 2167 cleanup  started at [  400] 
pbx_realtime.c load_module()
0x7f97fec23700 2165 lock_broker  started at [  524] 
func_lock.c load_module()
0x7f97fee13700 2161 cal->tech->load_calendar started at [  489] 
res_calendar.c build_calendar()
0x7f97fec9f700 2164 default_tps_processing_function started at [  200] 
taskprocessor.c default_listener_start()
0x7f97fed1b700 2163 cal->tech->load_calendar started at [  489] 
res_calendar.c build_calendar()
0x7f97fed97700 2162 cal->tech->load_calendar started at [  489] 
res_calendar.c build_calendar()
0x7f97fee8f700 2160 default_tps_processing_function started at [  200] 
taskprocessor.c default_listener_start()
0x7f97fef0b700 2159 default_tps_processing_function started at [  200] 
taskprocessor.c default_listener_start()
0x7f97fef87700 2158 default_tps_processing_function started at [  200] 
taskprocessor.c default_listener_start()
0x7f97ff003700 2157 do_monitor   started at [11645] 
chan_dahdi.c restart_monitor()
0x7f97ff07f700 2156 do_monitor   started at [29518] chan_sip.c 
restart_monitor()
0x7f97ff1f3700 2153 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97ff2eb700 2151 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97ff26f700 2152 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97ff3e3700 2149 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97ff45f700 2148 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97ff4db700 2147 worker_start started at [ 1077] 
threadpool.c worker_thread_start()
0x7f97ff5d3700 2145

[asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'

2016-10-11 Thread Jonas Kellens

Hello

I am experiencing a freeze of the Asterisk proces when issuing a 'sip 
reload'.


I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 
13.10.0 and certified-13.8-cert3.


I do not have this on versions certified-13.8-cert2, 
certified-13.8-cert1 and asterisk 1.8.32.3.


The only solution is a cold restart of Asterisk.

I can execute any command on CLI except 'sip reload'.

This is what I have on CLI :

sip5*CLI> sip reload
[Oct  7 23:58:40]  Reloading SIP
[Oct  7 23:58:40]   == Parsing '/etc/asterisk/sip.conf': Found
[Oct  7 23:58:40]   == Parsing '/etc/asterisk/sipTemplates.conf': Found
[Oct  7 23:58:40]   == Parsing '/etc/asterisk/users.conf': Found
[Oct  7 23:58:40]   == Using SIP TOS bits 96
[Oct  7 23:58:40]   == Using SIP CoS mark 3
[Oct  7 23:58:40]   == TLS/SSL ECDH initialized (secp256r1), faster PFS 
cipher-suites enabled

[Oct  7 23:58:40]   == TLS/SSL certificate ok

--> no more output on CLI. Asterisk has gone completely !

Another 'sip reload' gives :

sip5*CLI> sip reload
[Oct  8 00:01:10] Previous SIP reload not yet done

sip5*CLI> sip reload
sip5*CLI>


Other commands are no problem on the CLI (while the freeze occurs ! ) :

sip5*CLI> core show  version
Asterisk certified/13.8-cert3 built by root @ sip5.mydomain.tld on a 
x86_64 running Linux on 2016-10-07 21:27:15 UTC



sip5*CLI> sip show channelstats
Peer Call ID  Duration Recv: Pack  Lost ( %) Jitter 
Send: Pack  Lost   ( %) Jitter

0 active SIP channels


sip5*CLI> core show threads
0x7f97ff0fb700 2849 netconsole   started at [ 1639] asterisk.c 
listener()
0x7f97fe843700 2760 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff367700 2759 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97fe8bf700 2758 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97fe93b700 2173 monitor_sig_flagsstarted at [ 4768] asterisk.c 
asterisk_daemon()
0x7f97fe9b7700 2172 default_tps_processing_function started at [ 200] 
taskprocessor.c default_listener_start()
0x7f97fea33700 2171 default_tps_processing_function started at [ 200] 
taskprocessor.c default_listener_start()
0x7f97feaaf700 2170 default_tps_processing_function started at [ 200] 
taskprocessor.c default_listener_start()
0x7f97feb2b700 2169 scan_thread  started at [  920] pbx_spool.c 
load_module()
0x7f97feba7700 2167 cleanup  started at [  400] 
pbx_realtime.c load_module()
0x7f97fec23700 2165 lock_broker  started at [  524] func_lock.c 
load_module()
0x7f97fee13700 2161 cal->tech->load_calendar started at [ 489] 
res_calendar.c build_calendar()
0x7f97fec9f700 2164 default_tps_processing_function started at [ 200] 
taskprocessor.c default_listener_start()
0x7f97fed1b700 2163 cal->tech->load_calendar started at [ 489] 
res_calendar.c build_calendar()
0x7f97fed97700 2162 cal->tech->load_calendar started at [ 489] 
res_calendar.c build_calendar()
0x7f97fee8f700 2160 default_tps_processing_function started at [ 200] 
taskprocessor.c default_listener_start()
0x7f97fef0b700 2159 default_tps_processing_function started at [ 200] 
taskprocessor.c default_listener_start()
0x7f97fef87700 2158 default_tps_processing_function started at [ 200] 
taskprocessor.c default_listener_start()
0x7f97ff003700 2157 do_monitor   started at [11645] chan_dahdi.c 
restart_monitor()
0x7f97ff07f700 2156 do_monitor   started at [29518] chan_sip.c 
restart_monitor()
0x7f97ff1f3700 2153 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff2eb700 2151 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff26f700 2152 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff3e3700 2149 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff45f700 2148 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff4db700 2147 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff5d3700 2145 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff557700 2146 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff64f700 2144 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff6cb700 2143 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff747700 2142 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff7c3700 2141 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff83f700 2140 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff8bb700 2139 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff937700 2138 worker_start started at [ 1077] threadpool.c 
worker_thread_start()
0x7f97ff9b3700 2137 worker_start 

Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

2016-09-22 Thread Jonas Kellens

On 02-09-16 11:51, Administrator TOOTAI wrote:

Le 02/09/2016 à 11:26, Jonas Kellens a écrit :

Hello

when setting a local forward (in this case to extension 23) on a SIP
phone, I see the following on the Asterisk CLI :


[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back
from 11.22.33.44:40670
[Aug 31 14:59:34] -- Now forwarding
Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal'
(thanks to SIP/myaccount184-3729)


Question : how can I read the variable which contains the value
'myaccount184' in the context from-internal ?


From SIP_HEADER(TO) ?

[...]



Hello


SIP_HEADER(TO) is empty. So is SIP_HEADER(FROM).


My dialplan :

exten => _ZXX,n,NoOp(CallerIDnum = ${CALLERID(num)} CallerIDall = 
${CALLERID(all)})

exten => _ZXX,n,NoOp(sipheaderto = ${SIP_HEADER(TO)})
exten => _ZXX,n,NoOp(sipheaderfrom = ${SIP_HEADER(FROM)})


On the Asterisk CLI :


[Sep 22 09:43:04] -- Called SIP/nnsa135
[Sep 22 09:43:04] -- Got SIP response 302 "Moved Temporarily" back 
from 8.9.10.11:65466
[Sep 22 09:43:04] -- Now forwarding SIP/Incoming-0bd9 to 
'Local/208@from-context' (thanks to SIP/nnsa135-0be1)

...
[Sep 22 09:43:04] -- Executing [208@from-context:5] 
NoOp("Local/208@from-context-0079;2", "CallerIDnum = 09210 
CallerIDall = "Cpss" <09210>") in new stack
[Sep 22 09:43:04] -- Executing [208@from-context:6] 
NoOp("Local/208@from-context-0079;2", "sipheaderto = ") in new stack
[Sep 22 09:43:04] -- Executing [208@from-context:7] 
NoOp("Local/208@from-context-0079;2", "sipheaderfrom = ") in new stack





Any more ideas on how to get the value "nnsa135" (being the SIP 
username) please ?





Kind regards.


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Re: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ?

2016-09-19 Thread Jonas Kellens

Hello

I can confirm that the variable DIALEDPEERNAME contains the information 
that I would expect in the variable BRIDGEPEER.


But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of 
Asterisk version 13 ?!


So if this is not the intention, then yes this is probably a bug and 
should be reported.





Kind regards.

Jonas.



On 18-09-16 19:58, Ludovic Gasc wrote:

Hi,

You might use DIALEDPEERNAME instead of BRIDGEPEER.

Nevertheless, I've the same issue with another BRIDGE prefix variable:

I never retrieve at one moment BRIDGEPVTCALLID variable, even if it's 
documented in Asterisk wiki: 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables


Nevertheless, the variable seems to be set in the Asterisk source code:
https://github.com/asterisk/asterisk/blob/13.10/main/bridge.c#L1222
I see no issues open about that, do I need to open an issue ?

Have a nice week.
--
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/

2016-09-17 11:47 GMT+02:00 Jonas Kellens <jonas.kell...@telenet.be 
<mailto:jonas.kell...@telenet.be>>:


Hello

a call goes out and is answered :

[Sep 17 11:29:52] VERBOSE[23420][C-0051] app_dial.c:
SIP/myprovider-010b is making progress passing it to
SIP/mysippeer-0108
[Sep 17 11:30:05] VERBOSE[23420][C-0051] app_dial.c:
SIP/myprovider-010b answered SIP/mysippeer-0108
[Sep 17 11:30:05] VERBOSE[23522][C-0051] bridge_channel.c:
Channel SIP/myprovider-010b joined 'simple_bridge'
basic-bridge 
[Sep 17 11:30:05] VERBOSE[23420][C-0051] bridge_channel.c:
Channel SIP/mysippeer-0108 joined 'simple_bridge' basic-bridge


Call ends :
[Sep 17 11:34:36] VERBOSE[23420][C-0051] bridge_channel.c:
Channel SIP/mysippeer-0108 left 'simple_bridge' basic-bridge

[Sep 17 11:34:36] VERBOSE[23522][C-0051] bridge_channel.c:
Channel SIP/myprovider-010b left 'simple_bridge' basic-bridge




When the call ends in Asterisk version 1.8.32.3 I can read the
variable in h-context.
In Asterisk 13.11.2 this variable is always empty. How come ??


Dialplan logic :
...
exten => h,n,NoOp(bridgepeer = ${BRIDGEPEER})
...


CLI on Asterisk 13.11.2 :
 -- Executing [h@calling:15] NoOp("SIP/mysippeer-4c80",
"bridgepeer = SIP/myprovider-4c83") in new stack


CLI on Asterisk 13.11.2 :
VERBOSE[23420][C-0051] pbx.c: Executing [h@calling:15]
NoOp("SIP/mysippeer-0108", "bridgepeer = ") in new stack


What has changed and how can I get the 13.11 version of
${BRIDGEPEER} ??





Thanks in advance !


Kind regards.

Jonas.

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[asterisk-users] Ast 13.11.2 : bridgepeer variable empty ?

2016-09-17 Thread Jonas Kellens

Hello

a call goes out and is answered :

[Sep 17 11:29:52] VERBOSE[23420][C-0051] app_dial.c: 
SIP/myprovider-010b is making progress passing it to 
SIP/mysippeer-0108
[Sep 17 11:30:05] VERBOSE[23420][C-0051] app_dial.c: 
SIP/myprovider-010b answered SIP/mysippeer-0108
[Sep 17 11:30:05] VERBOSE[23522][C-0051] bridge_channel.c: Channel 
SIP/myprovider-010b joined 'simple_bridge' basic-bridge 

[Sep 17 11:30:05] VERBOSE[23420][C-0051] bridge_channel.c: Channel 
SIP/mysippeer-0108 joined 'simple_bridge' basic-bridge 



Call ends :
[Sep 17 11:34:36] VERBOSE[23420][C-0051] bridge_channel.c: Channel 
SIP/mysippeer-0108 left 'simple_bridge' basic-bridge 

[Sep 17 11:34:36] VERBOSE[23522][C-0051] bridge_channel.c: Channel 
SIP/myprovider-010b left 'simple_bridge' basic-bridge 





When the call ends in Asterisk version 1.8.32.3 I can read the variable 
in h-context.

In Asterisk 13.11.2 this variable is always empty. How come ??


Dialplan logic :
...
exten => h,n,NoOp(bridgepeer = ${BRIDGEPEER})
...


CLI on Asterisk 13.11.2 :
 -- Executing [h@calling:15] NoOp("SIP/mysippeer-4c80", "bridgepeer 
= SIP/myprovider-4c83") in new stack



CLI on Asterisk 13.11.2 :
VERBOSE[23420][C-0051] pbx.c: Executing [h@calling:15] 
NoOp("SIP/mysippeer-0108", "bridgepeer = ") in new stack



What has changed and how can I get the 13.11 version of ${BRIDGEPEER} ??





Thanks in advance !


Kind regards.

Jonas.
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Re: [asterisk-users] Queue show : failed to extend from 240 to 327

2016-09-10 Thread Jonas Kellens

On 10-09-16 09:42, Jonas Kellens wrote:


On 10-09-16 00:50, Richard Mudgett wrote:



On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


Hello

when I type on the Asterisk CLi 'queue show', I first get a list
of my queues and then the following :


failed to extend from 240 to 327




failed to extend from 240 to 334


I could not find any information on this on the web, except this
: https://issues.asterisk.org/jira/browse/ASTERISK-8828
<https://issues.asterisk.org/jira/browse/ASTERISK-8828>

which is an old 'bug' that should have been fixed meanwhile.

Any more thoughts on why I should be getting this message when
asking information about queues (I don't see this message on any
other command).


That message is a result of trying to build a string where the buffer 
is too
small to contain it.  I would expect that there is a truncated string 
in the

'queue show' output.

You haven't stated which Asterisk version you are using.  It may 
already be fixed.


Hello

I have this with asterisk-certified-13.8-cert1 and also with 
asterisk-certified-13.8-cert2


Could it be that the membername value (and interface value) in my 
realtime MySQL table queue_members is too long ??


It looks like this :

Local/01_vlaebidvxcrxrheebdin354@ExternalCallFromQueue
Local/02_vlaebidvxcrxrheebdin114@ExternalCallFromQueue
Local/03_vlaebidvxcrxrheebdin329@ExternalCallFromQueue

I have the idea that this is the "problem".

FYI : it also makes that Asterisk restarts (with core dump) whenever a 
queue is addressed. Very /annoying/ !



So string size too large and buffer too small.

FYI : I do not have this with any version of Asterisk 1.8. This is a 
"problem" that exists only in Asterisk 13.




How to fix this ??



This is an example output for queue show <> on Asterisk version 
asterisk-certified-13.8-cert2 (same on asterisk-certified-13.8-cert1) :



sip*CLI> queue show cvikbubohirndceiaetsq
cvikbubohirndceiaetsq has 0 calls (max unlimited) in 'ringall' strategy 
(0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s

   Members:
  cvikbubohirndceiaets012 
(Local/cvikbubohirndceiaets012@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets012@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no
  cvikbubohirndceiaets248 
(Local/cvikbubohirndceiaets248@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets248@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no
  cvikbubohirndceiaets428 
(Local/cvikbubohirndceiaets428@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets428@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no
  cvikbubohirndceiaets461 
(Local/cvikbubohirndceiaets461@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets461@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no
  cvikbubohirndceiaets629 
(Local/cvikbubohirndceiaets629@ExternalCallFromQueue from 
Local/cvikbubohirndceiaets629@ExternalCallFromQueue) (ringinuse 
disabled) (realtime) (Not in use) has taken no

   No Callers

failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327



Any idea on how to fix this ??


Kind regards.

J.

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Re: [asterisk-users] Queue show : failed to extend from 240 to 327

2016-09-10 Thread Jonas Kellens


On 10-09-16 00:50, Richard Mudgett wrote:



On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


Hello

when I type on the Asterisk CLi 'queue show', I first get a list
of my queues and then the following :


failed to extend from 240 to 327




failed to extend from 240 to 334


I could not find any information on this on the web, except this :
https://issues.asterisk.org/jira/browse/ASTERISK-8828
<https://issues.asterisk.org/jira/browse/ASTERISK-8828>

which is an old 'bug' that should have been fixed meanwhile.

Any more thoughts on why I should be getting this message when
asking information about queues (I don't see this message on any
other command).


That message is a result of trying to build a string where the buffer 
is too
small to contain it.  I would expect that there is a truncated string 
in the

'queue show' output.

You haven't stated which Asterisk version you are using.  It may 
already be fixed.


Hello

I have this with asterisk-certified-13.8-cert1 and also with 
asterisk-certified-13.8-cert2


Could it be that the membername value (and interface value) in my 
realtime MySQL table queue_members is too long ??


It looks like this :

Local/01_vlaebidvxcrxrheebdin354@ExternalCallFromQueue
Local/02_vlaebidvxcrxrheebdin114@ExternalCallFromQueue
Local/03_vlaebidvxcrxrheebdin329@ExternalCallFromQueue

I have the idea that this is the "problem".

FYI : it also makes that Asterisk restarts (with core dump) whenever a 
queue is addressed. Very /annoying/ !



So string size too large and buffer too small.

FYI : I do not have this with any version of Asterisk 1.8. This is a 
"problem" that exists only in Asterisk 13.




How to fix this ??


Kind regards.

Jonas.

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[asterisk-users] Queue show : failed to extend from 240 to 327

2016-09-09 Thread Jonas Kellens

Hello

when I type on the Asterisk CLi 'queue show', I first get a list of my 
queues and then the following :



failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 323
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 323
failed to extend from 240 to 323
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 323
failed to extend from 240 to 334
failed to extend from 240 to 334
failed to extend from 240 to 334
failed to extend from 240 to 327
failed to extend from 240 to 327
failed to extend from 240 to 334
failed to extend from 240 to 334


I could not find any information on this on the web, except this : 
https://issues.asterisk.org/jira/browse/ASTERISK-8828


which is an old 'bug' that should have been fixed meanwhile.

Any more thoughts on why I should be getting this message when asking 
information about queues (I don't see this message on any other command).




Kind regards


Jonas.



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[asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

2016-09-02 Thread Jonas Kellens

Hello

when setting a local forward (in this case to extension 23) on a SIP 
phone, I see the following on the Asterisk CLI :



[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back 
from 11.22.33.44:40670
[Aug 31 14:59:34] -- Now forwarding 
Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal' 
(thanks to SIP/myaccount184-3729)



Question : how can I read the variable which contains the value 
'myaccount184' in the context from-internal ?



The following variables I've tried are empty :

ChannelPeerip=${CHANNEL(peerip)}
Channelrecvip=${CHANNEL(recvip)}
Channelfrom=${CHANNEL(from)}
Channeluri=${CHANNEL(uri)}
Channeluseragent=${CHANNEL(useragent)})


You can see this on the CLI output here :

[Aug 31 14:59:34] -- Executing [23@from-internal:7] 
NoOp("Local/23@from-internal-07f5;2", "ChannelPeerip= Channelrecvip= 
Channelfrom=") in new stack
[Aug 31 14:59:34] -- Executing [23@from-internal:8] 
NoOp("Local/23@from-internal-07f5;2", "Channeluri= 
Channeluseragent=") in new stack






Anyone knows the correct variable to read ?



Kind regards

Jonas.
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Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

2016-08-18 Thread Jonas Kellens

On 17-08-16 23:24, George Joseph wrote:



On Wed, Aug 17, 2016 at 1:40 PM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


On 16-08-16 17:45, George Joseph wrote:



On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:

On 16-08-16 04:38, George Joseph wrote:



On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>>
wrote:

Hello

using pjproject 2.5.5
using asterisk-certified-13.8-cert1


IIRC there were API changes in pjproject 2.5 that aren't
accounted for in asterisk 13.8. Try pjproject 2.4.5 first
and let's see if that works


Compiled pjproject 2.5.5 with :
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1"
--prefix=/usr --libdir=/usr/lib64 --enable-shared
--disable-video --disable-sound --disable-opencore-amr

Compiled Asterisk 13 with
./configure --libdir=/usr/lib64

All pjproject modules are selectable in menuselect, so
here no problem.

Modules are present in /usr/lib64/asterisk/module (see
below).

But when I start asterisk, I get a lot of errors
concerning res_pjsip (see below) on the asterisk CLI.

Anyone have some input on this ?


Thanks.

Kind regards.





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



Hello

how can I disable all modules related to pjsip in modules.conf ??

I have now :

[modules]
autoload=yes
preload => res_config_mysql.so
noload => pbx_gtkconsole.so
noload => res_pjsip.so
noload => res_pjsip_pubsub.so
noload => res_pjsip_session.so
noload => chan_pjsip.so
noload => res_pjsip_exten_state.so
noload => res_pjsip_log_forwarder.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_oss.so
noload => chan_console.so


This does not make the CLI erros go away. I still have the
idea that pjsip is loaded.



I'm not sure what your objective is.  If you want to completely
disable pjsip, run ./configure --without-pjproject.


When I compile "--without-pjproject" I loose all webrtc
functionality. I get errors about the lack of "ice-frag and
ice-pwd in the SDP-body".

So I guess I DO need pjproject. But I do not want to use pjsip (I
prefer sip).

Do you have any other input or idea ?


Ok, I get it now.  Use pjproject-2.4.5 and in menuselect, disable all 
the res_pjsip modules.


I can confirm that compiling pjproject 2.4.5 (but ALSO pjproject 2.5.5) 
with asterisk-certified-13.8-cert1 AND "disable all the res_pjsip 
modules" works fine for me.



Kind regards

J.

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Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

2016-08-18 Thread Jonas Kellens

On 17-08-16 23:17, Jonathan H wrote:

On 17 August 2016 at 20:40, Jonas Kellens <jonas.kell...@telenet.be> wrote:


When I compile "--without-pjproject" I loose all webrtc functionality. I get errors about 
the lack of "ice-frag and ice-pwd in the SDP-body".
So I guess I DO need pjproject. But I do not want to use pjsip (I prefer sip).
Do you have any other input or idea ?

Yes.

I've never had a problem compiling or installing Asterisk; I simply
download the latest version, follow the instructions, and 10 minutes
later I'm compiled and up and running.
No messing about with weird seperate downloads of unsupported versions
of pjsip - I just use the bundled pjsip install and off I go.

But from your posts, it seems you want to do modern web stuff like
WebRTC and so on, on old version of centos, old versions of asterisk,
old version of the SIP channel driver.

What particular reason is there to even bother with the certified
version - the instructions say the regular most recent LTS download
should be first choice.

And why do you prefer SIP? pjsip was introduced in Asterisk 12 nearly
3 years ago, and SIP is pretty much deprecated now.

As a newbie, I looked at SIP and it all seemed a bit bonkers -
"type=friend, insecure=very" - what's THAT all about?!

In pjsip, I just setup a pjsip_wizard and template my endpoints in
pjsip.conf, and I'm done in a few lines.
https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-13-on-Ubuntu.md

This is me, creating a brand new Asterisk install on a low end $5 VPS
which handles more concurrent calls than I need it to (at least 20 so
far!);
https://www.youtube.com/watch?v=h12NkJQwpYo (I just found out that the
Youtube annotations don't work on mobile, so watch on desktop for it
to make sense!).

I'm probably the newbiest of noobs here, but just using the latest
current stable version of everything available and following the
install page on the Asterisk Wiki I can fire up a VPS and be receiving
calls in 20 minutes, from scratch. And I'm genuinely interested in why
people struggle on for days with old versions of things. I'm not
asking all this to create argument, but I am genuinely interested.
Perhaps I am missing a major point here?


Because in some environments stability is far more important than 
'latest' and 'newest'.



Kind regards.

J.



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Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

2016-08-17 Thread Jonas Kellens

On 16-08-16 17:45, George Joseph wrote:



On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


On 16-08-16 04:38, George Joseph wrote:



On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:

Hello

using pjproject 2.5.5
using asterisk-certified-13.8-cert1


IIRC there were API changes in pjproject 2.5 that aren't
accounted for in asterisk 13.8.  Try pjproject 2.4.5 first and
let's see if that works


Compiled pjproject 2.5.5 with :
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr
--libdir=/usr/lib64 --enable-shared --disable-video
--disable-sound --disable-opencore-amr

Compiled Asterisk 13 with
./configure --libdir=/usr/lib64

All pjproject modules are selectable in menuselect, so here
no problem.

Modules are present in /usr/lib64/asterisk/module (see below).

But when I start asterisk, I get a lot of errors concerning
res_pjsip (see below) on the asterisk CLI.

Anyone have some input on this ?


Thanks.

Kind regards.





-- 
George Joseph

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



Hello

how can I disable all modules related to pjsip in modules.conf ??

I have now :

[modules]
autoload=yes
preload => res_config_mysql.so
noload => pbx_gtkconsole.so
noload => res_pjsip.so
noload => res_pjsip_pubsub.so
noload => res_pjsip_session.so
noload => chan_pjsip.so
noload => res_pjsip_exten_state.so
noload => res_pjsip_log_forwarder.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_oss.so
noload => chan_console.so


This does not make the CLI erros go away. I still have the idea
that pjsip is loaded.



I'm not sure what your objective is.  If you want to completely 
disable pjsip, run ./configure --without-pjproject.


When I compile "--without-pjproject" I loose all webrtc functionality. I 
get errors about the lack of "ice-frag and ice-pwd in the SDP-body".


So I guess I DO need pjproject. But I do not want to use pjsip (I prefer 
sip).


Do you have any other input or idea ?


Kind regards.

J.

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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

2016-08-17 Thread Jonas Kellens

Remove yourself !

Don't hijack my thread !



On 17-08-16 14:53, Dario Estupinan wrote:

REMOVE ME please.

2016-08-15 15:16 GMT-05:00 Jonas Kellens <jonas.kell...@telenet.be 
<mailto:jonas.kell...@telenet.be>>:


Hello

after I have upgraded from Asterisk 12 to
asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved
in MySQL DB) register anymore.


[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'<sip:testacc77@178.19.90.240
<mailto:sip%3Atestacc77@178.19.90.240>>' failed for
'78.119.140.190:5076 <http://78.119.140.190:5076>' - Wrong password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'<sip:testacc78@178.19.90.240
<mailto:sip%3Atestacc78@178.19.90.240>>' failed for
'78.119.140.190:5072 <http://78.119.140.190:5072>' - Wrong password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'<sip:testacc79@178.19.90.240
<mailto:sip%3Atestacc79@178.19.90.240>>' failed for
'78.119.140.190:5062 <http://78.119.140.190:5062>' - Wrong password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'<sip:testacc80@178.19.90.240
<mailto:sip%3Atestacc80@178.19.90.240>>' failed for
'78.119.140.190:5060 <http://78.119.140.190:5060>' - Wrong password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'<sip:testacc81@178.19.90.240
<mailto:sip%3Atestacc81@178.19.90.240>>' failed for
'78.119.140.190:5060 <http://78.119.140.190:5060>' - Wrong password


Is this a known problem ??


Second question I have : can I get the complete list of columns
that can be used in realtime database for sip peers somewhere
(update for Ast 13) ? Are columns like dtlsenable, dtlsverify,
dtlscertfile, dtlscafile, dtlssetup possible ??




Thanks for the help.


Kind regards.

Jonas.

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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

2016-08-17 Thread Jonas Kellens


On 15-08-16 23:00, Carlos Chavez wrote:



I highly recommend that you use alembic to set up your database as 
this will make sure you are always using the correct database schema.  
You should be able to find the "official" structure in the 
contrib/realtime/mysql directory of the Asterisk source.




Hello

in contrib/realtime/mysql I see a table 'sippeers' with a column 
"transport ENUM('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp') " but 
I see no columns dtlsenable, dtlsverify, dtlscertfile, dtlscafile, 
dtlssetup ?


So if we can define a sip peer with transport 'ws' or 'wss', then why 
are there no columns for the 'dtls'-part (which is kinda mandatory) ?




Kind regards.



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Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

2016-08-16 Thread Jonas Kellens

On 16-08-16 04:38, George Joseph wrote:



On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens 
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>> wrote:


Hello

using pjproject 2.5.5
using asterisk-certified-13.8-cert1


IIRC there were API changes in pjproject 2.5 that aren't accounted for 
in asterisk 13.8.  Try pjproject 2.4.5 first and let's see if that works



Compiled pjproject 2.5.5 with :
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr
--libdir=/usr/lib64 --enable-shared --disable-video
--disable-sound --disable-opencore-amr

Compiled Asterisk 13 with
./configure --libdir=/usr/lib64

All pjproject modules are selectable in menuselect, so here no
problem.

Modules are present in /usr/lib64/asterisk/module (see below).

But when I start asterisk, I get a lot of errors concerning
res_pjsip (see below) on the asterisk CLI.

Anyone have some input on this ?


Thanks.

Kind regards.





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Hello

how can I disable all modules related to pjsip in modules.conf ??

I have now :

[modules]
autoload=yes
preload => res_config_mysql.so
noload => pbx_gtkconsole.so
noload => res_pjsip.so
noload => res_pjsip_pubsub.so
noload => res_pjsip_session.so
noload => chan_pjsip.so
noload => res_pjsip_exten_state.so
noload => res_pjsip_log_forwarder.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_oss.so
noload => chan_console.so


This does not make the CLI erros go away. I still have the idea that 
pjsip is loaded.




Kind regards.


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[asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

2016-08-15 Thread Jonas Kellens

Hello

after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 
none of my realtime SIP peers (saved in MySQL DB) register anymore.



[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5076' - Wrong 
password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5072' - Wrong 
password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5062' - Wrong 
password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - Wrong 
password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - Wrong 
password



Is this a known problem ??


Second question I have : can I get the complete list of columns that can 
be used in realtime database for sip peers somewhere (update for Ast 13) 
? Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, 
dtlssetup possible ??





Thanks for the help.


Kind regards.

Jonas.

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[asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

2016-08-15 Thread Jonas Kellens

Hello

using pjproject 2.5.5
using asterisk-certified-13.8-cert1

Compiled pjproject 2.5.5 with :
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr 
--libdir=/usr/lib64 --enable-shared --disable-video --disable-sound 
--disable-opencore-amr


Compiled Asterisk 13 with
./configure --libdir=/usr/lib64

All pjproject modules are selectable in menuselect, so here no problem.

Modules are present in /usr/lib64/asterisk/module (see below).

But when I start asterisk, I get a lot of errors concerning res_pjsip 
(see below) on the asterisk CLI.


Anyone have some input on this ?


Thanks.

Kind regards.




[root@sip admin]# ls /usr/lib64/asterisk/modules | grep pjsip
chan_pjsip.so
func_pjsip_aor.so
func_pjsip_contact.so
func_pjsip_endpoint.so
res_pjsip_acl.so
res_pjsip_authenticator_digest.so
res_pjsip_caller_id.so
res_pjsip_config_wizard.so
res_pjsip_dialog_info_body_generator.so
res_pjsip_diversion.so
res_pjsip_dlg_options.so
res_pjsip_dtmf_info.so
res_pjsip_endpoint_identifier_anonymous.so
res_pjsip_endpoint_identifier_ip.so
res_pjsip_endpoint_identifier_user.so
res_pjsip_exten_state.so
res_pjsip_header_funcs.so
res_pjsip_logger.so
res_pjsip_messaging.so
res_pjsip_multihomed.so
res_pjsip_mwi_body_generator.so
res_pjsip_mwi.so
res_pjsip_nat.so
res_pjsip_notify.so
res_pjsip_one_touch_record_info.so
res_pjsip_outbound_authenticator_digest.so
res_pjsip_outbound_publish.so
res_pjsip_outbound_registration.so
res_pjsip_path.so
res_pjsip_pidf_body_generator.so
res_pjsip_pidf_digium_body_supplement.so
res_pjsip_pidf_eyebeam_body_supplement.so
res_pjsip_publish_asterisk.so
res_pjsip_pubsub.so
res_pjsip_refer.so
res_pjsip_registrar_expire.so
res_pjsip_registrar.so
res_pjsip_rfc3326.so
res_pjsip_sdp_rtp.so
res_pjsip_send_to_voicemail.so
res_pjsip_session.so
res_pjsip_sips_contact.so
res_pjsip.so
res_pjsip_t38.so
res_pjsip_transport_management.so
res_pjsip_transport_websocket.so
res_pjsip_xpidf_body_generator.so


Asterisk CLI :

[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_registrar.so': 
/usr/lib64/asterisk/modules/res_pjsip_registrar.so: undefined symbol: 
ast_sip_location_retrieve_aor_contacts_nolock
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_registrar.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_path.so': 
/usr/lib64/asterisk/modules/res_pjsip_path.so: undefined symbol: 
ast_sip_location_retrieve_aor
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_path.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_authenticator_digest.so': 
/usr/lib64/asterisk/modules/res_pjsip_authenticator_digest.so: undefined 
symbol: ast_sip_retrieve_auths
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_authenticator_digest.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_dialog_info_body_generator.so': 
/usr/lib64/asterisk/modules/res_pjsip_dialog_info_body_generator.so: 
undefined symbol: ast_sip_pubsub_unregister_body_generator
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_dialog_info_body_generator.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_sdp_rtp.so': 
/usr/lib64/asterisk/modules/res_pjsip_sdp_rtp.so: undefined symbol: 
ast_sip_session_unregister_supplement
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_sdp_rtp.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_publish_asterisk.so': 
/usr/lib64/asterisk/modules/res_pjsip_publish_asterisk.so: undefined 
symbol: ast_sip_register_publish_handler
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_publish_asterisk.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_send_to_voicemail.so': 
/usr/lib64/asterisk/modules/res_pjsip_send_to_voicemail.so: undefined 
symbol: ast_sip_session_unregister_supplement
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_send_to_voicemail.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_diversion.so': 
/usr/lib64/asterisk/modules/res_pjsip_diversion.so: undefined symbol: 
ast_sip_session_unregister_supplement
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_diversion.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_dlg_options.so': 
/usr/lib64/asterisk/modules/res_pjsip_dlg_options.so: undefined symbol: 

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-14 Thread Jonas Kellens

Hello


I've succeeded in installing Asterisk 13 and more important : I can make 
webRTC call and I have audio !!


For those on the search like myself, I want to spare some weeks of headache.

My steps (CentOS 6.8) :

yum install uuid-devel libuuid-devel autoconf patch automake 
libcurl-devel libogg-devel libvorbis-devel speex-devel popt-devel 
libtool-ltdl-devel libresample-devel gsm-devel libedit-devel 
python-devel jansson-devel binutils-devel


wget http://www.pjsip.org/release/2.5.5/pjproject-2.5.5.tar.bz2
tar -xjvf pjproject-2.5.5.tar.bz2
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr 
--libdir=/usr/lib64 --enable-shared --disable-video --disable-sound 
--disable-opencore-amr

make dep
make
make install
ldconfig -p | grep pj
ldconfig

wget 
http://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-13.8-current.tar.gz
[root@siptest asterisk-certified-13.8-cert1]# ./configure 
--libdir=/usr/lib64

[root@siptest asterisk-certified-13.8-cert1]# make menuselect
[root@siptest asterisk-certified-13.8-cert1]# make && make install


Forget the option "--with-pjproject-bundled" I would say. Did not work 
for me on : CentOS release 6.8 (Final)




Kind regards.


On 12-08-16 17:22, Jonas Kellens wrote:

Hello


running into several problems when installing 
asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 
12).


I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled

First, I do not seem to have res_srtp module available, although all 
necessary libs are present on the system


Second, I am not able to start Asterisk with following error : 
"/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2: 
cannot open shared object file: No such file or directory"





Help appreciated.

Kind regards.




On 12-08-16 16:58, Jonas Kellens wrote:


On 12-08-16 16:38, Joshua Colp wrote:

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm 
having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. 
It's okay if pjproject doesn't as we don't use their media layer. Do 
you have the res_srtp module in Asterisk?




Hello

Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest 
version

Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version

However, I am not able to select res_srtp module in menuselect. It 
says XXX res_srtp module




Kind regards.







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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens

Hello


running into several problems when installing 
asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 12).


I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled

First, I do not seem to have res_srtp module available, although all 
necessary libs are present on the system


Second, I am not able to start Asterisk with following error : 
"/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2: 
cannot open shared object file: No such file or directory"





Help appreciated.

Kind regards.




On 12-08-16 16:58, Jonas Kellens wrote:


On 12-08-16 16:38, Joshua Colp wrote:

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm 
having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. 
It's okay if pjproject doesn't as we don't use their media layer. Do 
you have the res_srtp module in Asterisk?




Hello

Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest 
version

Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version

However, I am not able to select res_srtp module in menuselect. It 
says XXX res_srtp module




Kind regards.





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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens


On 12-08-16 16:38, Joshua Colp wrote:

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm 
having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. 
It's okay if pjproject doesn't as we don't use their media layer. Do 
you have the res_srtp module in Asterisk?




Hello

Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest 
version

Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version

However, I am not able to select res_srtp module in menuselect. It says 
XXX res_srtp module




Kind regards.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens
Question : I noticed I received an error when installing pjproject 
--with-external-srtp


I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm having ??



Kind regards.


On 12-08-16 15:02, Jonas Kellens wrote:

Hello


setting "nat=no" or omitting "nat=" in peer definition does not help 
either. Still no audio.


Why do you think this is a NAT issue ? IP and port information in 
SDP-body is correct.





Kind regards.


On 12-08-16 09:25, Антон Сацкий wrote:


Try delete nat from 77wrtc settings ice should do the same


On Aug 11, 2016 10:00 PM, "Jonas Kellens" <jonas.kell...@telenet.be 
<mailto:jonas.kell...@telenet.be>> wrote:


On 11-08-16 18:03, Matt Fredrickson wrote:

    On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>>
wrote:

My main reason not to upgrade to Ast 13 is because I'm
afraid of losing
functionality as there are certain functions
deprecated/replaced. This can
also cause headache :-)

I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in
this case ? If ICE
and NAT is working (not causing problems) why should Ast
13 bring me audio
and Ast 12 don't ??

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature
standard
like ISDN or SIP.  If you find interop problems in an older
version of
Asterisk with WebRTC, it's likely that it has been fixed in
13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking
to is
not on the same network as the Asterisk server, it's
*possible* you
might need a true TURN server as well, instead of just an ICE
server.

Matthew Fredrickson


Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp
<http://192.158.29.39:3478?transport=udp>
turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA


then Asterisk 12 gets really slow and sometimes unresponsive.
Calls result in 480 request timeout (possibly due to the freeze
of Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.


(putting the above lines back in comment resolves the issue of
Asterisk freeze. This is all EXTREMELY BUGGY !)


Asterisk 13 here I come (with very high expectations).


Kind regards.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens

Hello


setting "nat=no" or omitting "nat=" in peer definition does not help 
either. Still no audio.


Why do you think this is a NAT issue ? IP and port information in 
SDP-body is correct.





Kind regards.


On 12-08-16 09:25, Антон Сацкий wrote:


Try delete nat from 77wrtc settings ice should do the same


On Aug 11, 2016 10:00 PM, "Jonas Kellens" <jonas.kell...@telenet.be 
<mailto:jonas.kell...@telenet.be>> wrote:


On 11-08-16 18:03, Matt Fredrickson wrote:

    On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@telenet.be>>
wrote:

My main reason not to upgrade to Ast 13 is because I'm
afraid of losing
functionality as there are certain functions
deprecated/replaced. This can
also cause headache :-)

I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in
this case ? If ICE
and NAT is working (not causing problems) why should Ast
13 bring me audio
and Ast 12 don't ??

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature
standard
like ISDN or SIP.  If you find interop problems in an older
version of
Asterisk with WebRTC, it's likely that it has been fixed in
13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking
to is
not on the same network as the Asterisk server, it's
*possible* you
might need a true TURN server as well, instead of just an ICE
server.

Matthew Fredrickson


Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp
<http://192.158.29.39:3478?transport=udp>
turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA


then Asterisk 12 gets really slow and sometimes unresponsive.
Calls result in 480 request timeout (possibly due to the freeze of
Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.


(putting the above lines back in comment resolves the issue of
Asterisk freeze. This is all EXTREMELY BUGGY !)


Asterisk 13 here I come (with very high expectations).


Kind regards.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Jonas Kellens

On 11-08-16 18:03, Matt Fredrickson wrote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote:

My main reason not to upgrade to Ast 13 is because I'm afraid of losing
functionality as there are certain functions deprecated/replaced. This can
also cause headache :-)

I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in this case ? If ICE
and NAT is working (not causing problems) why should Ast 13 bring me audio
and Ast 12 don't ??

If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature standard
like ISDN or SIP.  If you find interop problems in an older version of
Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking to is
not on the same network as the Asterisk server, it's *possible* you
might need a true TURN server as well, instead of just an ICE server.

Matthew Fredrickson



Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp
turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA


then Asterisk 12 gets really slow and sometimes unresponsive. Calls 
result in 480 request timeout (possibly due to the freeze of Asterisk).


So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.


(putting the above lines back in comment resolves the issue of Asterisk 
freeze. This is all EXTREMELY BUGGY !)



Asterisk 13 here I come (with very high expectations).


Kind regards.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Jonas Kellens
.190:58814 (type 
08, seq 014122, ts 3292375607, len 000160)
[Aug 11 16:01:20] Sent RTP packet to  178.119.146.190:50760 (via 
ICE) (type 08, seq 033795, ts 3292375600, len 000160)
[Aug 11 16:01:20] Got  RTP packet from178.119.146.190:58814 (type 
08, seq 014123, ts 3292375767, len 000160)
[Aug 11 16:01:20] Sent RTP packet to  178.119.146.190:50760 (via 
ICE) (type 08, seq 033796, ts 3292375760, len 000160)
[Aug 11 16:01:20] Got  RTP packet from178.119.146.190:58814 (type 
08, seq 014124, ts 3292375927, len 000160)
[Aug 11 16:01:20] Sent RTP packet to  178.119.146.190:50760 (via 
ICE) (type 08, seq 033797, ts 3292375920, len 000160)
[Aug 11 16:01:20] Got  RTP packet from178.119.146.190:58814 (type 
08, seq 014125, ts 3292376087, len 000160)
[Aug 11 16:01:20] Sent RTP packet to  178.119.146.190:50760 (via 
ICE) (type 08, seq 033798, ts 3292376080, len 000160)
[Aug 11 16:01:20] Got  RTP packet from178.119.146.190:58814 (type 
08, seq 014126, ts 3292376247, len 000160)
[Aug 11 16:01:20] Sent RTP packet to  178.119.146.190:50760 (via 
ICE) (type 08, seq 033799, ts 3292376240, len 000160)
[Aug 11 16:01:20] Got  RTP packet from178.119.146.190:58814 (type 
08, seq 014127, ts 3292376407, len 000160)
[Aug 11 16:01:20] Sent RTP packet to  178.119.146.190:50760 (via 
ICE) (type 08, seq 033800, ts 3292376400, len 000160)
[Aug 11 16:01:20] Got  RTP packet from178.119.146.190:58814 (type 
08, seq 014128, ts 3292376567, len 000160)







On 10-08-16 22:03, Matt Fredrickson wrote:

My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse.  WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.

Matthew Fredrickson

On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens <jonas.kell...@telenet.be> wrote:

Hello

thank you for your answer.

I don't understand how there are many tutorials and examples on the web
where every time the outcome is a working setup. Very strange I feel now
after my personal experience with Asterisk 11 and webRTC.

You also say Asterisk 13. How about Asterisk 12 then ??



Kind regards.



On 10-08-16 21:53, Matt Fredrickson wrote:

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE.  Are
you sure that the stun server you're using (the google one) still
works?  I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kell...@telenet.be>
wrote:

On 10-08-16 08:52, Ludovic Gasc wrote:

For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/




Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??



Kind regards.



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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Jonas Kellens
My main reason not to upgrade to Ast 13 is because I'm afraid of losing 
functionality as there are certain functions deprecated/replaced. This 
can also cause headache :-)


I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in this case ? If 
ICE and NAT is working (not causing problems) why should Ast 13 bring me 
audio and Ast 12 don't ??




I indeed use SIPML5 demo as quick test-case. So do many tutorials on the 
web.


Self-signed certificates should be OK as long as they are imported in 
the browser. Never knew this could cause audio problems ?





Kind regards.



On 11-08-16 16:25, Jonathan H wrote:
I'm genuinely fascinated why you are insisting on using a version of 
Asterisk almost 3 years old, for which EOL support ended last year.


Is there any particular reason you cannot or will not use the current 
version as others have suggested?


Also, I see you are using Doubango and WebRTC, but in the logs, I see 
WS and WSS.


You NEED to be using 100% WSS otherwise you've not got a hope in hell 
of anything working with WEBRTC.
Check the console of the web browser you are trying to make the call 
from (CTRL-SHIFT-I in Chrome on Windows, for example).


Also, you'll need to be using valid certificates - self-signed 
certificates won't work for any current implementation of WebRTC that 
I know of, certainly not if anything involves current versions of 
Chrome or Firefox. That said, LetsEncrypt certs work fine for this, so 
no need to spend out on one.


Switch to Asterisk 13.10 and save yourself a whole lotta headache.

On 11 August 2016 at 15:09, Jonas Kellens <jonas.kell...@telenet.be 
<mailto:jonas.kell...@telenet.be>> wrote:


Hello

Using Asterisk 12.8.2.


On 10-08-16 22:03, Matt Fredrickson wrote:

My suggestion is to verify and debug against Asterisk 13
first, and
then you can try backing down versions, rather than reverse. 
WebRTC

is a rapidly moving target, and has required ongoing changes
that may
not have made it into older and feature frozen versions of
Asterisk.





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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-10 Thread Jonas Kellens

Hello

thank you for your answer.

I don't understand how there are many tutorials and examples on the web 
where every time the outcome is a working setup. Very strange I feel now 
after my personal experience with Asterisk 11 and webRTC.


You also say Asterisk 13. How about Asterisk 12 then ??



Kind regards.



On 10-08-16 21:53, Matt Fredrickson wrote:

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE.  Are
you sure that the stun server you're using (the google one) still
works?  I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote:

On 10-08-16 08:52, Ludovic Gasc wrote:

For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/




Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??



Kind regards.



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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-10 Thread Jonas Kellens


On 10-08-16 08:52, Ludovic Gasc wrote:


For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/





Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??



Kind regards.


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[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-09 Thread Jonas Kellens

Hello

I'm trying for several days now to get ICE support for my Asterisk 11.23 
on CentOS 6.


My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 
--> softphone Zoiper

(problem : no audio)

Reverse does not work either.
(problem : failed get local SDP)

I followed this guide :

https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

I researched on the web and found this useful thread : 
http://forums.digium.com/viewtopic.php?f=1=90167


This is no question "what is wrong ?". I know what is wrong : I need ICE 
support !

So the question here is : how to get ICE support in my Asterisk ?


I've compiled asterisk as follow :

[root@myserver admin]# yum install uuid-devel libuuid-devel
[root@myserver admin]# ./configure --libdir=/usr/lib64
[root@myserver admin]# make menuselect
[root@myserver admin]# make && make install

In my sip.conf I have :

icesupport = yes

In my rtp.conf I have :

icesupport=yes
stunaddr=stun.l.google.com:19302

My SIP peer definition for webRTC client (sipml5) :

[77wrtc]
type=peer
host=dynamic
username=77wrtc
defaultuser=77wrtc
fromuser=77wrtc
secret=987654
disallow=all
allow=alaw
;allow=gsm
qualify=yes
canreinvite=no
dtmfmode=rfc2833
amaflags=billing
context=testwebrtc
nat=force_rport,comedia
transport=udp,ws,wss
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass

SIP registration works fine :

[Aug  9 22:12:00]   == WebSocket connection from '178.119.146.190:36940' 
for protocol 'sip' accepted using version '13'
[Aug  9 22:12:00] -- Registered SIP '77wrtc' at 
178.119.146.190:36940
[Aug  9 22:12:00]> Saved useragent "IM-client/OMA1.0 
sipML5-v1.2016.03.04" for peer 77wrtc


But when I call from my webRTc client (sipml5 website demo) I have no 
audio. I think this is because there is no ICE support.


You can see in de SIP trace below and the RTP trace below that there is 
no ICE support in Asterisk.



[Aug  9 22:15:50] <--- SIP read from WS:178.119.146.190:36940 --->
[Aug  9 22:15:50] INVITE sip:419@178.18.90.230 SIP/2.0
[Aug  9 22:15:50] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKk2KDePVLlTquEfJzIk7LCMdnOHHk4wn1;rport
[Aug  9 22:15:50] From: 
"77";tag=sRCvFQq3gUMqkl6TKTcl

[Aug  9 22:15:50] To: 
[Aug  9 22:15:50] Contact: 
"77";+g.oma.sip-im;language="en,fr"

[Aug  9 22:15:50] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32
[Aug  9 22:15:50] CSeq: 21553 INVITE
[Aug  9 22:15:50] Content-Type: application/sdp
[Aug  9 22:15:50] Content-Length: 1815
[Aug  9 22:15:50] Max-Forwards: 70
[Aug  9 22:15:50] Authorization: Digest 
username="77wrtc",realm="178.18.90.230",nonce="1d8fa83d",uri="sip:419@178.18.90.230",response="cd2da8d1cbf0a2795b38b2048a3a3c49",algorithm=MD5

[Aug  9 22:15:50] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
[Aug  9 22:15:50] Organization: Doubango Telecom
[Aug  9 22:15:50]
[Aug  9 22:15:50] v=0
[Aug  9 22:15:50] o=- 9108976588890881000 2 IN IP4 127.0.0.1
[Aug  9 22:15:50] s=Doubango Telecom - chrome
[Aug  9 22:15:50] t=0 0
[Aug  9 22:15:50] a=group:BUNDLE audio
[Aug  9 22:15:50] a=msid-semantic: WMS BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps
[Aug  9 22:15:50] m=audio 41178 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 
105 13 126

[Aug  9 22:15:50] c=IN IP4 178.119.146.190
[Aug  9 22:15:50] a=rtcp:42197 IN IP4 178.119.146.190
[Aug  9 22:15:50] a=candidate:1668076467 1 udp 2122260223 192.168.1.122 
41178 typ host generation 0
[Aug  9 22:15:50] a=candidate:1668076467 2 udp 2122260222 192.168.1.122 
42197 typ host generation 0
[Aug  9 22:15:50] a=candidate:3794064647 1 udp 1686052607 
178.119.146.190 41178 typ srflx raddr 192.168.1.122 rport 41178 generation 0
[Aug  9 22:15:50] a=candidate:3794064647 2 udp 1686052606 
178.119.146.190 42197 typ srflx raddr 192.168.1.122 rport 42197 generation 0
[Aug  9 22:15:50] a=candidate:770649923 1 tcp 1518280447 192.168.1.122 0 
typ host tcptype active generation 0
[Aug  9 22:15:50] a=candidate:770649923 2 tcp 1518280446 192.168.1.122 0 
typ host tcptype active generation 0

[Aug  9 22:15:50] a=ice-ufrag:cd8nLIL1irEPdLZt
[Aug  9 22:15:50] a=ice-pwd:97awKXGiAt1TO5jlmb3GMXRy
[Aug  9 22:15:50] a=fingerprint:sha-256 
A2:EF:18:69:AE:9D:D9:90:45:0E:0D:84:5C:A4:AE:59:1C:53:09:11:F2:10:DF:F9:BB:20:E0:9D:6D:ED:BC:13

[Aug  9 22:15:50] a=setup:actpass
[Aug  9 22:15:50] a=mid:audio
[Aug  9 22:15:50] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[Aug  9 22:15:50] a=extmap:3 
http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

[Aug  9 22:15:50] a=sendrecv
[Aug  9 22:15:50] a=rtcp-mux
[Aug  9 22:15:50] a=rtpmap:111 opus/48000/2
[Aug  9 22:15:50] a=fmtp:111 minptime=10; useinbandfec=1
[Aug  9 

Re: [asterisk-users] Setting realm=blabla in sip.conf ignored ?

2016-06-27 Thread Jonas Kellens

Hello

nobody who can help me with this realm issue ??




On 21-06-16 16:36, Jonas Kellens wrote:

Hello


no matter what I set in sip.conf for the param "realm=blablabla" , I 
notice in a wireshark trace file that the realm is completely ignored. 
I see that realm value is still 'asterisk', being the default. Why is 
this ?


(I would like to add a printscreen of the wiresharl trace but then 
this thread is rejected due to message size)



So how can I really change the realm value ?


Thanks.

Jonas.


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[asterisk-users] Setting realm=blabla in sip.conf ignored ?

2016-06-21 Thread Jonas Kellens

Hello


no matter what I set in sip.conf for the param "realm=blablabla" , I 
notice in a wireshark trace file that the realm is completely ignored. I 
see that realm value is still 'asterisk', being the default. Why is this ?


(I would like to add a printscreen of the wiresharl trace but then this 
thread is rejected due to message size)



So how can I really change the realm value ?


Thanks.

Jonas.


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[asterisk-users] function SHARED and function IMPORT : 2 questions

2016-03-02 Thread Jonas Kellens

Hello

I am trying to use the functions SHARED and IMPORT to share variables 
across SIP-channels.


During my use I encounter 2 problems/questions.


Question 1. only 1 shared variable per channel ??

When I set 2 shared variables on a channel, and I read them a bit futher 
in the dialplan, there is only 1 variable that has its value :



Dialplan :
exten => s,n,Set(SHARED(TheINPUT)=1)
exten => s,n,Set(SHARED(MyVar)=222)
...
exten => s,n,Set(Import=${IMPORT(${CHANNEL},TheINPUT)})
exten => s,n,Set(ImportAdv=${IMPORT(${CHANNEL},MyVar)})


Execution :

[Mar  2 14:19:20] -- Executing [s@routing:56] 
Set("SIP/980419-0016", "SHARED(TheINPUT)=1") in new stack
[Mar  2 14:19:20] -- Executing [s@routing:57] 
Set("SIP/980419-0016", "SHARED(MyVar)=222") in new stack

...
[Mar  2 14:19:26] -- Executing [s@routing:80] 
Set("SIP/980419-0016", "Import=1") in new stack
[Mar  2 14:19:26] -- Executing [s@routing:81] 
Set("SIP/980419-0016", "ImportAdv=") in new stack



As you can see, only variable "TheINPUT" has its value ( 1 ). Variable 
'MyVar' is empty.



How come ??




Question 2 : how to set a variable on another channel ??


I try to set a Shared Variable on 1 channel :

exten => s,n,Set(SHARED(TheINPUT,${BRIDGECH})=1)

And read the variable on another channel :

exten => s,n,Set(Import=${IMPORT(${BRIDGECH},TheINPUT)})
exten => s,n,Set(Hell=${IMPORT(TheINPUT)})


Execution :

(here the shared var is set)
[Mar  2 14:58:44] -- Executing [s@routing:58] 
NoOp("SIP/980419-0025", "bridgech = SIP/SipT01-0021") in new stack
[Mar  2 14:58:44] -- Executing [s@routing:59] 
Set("SIP/980419-0025", "SHARED(TheINPUT,SIP/SipT01-0021)=1") in 
new stack


(here the hared var is read)
[Mar  2 14:58:54] -- Executing [s@routing:42] 
Set("SIP/SipT01-0021", "Import=") in new stack
[Mar  2 14:58:54] -- Executing [s@routing:43] 
Set("SIP/SipT01-0021", "Hell=") in new stack



So why is the shared variable "TheINPUT" empty on the channel 
SIP/SipT01-0021 ?? It clealry has been set by the channel 
SIP/980419-0025.





Thank you for answering these 2 questions of mine.




Kind regards,

Jonas.
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Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

2015-10-28 Thread Jonas Kellens

Hello


so I got this working with Google Calendar and meanwhile also with MS 
Exchange.



Does anyone have a working example with Horde Calendar (kronolith)? This 
one seems very tough !






Kind regards

Jonas.



On 27-10-15 14:52, Jonas Kellens wrote:

Mark

thank you for your input.

I am using Asterisk 1.8.32.3 (latest).


I indeed use the "Private iCal url" as presented by Google in Calendar 
settings.



This is my calendar.conf :

[cal0]
type = caldav
url = 
https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60

[cal1]
type = ical
url = 
https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60

[cal2]
type = caldav
url = https://www.google.com/calendar/dav/i...@domain.tld/events/ ; 
Main GMail calendar (the trailing slash is significant!)

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60

[cal3]
type = ical
url = https://www.google.com/calendar/dav/i...@domain.tld/events/ ; 
Main GMail calendar (the trailing slash is significant!)

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60



You see that I try every combination possible.



[Oct 27 14:28:28] WARNING[26748]: res_calendar_caldav.c:118 
auth_credentials: Invalid username or password for CalDAV calendar 'cal2'
[Oct 27 14:28:28] WARNING[26748]: res_calendar_caldav.c:157 
caldav_request: Unknown response to CalDAV calendar cal2, request 
REPORT to /calendar/dav/i...@domain.tld/events/: Could not 
authenticate to server: rejected Basic challenge
[Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:117 
auth_credentials: Invalid username or password for iCalendar 'cal3'
[Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:150 
fetch_icalendar: Unable to retrieve iCalendar 'cal3' from 
'https://www.google.com/calendar/dav/i...@domain.tld/events/': Could 
not authenticate to server: rejected Basic challenge
[Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:477 
ical_load_calendar: Unable to parse iCalendar 'cal3'




Calendar Type   Status
    --
77cal3   ical   free
77cal2   caldav free
77cal1   ical   busy
77cal0   caldav free



It seems I finally have a working example !! Namely :


[cal1]
type = ical
url = 
https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60


So the "Private iCal url" of Google Calendar is the one to go !



Jonas.







On 27-10-15 14:04, Mark Wiater wrote:

On 10/27/2015 8:56 AM, Jonas Kellens wrote:


I have changed this setting at Google but it brings me no success.


Jonas,

I've been using google calendar and Asterisk 1.8 for a couple of 
years now without issue.


I have a note in my configuration that says that I'm using the 
Private ICAL URL from gmail and that it's the only one that worked 
for me. Is that the URL that you're using?


Did you change your type to ical in calendar.conf?

Mark








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Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

2015-10-27 Thread Jonas Kellens

Hello


I have changed type 'caldav' to 'ical', but still no succes :


[Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:117 
auth_credentials: Invalid username or password for iCalendar 'cal1'
[Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:150 
fetch_icalendar: Unable to retrieve iCalendar 'cal1' from 
'https://www.google.com/calendar/dav/i...@mydomain.tld/events/': Could 
not authenticate to server: rejected Basic challenge




siptest*CLI> calendar show calendars
Calendar Type   Status
    --
cal1ical   free




Am I missing something obvious here ?



Kind regards

Jonas




On 26-10-15 17:02, Marek Červenka wrote:

try ical url

caldav switched to Oauth
https://blog.mozilla.org/calendar/2013/09/google-is-changing-the-location-url-of-their-caldav-calendars/

and this looks like you must use Oauth 2.0
https://developers.google.com/google-apps/calendar/caldav/v2/guide

Dne 26.10.2015 v 12:17 Jonas Kellens napsal(a):

Hello


I find very little feedback on the following warning/error when 
trying to connect to Google calendar :



[Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 
auth_credentials: Invalid username or password for CalDAV calendar 'cal1'
[Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 
caldav_request: Unknown response to CalDAV calendar cal1, request 
REPORT to /calendar/dav/i...@mydomain.tld/events/: Could not 
authenticate to server: rejected Basic challenge



[cal1]
type = caldav
url = https://www.google.com/calendar/dav/i...@mydomain.tld/events/
user = i...@mydomain.tld
secret = mysecret
refresh = 15
timeframe = 60



When I go to the URL 
https://www.google.com/calendar/dav/i...@mydomain.tld/events/ I can 
log in with the credentials to the calendar (and get a download 
window for the calendar file).


So it seems not a problem of authentication to me.


But what then could be the real issue here ?





Thanks

Kind regards

Jonas.





--
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Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

2015-10-27 Thread Jonas Kellens

Hello


I have changed this setting at Google but it brings me no success.


I add that I have the same problem with another calendar (Horde) :


[Oct 27 12:08:32] WARNING[24844]: res_calendar_caldav.c:118 
auth_credentials: Invalid username or password for CalDAV calendar 'cal2'
[Oct 27 12:08:32] WARNING[24844]: res_calendar_caldav.c:157 
caldav_request: Unknown response to CalDAV calendar cal2, request REPORT 
to /rpc.php/kronolith/jo...@mydomain.tld/jo...@mydomain.tld.ics: Could 
not authenticate to server: rejected Basic challenge




When using "caldav" or "ical" with Google Calendar, I now get this notice :

[Oct 27 13:43:51] WARNING[25202]: res_calendar_caldav.c:157 
caldav_request: Unknown response to CalDAV calendar cal0, request REPORT 
to 
/calendar/ical/info%40domain.tld/private-6e3543acbc7e2ad02b3d414124a/basic.ics: 
SSL handshake failed: SSL error: GnuTLS internal error.


I have also taken the URL as presented by Google in the Calendar settings.

But as u see... it does not work !




So it does not seem to be a problem with Google Calendar, as the same 
problem occurs with Horde Calendar.




Anyone has a working example please ??




Kind regards

Jonas.




On 27-10-15 13:19, Dan Heywood wrote:


Hi Jonas,

Is it google apps? Try checking the following in your google account 
settings:


Allow less secure apps: ON

Some non-Google apps and devices use less secure sign-in technology, 
which could leave your account vulnerable. You can turn off access for 
these apps (which we recommend) or choose to use them despite the risks.


I had to enable this to allow login from a linux based application in 
order to send out email.


Thanks,

Dan

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas 
Kellens

*Sent:* Tuesday, October 27, 2015 1:33 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Calendar integration : Could not 
authenticate to server: rejected Basic challenge


Hello


I have changed type 'caldav' to 'ical', but still no succes :


[Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:117 
auth_credentials: Invalid username or password for iCalendar 'cal1'
[Oct 27 10:30:38] WARNING[23388]: res_calendar_icalendar.c:150 
fetch_icalendar: Unable to retrieve iCalendar 'cal1' from 
'https://www.google.com/calendar/dav/i...@mydomain.tld/events/': Could 
not authenticate to server: rejected Basic challenge




siptest*CLI> calendar show calendars
Calendar Type   Status
    --
cal1ical   free




Am I missing something obvious here ?



Kind regards

Jonas



On 26-10-15 17:02, Marek Červenka wrote:

try ical url

caldav switched to Oauth

https://blog.mozilla.org/calendar/2013/09/google-is-changing-the-location-url-of-their-caldav-calendars/

and this looks like you must use Oauth 2.0
https://developers.google.com/google-apps/calendar/caldav/v2/guide

Dne 26.10.2015 v 12:17 Jonas Kellens napsal(a):

Hello


I find very little feedback on the following warning/error
when trying to connect to Google calendar :


[Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118
auth_credentials: Invalid username or password for CalDAV
calendar 'cal1'
[Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157
caldav_request: Unknown response to CalDAV calendar cal1,
request REPORT to /calendar/dav/i...@mydomain.tld/events/
<mailto:/calendar/dav/i...@mydomain.tld/events/>: Could not
authenticate to server: rejected Basic challenge


[cal1]
type = caldav
url =
https://www.google.com/calendar/dav/i...@mydomain.tld/events/
user = i...@mydomain.tld <mailto:i...@mydomain.tld>
secret = mysecret
refresh = 15
timeframe = 60



When I go to the URL
https://www.google.com/calendar/dav/i...@mydomain.tld/events/
I can log in with the credentials to the calendar (and get a
download window for the calendar file).

So it seems not a problem of authentication to me.


But what then could be the real issue here ?





Thanks

Kind regards

Jonas.





-- 


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Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

2015-10-27 Thread Jonas Kellens

Mark

thank you for your input.

I am using Asterisk 1.8.32.3 (latest).


I indeed use the "Private iCal url" as presented by Google in Calendar 
settings.



This is my calendar.conf :

[cal0]
type = caldav
url = 
https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60

[cal1]
type = ical
url = 
https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60

[cal2]
type = caldav
url = https://www.google.com/calendar/dav/i...@domain.tld/events/ ; Main 
GMail calendar (the trailing slash is significant!)

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60

[cal3]
type = ical
url = https://www.google.com/calendar/dav/i...@domain.tld/events/ ; Main 
GMail calendar (the trailing slash is significant!)

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60



You see that I try every combination possible.



[Oct 27 14:28:28] WARNING[26748]: res_calendar_caldav.c:118 
auth_credentials: Invalid username or password for CalDAV calendar 'cal2'
[Oct 27 14:28:28] WARNING[26748]: res_calendar_caldav.c:157 
caldav_request: Unknown response to CalDAV calendar cal2, request REPORT 
to /calendar/dav/i...@domain.tld/events/: Could not authenticate to 
server: rejected Basic challenge
[Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:117 
auth_credentials: Invalid username or password for iCalendar 'cal3'
[Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:150 
fetch_icalendar: Unable to retrieve iCalendar 'cal3' from 
'https://www.google.com/calendar/dav/i...@domain.tld/events/': Could not 
authenticate to server: rejected Basic challenge
[Oct 27 14:28:28] WARNING[26746]: res_calendar_icalendar.c:477 
ical_load_calendar: Unable to parse iCalendar 'cal3'




Calendar Type   Status
    --
77cal3   ical   free
77cal2   caldav free
77cal1   ical   busy
77cal0   caldav free



It seems I finally have a working example !! Namely :


[cal1]
type = ical
url = 
https://calendar.google.com/calendar/ical/info%40domain.tld/private-6e3543acbc76853414124a/basic.ics

user = i...@domain.tld
secret = mysecretpasswd
refresh = 15
timeframe = 60


So the "Private iCal url" of Google Calendar is the one to go !



Jonas.







On 27-10-15 14:04, Mark Wiater wrote:

On 10/27/2015 8:56 AM, Jonas Kellens wrote:


I have changed this setting at Google but it brings me no success.


Jonas,

I've been using google calendar and Asterisk 1.8 for a couple of years 
now without issue.


I have a note in my configuration that says that I'm using the Private 
ICAL URL from gmail and that it's the only one that worked for me. Is 
that the URL that you're using?


Did you change your type to ical in calendar.conf?

Mark




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[asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

2015-10-26 Thread Jonas Kellens

Hello


I find very little feedback on the following warning/error when trying 
to connect to Google calendar :



[Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 
auth_credentials: Invalid username or password for CalDAV calendar 'cal1'
[Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 
caldav_request: Unknown response to CalDAV calendar cal1, request REPORT 
to /calendar/dav/i...@mydomain.tld/events/: Could not authenticate to 
server: rejected Basic challenge



[cal1]
type = caldav
url = https://www.google.com/calendar/dav/i...@mydomain.tld/events/
user = i...@mydomain.tld
secret = mysecret
refresh = 15
timeframe = 60



When I go to the URL 
https://www.google.com/calendar/dav/i...@mydomain.tld/events/ I can log 
in with the credentials to the calendar (and get a download window for 
the calendar file).


So it seems not a problem of authentication to me.


But what then could be the real issue here ?





Thanks

Kind regards

Jonas.
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[asterisk-users] Queue priority not respected

2015-10-05 Thread Jonas Kellens

Hello

I notice that priority of queue members is not being respected.

Using mysql realtime.


These are the queue members (in table queue_members) :

Local/queuemem0@ExternalCallFromQueue
Local/queuemem1@ExternalCallFromQueue
Local/queuemem2@ExternalCallFromQueue
Local/queuemem3@ExternalCallFromQueue
Local/queuemem4@ExternalCallFromQueue
Local/queuemem5@ExternalCallFromQueue
Local/queuemem6@ExternalCallFromQueue


Asterisk queue show :


sipserver*CLI> queue show myqueueq
myqueueq has 0 calls (max unlimited) in 'ringall' strategy (10s 
holdtime, 99s talktime), W:0, C:13, A:0, SL:0.0% within 0s

   Members:
  queuemem0 (Local/queuemem0@ExternalCallFromQueue) with penalty 1 
(realtime) (Not in use) has taken 5 calls (last was 551 secs ago)
  queuemem1 (Local/queuemem1@ExternalCallFromQueue) with penalty 1 
(realtime) (Not in use) has taken no calls yet
  queuemem2 (Local/queuemem2@ExternalCallFromQueue) with penalty 2 
(realtime) (Not in use) has taken 4 calls (last was 1314 secs ago)
  queuemem3 (Local/queuemem3@ExternalCallFromQueue) with penalty 2 
(realtime) (Not in use) has taken 3 calls (last was 1408 secs ago)
  queuemem4 (Local/queuemem4@ExternalCallFromQueue) with penalty 4 
(realtime) (Not in use) has taken 1 calls (last was 1937 secs ago)
  queuemem5 (Local/queuemem5@ExternalCallFromQueue) with penalty 4 
(realtime) (Not in use) has taken no calls yet
  queuemem6 (Local/queuemem6@ExternalCallFromQueue) with penalty 3 
(realtime) (Not in use) has taken no calls yet

   No Callers


In verbose log I can see that queuemem6 with penalty 3 is not being 
contacted :


[Oct  5 10:07:17] VERBOSE[21097] app_dial.c: [Oct  5 10:07:17] -- Called 
SIP/queuemem1
[Oct  5 10:07:17] VERBOSE[21098] app_dial.c: [Oct  5 10:07:17] -- Called 
SIP/queuemem0

--> busy
[Oct  5 10:07:17] VERBOSE[21100] app_dial.c: [Oct  5 10:07:17] -- Called 
SIP/queuemem2
[Oct  5 10:07:17] VERBOSE[21099] app_dial.c: [Oct  5 10:07:17] -- Called 
SIP/queuemem3

--> busy
[Oct  5 10:07:17] VERBOSE[21101] app_dial.c: [Oct  5 10:07:17] -- Called 
SIP/queuemem5
[Oct  5 10:07:17] VERBOSE[21102] app_dial.c: [Oct  5 10:07:17] -- Called 
SIP/queuemem4



queuemem0 and queuemem1 (priority 1) are busy, so queuemem2 and 
queuemem3 (priority 2) are being called. So far so good.


But we placed queuemem2 and queuemem3 (priority 2) also busy. So 
queuemem6 should be rung as priority 3.


What do we see : queuemem4 and queuemem5 (priority 4) are being called ! 
This not correct.


queuemem6 is never contacted.



Why are priorities here not beining respected ?



Kind regards

Jonas.


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Re: [asterisk-users] Call Queues : linear strategy WITH priority

2015-08-12 Thread Jonas Kellens


On 12-08-15 16:31, A J Stiles wrote:

On Wednesday 12 Aug 2015, Jonas Kellens wrote:

Hello

I was wondering of it is possible to have Queue Agents with the same
priority (penalty) but with a certain order ?

So I have 20 Agents.

Agent 1 till Agent 10 has penalty 1.

Agent 11 till Agent 15 has penalty 2.
(only contacted if 1 - 10 are busy)

Agent 16 till Agent 20 has penalty 3.
(only contacted if 1 - 10 and 11 - 15 are busy)

Within the range of Agent 1 till Agent 10, can I have a certain order in
these Agents in which they are rung ?? Like Agent 1 - Agent 5 - Agent
2  3  4 - Agent 6 - Agent 7 - Agent 8  9  10.

What's wrong with giving agent 1 penalty 1; agent 5 penalty 2; agents 2, 3 and
4 penalty 3; agent 6 penalty 4, agent 7 penalty 5, and so forth?



By giving a different penalty to Agents 1 to 10, there is no order. With 
penalty, the Agent keeps on being contacted untill it takes the call. 
Many forget that this is how penalties work !



So in stead of going from Agent 1 to Agent 5 to Agent 2,3,4 it is very 
possible that Agent 5 keeps on ringing when Agent 1 is 'busy calling', 
in stead of going further to Agents 2,3,5.


In your case, Agent 5 will be called over and over again untill it takes 
the call.



Not exactly what I'm looking for.



Kind regards,

Jonas.

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[asterisk-users] Call Queues : linear strategy WITH priority

2015-08-12 Thread Jonas Kellens

Hello

I was wondering of it is possible to have Queue Agents with the same 
priority (penalty) but with a certain order ?


So I have 20 Agents.

Agent 1 till Agent 10 has penalty 1.

Agent 11 till Agent 15 has penalty 2.
(only contacted if 1 - 10 are busy)

Agent 16 till Agent 20 has penalty 3.
(only contacted if 1 - 10 and 11 - 15 are busy)


Within the range of Agent 1 till Agent 10, can I have a certain order in 
these Agents in which they are rung ?? Like Agent 1 - Agent 5 - Agent 
2  3  4 - Agent 6 - Agent 7 - Agent 8  9  10.




I guess I need 'linear' strategy, but will penalty option still work ?




Thank you for your feedback.

Kind regards.


Jonas.
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[asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes

2015-08-07 Thread Jonas Kellens

Hello


I have 2 strange errors when using the Background()-application and 
DTMF-input that is received.


First of all, my first 2 lines are not being executed. The first line 
being executed is the Set() application, thus line 3.


Secondly, the received digits (911) is not the same as the EXTEN (which 
is set to 91).




exten = ivr,n,Set(TIMEOUT(digit)=2)
exten = ivr,n,Background(/var/lib/asterisk/sounds/${ASTPROMPT})

exten = _X.,1,NoOp()
exten = _X.,n,NoOp(input=${EXTEN})
exten = _X.,n,Set(choice=${EXTEN})



[Aug  7 12:31:26] -- Executing [ivr@pbx-routing:7] 
Set(SIP/SipAgenT-0626, TIMEOUT(digit)=2) in new stack

[Aug  7 12:31:26] -- Digit timeout set to 2.000
[Aug  7 12:31:26] -- Executing [ivr@pbx-routing:8] 
BackGround(SIP/SipAgenT-0626, /var/lib/asterisk/sounds/5003) in 
new stack
[Aug  7 12:31:26] -- SIP/SipAgenT-0626 Playing 
'/var/lib/asterisk/sounds/5003.slin'


[Aug  7 12:31:41] NOTICE[3886]: ast_expr2.y:763 compose_func_args: 
argbuf allocated 4 bytes;
[Aug  7 12:31:41] NOTICE[3886]: ast_expr2.y:782 compose_func_args: 
argbuf uses 3 bytes;
[Aug  7 12:31:41] -- Executing [911@pbx-routing:1] 
Set(SIP/SipAgenT-0626, choice=91) in new stack




I have reloaded the dialplan several times, but the first 2 lines never 
get executed. In stead they generate the error : ast_expr2.y:763 
compose_func_args: argbuf allocated 4 bytes;



Anyone know what is going on here ?



Kind regards,

Jonas.
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Re: [asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes

2015-08-07 Thread Jonas Kellens

On 07-08-15 13:23, Ethy H. Brito wrote:

On Fri, 07 Aug 2015 12:47:40 +0200
Jonas Kellens jonas.kell...@telenet.be wrote:


Hello


I have 2 strange errors when using the Background()-application and
DTMF-input that is received.

First of all, my first 2 lines are not being executed. The first line
being executed is the Set() application, thus line 3.

Secondly, the received digits (911) is not the same as the EXTEN (which
is set to 91).



exten = ivr,n,Set(TIMEOUT(digit)=2)
exten = ivr,n,Background(/var/lib/asterisk/sounds/${ASTPROMPT})

exten = _X.,1,NoOp()
exten = _X.,n,NoOp(input=${EXTEN})
exten = _X.,n,Set(choice=${EXTEN})



[Aug  7 12:31:26] -- Executing [ivr@pbx-routing:7]
Set(SIP/SipAgenT-0626, TIMEOUT(digit)=2) in new stack
[Aug  7 12:31:26] -- Digit timeout set to 2.000
[Aug  7 12:31:26] -- Executing [ivr@pbx-routing:8]
BackGround(SIP/SipAgenT-0626, /var/lib/asterisk/sounds/5003) in
new stack
[Aug  7 12:31:26] -- SIP/SipAgenT-0626 Playing
'/var/lib/asterisk/sounds/5003.slin'

[Aug  7 12:31:41] NOTICE[3886]: ast_expr2.y:763 compose_func_args:
argbuf allocated 4 bytes;
[Aug  7 12:31:41] NOTICE[3886]: ast_expr2.y:782 compose_func_args:
argbuf uses 3 bytes;
[Aug  7 12:31:41] -- Executing [911@pbx-routing:1]
Set(SIP/SipAgenT-0626, choice=91) in new stack



I have reloaded the dialplan several times, but the first 2 lines never
get executed. In stead they generate the error : ast_expr2.y:763
compose_func_args: argbuf allocated 4 bytes;


Anyone know what is going on here ?



Kind regards,

Jonas.

Hi Jonas

What is the output from dialplan show for this particular piece of code?

cheers

Ethy



Hello


dialplan show shows the following :


'_X.' =  1. NoOp() [pbx_config]
2. NoOp(input=${EXTEN}) [pbx_config]
3. Set(choice=${EXTEN}) [pbx_config]



But like I said, the first 2 lines do not get executed.

I don't understand why if extension is 911 the code says : choice=91 
in this line : -- Executing [911@pbx-routing:1] 
Set(SIP/SipAgenT-0626, choice=91)


You see exten is 911, but when allocating ${EXTEN} to the variable 
choice it suddenly is 91 ?!






Kind regards,

Jonas.

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[asterisk-users] Problem with realtime mysql I can't seem to resolve

2015-05-22 Thread Jonas Kellens

Hello

I have already several Asterisk servers running with similar 
configuration, but now I stumble into a problem.


I have mysql configuration res_config_mysql.conf :

[MyAsteriskDB]
dbhost = 127.0.0.1
dbname = MyAsteriskDB
dbuser = astadmin
dbpass = mysecret
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
requirements=warn ; or createclose or createchar


Realtime seems to be loaded :

*CLI realtime mysql status
general configured for asterisk on socket file /var/lib/mysql/mysql.sock 
with username asterisk.
MyAsteriskDB connected to MyAsteriskDB@127.0.0.1, port 3306 with 
username astadmin for 12 minutes.
[May 22 10:32:02] ERROR[11269]: res_config_mysql.c:1599 mysql_reconnect: 
MySQL RealTime: Failed to connect database server asterisk on 
/var/lib/mysql/mysql.sock (err 1045). Check debug for more info.

*CLI

However, SIP-registration for SIP peer can not be found :

[May 22 10:32:50] NOTICE[11077]: chan_sip.c:24957 
handle_request_register: Registration from 
'sip:testac...@my.ip.ast.ser' failed for '11.22.33.44:5060' - No 
matching peer found


Debug logs say :

[May 22 10:48:06] DEBUG[11077] res_config_mysql.c: MySQL RealTime: 
Connection okay.
[May 22 10:48:06] DEBUG[11077] res_config_mysql.c: MySQL RealTime: 
Retrieve SQL: SELECT * FROM sip_buddies WHERE name = 'testacc66' AND 
host = 'dynamic'
[May 22 10:48:06] DEBUG[11077] res_config_mysql.c: MySQL RealTime: 
Connection okay.
[May 22 10:48:06] DEBUG[11077] res_config_mysql.c: MySQL RealTime: 
Retrieve SQL: SELECT * FROM sip_buddies WHERE name = 'testacc66'



But sip peer testacc66 really exists in my database in table 
sip_buddies. It can not be found ?!



What else is there for me to investigate ? Can u help me ?

Thanks !

Jonas.
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[asterisk-users] Use dialplan variables from MySQL database and replace with value

2015-03-16 Thread Jonas Kellens

Hello


i have the following field (text string) in a MySQL database : 
${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}


I read this string form the database and want to have the dialplan 
variables to be replaced with the correct content.


How can I do this ?


Currently this is not working. The variable ${PARAMS} contains the exact 
string of the database field :


my dialplan :

exten = s,n,MYSQL(Connect connid localhost dbuser dbpass MyTable)
exten = s,n,MYSQL(Query resultid ${connid} SELECT script_url, 
script_params FROM my_tbl WHERE ID=${myID})

exten = s,n,MYSQL(Fetch fetchid ${resultid} scriptURL PARAMS)
exten = s,n,NoOp(scriptURL = ${scriptURL} PARAMS = ${PARAMS})

becomes :

-- Executing [s@sub-details:4] MYSQL(SIP/SipT01-0012, Connect 
connid localhost dbuser dbpass MyTable) in new stack
-- Executing [s@sub-details:5] MYSQL(SIP/SipT01-0012, Query 
resultid 1 SELECT  script_url, script_params FROM my_tbl WHERE ID=2) 
in new stack
-- Executing [s@sub-details:6] MYSQL(SIP/SipT01-0012, Fetch 
fetchid 2 scriptURL PARAMS) in new stack
-- Executing [s@sub-details:7] NoOp(SIP/SipT01-0012, scriptURL = 
call_end.php PARAMS = ${KNUMMER} ${phone_number_to} ${phone_number_from} 
${CHANNEL:4}) in new stack



If the variable ${PARAMS} contains other variables  ${KNUMMER} 
${phone_number_to} ${phone_number_from} ${CHANNEL:4} , how can I use 
the values of these variables in my dialplan ??



I want to use ${KNUMMER} ${phone_number_to} ${phone_number_from} 
${CHANNEL:4}  as input to my script inside my dialplan :


-- Executing [h@pbx-routing:43] System(SIP/SipT01-0012, 
/usr/bin/php /var/lib/asterisk/agi-bin/call_end.php ${KNUMMER} 
${phone_number_to} ${phone_number_from} ${CHANNEL:4}) in new stack


But in stead of having ${KNUMMER} I want to have 112233, and in stead 
of having ${phone_number_to} I want to have 31023456789 and so on...



Is this possible ??


Kind regards,

Jonas.

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[asterisk-users] park()-command always parks on default 701

2014-11-25 Thread Jonas Kellens

Hello,

I have the following in my dialplan :

exten = callpark,n,Set(PARKINGDYNPOS=200-210)
exten = callpark,n,Set(PARKINGDYNCONTEXT=parked_001)
exten = callpark,n,Park(2s,parkinglot_001)

I see on the CLI :

[Nov 25 15:08:47] -- Executing [callpark@pbx-routing:10] 
Set(SIP/SipT01-000b, PARKINGDYNPOS=200-210) in new stack
[Nov 25 15:08:47] -- Executing [callpark@pbx-routing:11] 
Set(SIP/SipT01-000b, PARKINGDYNCONTEXT=parked_001) in new stack
[Nov 25 15:08:47] -- Executing [callpark@pbx-routing:12] 
Park(SIP/SipT01-000b, 5s,parkinglot_001) in new stack
[Nov 25 15:08:47]   == Parked SIP/SipT01-000b on 701 (lot 
parkinglot_001). Will timeout back to extension [pbx-routing] s, 1 in 50 
seconds

[Nov 25 15:08:47] -- Added extension '701' priority 1 to parked_77

Why does Asterisk park on 701 ? Why not on 200 ?


Kind regards,

Jonas.
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Re: [asterisk-users] queue log realtime mysql

2014-11-05 Thread Jonas Kellens

On 04-11-14 11:52, Jonas Kellens wrote:

On 04-11-14 11:50, Ishfaq Malik wrote:


On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be 
mailto:jonas.kell...@telenet.be wrote:


Hello,

I have 5 Asterisk servers all using mysql realtime to store queue
log information.

There is 1 out of 5 servers which stores the data in 4 columns :
'data1' -- 'data 5'.

All other servers store data in 1 column 'data' with the data
seperated by pipe.

I see no difference in my configuration of extconfig.conf and
logger.conf. Maybe a hidden default value ?

Can someone tell me which setting makes the mysql realtime driver
store data in 1 column or in seperate columns ?

Using Asterisk 1.8.12.2



Kind regards,

Jonas.



Are you using mysql_realtime or odbc with a mysql back end?



Using mysql_realtime, not using odbc.



Hello,

is there any more feedback on this ?

I still haven't found the difference in realtime configuration between 
this 1 server and my 4 other servers.



Kind regards,

Jonas.


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[asterisk-users] queue log realtime mysql

2014-11-04 Thread Jonas Kellens

Hello,

I have 5 Asterisk servers all using mysql realtime to store queue log 
information.


There is 1 out of 5 servers which stores the data in 4 columns : 'data1' 
-- 'data 5'.


All other servers store data in 1 column 'data' with the data seperated 
by pipe.


I see no difference in my configuration of extconfig.conf and 
logger.conf. Maybe a hidden default value ?


Can someone tell me which setting makes the mysql realtime driver store 
data in 1 column or in seperate columns ?


Using Asterisk 1.8.12.2



Kind regards,

Jonas.
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Re: [asterisk-users] queue log realtime mysql

2014-11-04 Thread Jonas Kellens

On 04-11-14 11:50, Ishfaq Malik wrote:


On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be 
mailto:jonas.kell...@telenet.be wrote:


Hello,

I have 5 Asterisk servers all using mysql realtime to store queue
log information.

There is 1 out of 5 servers which stores the data in 4 columns :
'data1' -- 'data 5'.

All other servers store data in 1 column 'data' with the data
seperated by pipe.

I see no difference in my configuration of extconfig.conf and
logger.conf. Maybe a hidden default value ?

Can someone tell me which setting makes the mysql realtime driver
store data in 1 column or in seperate columns ?

Using Asterisk 1.8.12.2



Kind regards,

Jonas.



Are you using mysql_realtime or odbc with a mysql back end?



Using mysql_realtime, not using odbc.


Kind regards,

Jonas
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[asterisk-users] ${HASH(SIP_CAUSE,channel-name)}

2014-10-30 Thread Jonas Kellens

Hello,

I read on the wiki :

Asterisk 1.8 will allow to read SIP response codes in the dialplan via 
*${HASH(SIP_CAUSE,channel-name)}*. Additionally make sure you're using 
the destination channel, not the source channel.


But when I use this in my dialplan, this 'variable' is empty.

Dialplan :

exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})

CLI :

[Oct 30 14:48:03] -- Executing [h@pbx-routing:5] 
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6] 
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack



Can anyone tell me how this should be used ?


Kind regards,

Jonas.
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[asterisk-users] dialplan reload context

2014-10-28 Thread Jonas Kellens

Hello,

is it possible to reload just a context in stead of the whole dialplan ?

I see this on the tracker : 
https://issues.asterisk.org/jira/browse/ASTERISK-19934


But is it possible in some Asterisk version ?




Kind regards,

Jonas.
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[asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto

2014-10-09 Thread Jonas Kellens

Hello,

I have added the following to the peer definition :

ignorecryptolifetime=yes


But still Asterisk tells me :


[Oct  9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244 
sdp_crypto_process: Crypto life time unsupported: crypto:1 
AES_CM_128_HMAC_SHA1_80 inline:ikW6yFvdVkSaeTuVO1isTQkdaxOjgQjMEMSGUf+K|2^32
[Oct  9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:254 
sdp_crypto_process: SRTP crypto offer not acceptable
[Oct  9 14:02:34] WARNING[31980]: chan_sip.c:9129 process_sdp: Can't 
provide secure audio requested in SDP offer



What else do I need to configure ?



Kind regards,

Jonas.
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Re: [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto

2014-10-09 Thread Jonas Kellens

On 09-10-14 14:11, Joshua Colp wrote:

Jonas Kellens wrote:

Hello,


Kia ora,


I have added the following to the peer definition :

ignorecryptolifetime=yes


This is not an option within the official tree so unless you've added 
a patch this won't actually do anything.




But still Asterisk tells me :


[Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244 sdp_crypto_process:
Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80
inline:ikW6yFvdVkSaeTuVO1isTQkdaxOjgQjMEMSGUf+K|2^32
[Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:254 sdp_crypto_process:
SRTP crypto offer not acceptable
[Oct 9 14:02:34] WARNING[31980]: chan_sip.c:9129 process_sdp: Can't
provide secure audio requested in SDP offer


What else do I need to configure ?


Currently there is no way to turn this off without modifying the 
source code. I expect this to change in the future based on testing we 
did at SIPit and stuff we learned.


Hello,

any idea where and what to change in the source code then ?

I am able to change the source code, but to do minimal damage I would 
like to know where to change what exactly.


Using asterisk 1.8.12




Kind regards,

Jonas.

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Re: [asterisk-users] sdp_crypto_process: Crypto life time unsupported: crypto

2014-10-09 Thread Jonas Kellens

On 09-10-14 14:28, Joshua Colp wrote:

Jonas Kellens wrote:

Hello,

any idea where and what to change in the source code then ?

I am able to change the source code, but to do minimal damage I would
like to know where to change what exactly.


Yes. In channels/sip/sdp_crypto.c where the line:

ast_log(LOG_NOTICE, Crypto life time unsupported: %s\n, attr);

is remove the:

continue;

Afterwards.



Ok this seems to work ! Thanks.

Does Asterisk now ignore the SRTP crypto offer ? Or does it just ignore 
the lifetime (in this case : |2^32) ?
It does not seem right that Asterisk now should ignore the whole crypto 
offer.




Kind regards,

Jonas.
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Re: [asterisk-users] Grandstream GXP2160 + SRTP

2014-10-08 Thread Jonas Kellens

On 07-10-14 12:32, Jonas Kellens wrote:

Hello,

I am trying to setup a Grandstream GXP2160 IP-phone with secure 
calling (SRTP).


Secure signaling SSIP for registration is working great !

I follow this guide : 
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial


But when I try to make a call with SRTP, I get stuck. There is an 
initial INVITE which is anwered with a 401. There should follow a new 
INVITE with a nonce, but this does not happen. Any idea why ? Is it 
the Grandstream IP-phone ??




--- SIP read from TLS:my.pub.lic.ip:53416 ---
INVITE sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias
From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018
To: sip:0123123...@ast.ser.ver.ip:5061
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 INVITE
Contact: sips:testacc77005@192.168.1.104:5068;transport=tls
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.2.9
Privacy: none
P-Preferred-Identity: sip:testacc77...@ast.ser.ver.ip:5061
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, 
REFER, UPDATE, MESSAGE

Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 522

v=0
o=testacc77005 8004 8000 IN IP4 192.168.1.104
s=SIP Call
c=IN IP4 192.168.1.104
t=0 0
m=audio 5020 RTP/SAVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:8m7ZfG+0t3KBFGK40IfDO11SZ6D54glKKIwdgo00|2^32
a=crypto:2 AES_CM_128_HMAC_SHA1_32 
inline:nn+id/sSK7OErMfnZZduKNPLejpscxx1vUQB2seO|2^32



--- Reliably Transmitting (NAT) to my.pub.lic.ip:53416 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 
192.168.1.104:5068;branch=z9hG4bK60724585;alias;received=my.pub.lic.ip;rport=53416

From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018
To: sip:0123123...@ast.ser.ver.ip:5061;tag=as1e527556
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 INVITE
Server: mydomain
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=mydomain.be, 
nonce=13b47342

Content-Length: 0


--- SIP read from TLS:my.pub.lic.ip:53416 ---
ACK sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias
From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018
To: sip:0123123...@ast.ser.ver.ip:5061;tag=as1e527556
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 ACK
Content-Length: 0



Hello,

I seem to have the same problem with Snom 370 IP-phone. Registration 
works fine ! But I can not make calls with encrypted rtp.



--- SIP read from TLS:my.pub.lic.ip:1068 ---
INVITE sip:0123123...@ast.ser.ver.ip;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.1.107:1068;branch=z9hG4bK-gxm8w1q7l2co;rport
From: sip:testacc77...@ast.ser.ver.ip;tag=zdwiwg10qx
To: sip:0123123...@ast.ser.ver.ip;user=phone
Call-ID: 3c2679977b67-9j0euqvseh5v
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:testacc77003@192.168.1.107:1068;transport=tls;reg-id=1
X-Serialnumber: 0004132E2809
P-Key-Flags: resolution=31x13, keys=4
User-Agent: snom370/8.4.35
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
PRACK, MESSAGE, INFO, UPDATE

Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Call-Info: sip:ast.ser.ver.ip;appearance-index=1
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 632

v=0
o=root 1052895538 1052895538 IN IP4 192.168.1.107
s=call
c=IN IP4 192.168.1.107
t=0 0
m=audio 65418 RTP/SAVP 8 3 18 99 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:KiXn5H+mKwavoDNa1PfnBqPoODTnxK6hOlWSNJM7

a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:99 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=audio 65418 RTP/AVP 8 3 18 99 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:99 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-



--- Reliably Transmitting (NAT) to my.pub.lic.ip:1068 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 
192.168.1.107:1068;branch=z9hG4bK-gxm8w1q7l2co;received=my.pub.lic.ip;rport=1068

From: sip:testacc77...@ast.ser.ver.ip;tag=zdwiwg10qx
To: sip:0123123...@ast.ser.ver.ip;user=phone;tag=as1cd819c5
Call-ID: 3c2679977b67-9j0euqvseh5v
CSeq: 1 INVITE
Server: mydomain
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=mydomain.be, 
nonce=323823f6

Content-Length: 0




--- SIP read from TLS:my.pub.lic.ip:1068 ---
ACK sip:0123123

[asterisk-users] Grandstream GXP2160 + SRTP

2014-10-07 Thread Jonas Kellens

Hello,

I am trying to setup a Grandstream GXP2160 IP-phone with secure calling 
(SRTP).


Secure signaling SSIP for registration is working great !

I follow this guide : 
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial


But when I try to make a call with SRTP, I get stuck. There is an 
initial INVITE which is anwered with a 401. There should follow a new 
INVITE with a nonce, but this does not happen. Any idea why ? Is it the 
Grandstream IP-phone ??




--- SIP read from TLS:my.pub.lic.ip:53416 ---
INVITE sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias
From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018
To: sip:0123123...@ast.ser.ver.ip:5061
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 INVITE
Contact: sips:testacc77005@192.168.1.104:5068;transport=tls
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.2.9
Privacy: none
P-Preferred-Identity: sip:testacc77...@ast.ser.ver.ip:5061
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, 
REFER, UPDATE, MESSAGE

Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 522

v=0
o=testacc77005 8004 8000 IN IP4 192.168.1.104
s=SIP Call
c=IN IP4 192.168.1.104
t=0 0
m=audio 5020 RTP/SAVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:8m7ZfG+0t3KBFGK40IfDO11SZ6D54glKKIwdgo00|2^32
a=crypto:2 AES_CM_128_HMAC_SHA1_32 
inline:nn+id/sSK7OErMfnZZduKNPLejpscxx1vUQB2seO|2^32



--- Reliably Transmitting (NAT) to my.pub.lic.ip:53416 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 
192.168.1.104:5068;branch=z9hG4bK60724585;alias;received=my.pub.lic.ip;rport=53416

From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018
To: sip:0123123...@ast.ser.ver.ip:5061;tag=as1e527556
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 INVITE
Server: mydomain
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=mydomain.be, 
nonce=13b47342

Content-Length: 0


--- SIP read from TLS:my.pub.lic.ip:53416 ---
ACK sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias
From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018
To: sip:0123123...@ast.ser.ver.ip:5061;tag=as1e527556
Call-ID: 1695864968-506...@bjc.bgi.b.bae
CSeq: 50 ACK
Content-Length: 0

-- 
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