Re: [asterisk-users] SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED

Hello List I would very much like to have some feedback on this. Where do I have to look ? Is it in the Asterisk version (13.38.3) maybe ? Is it for sure in my config ?! Kind regards. Op 28/06/2023 om 16:14 schreef Jonas Kellens: Hello list when trying to set up webRTC

[asterisk-users] SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED

Hello list when trying to set up webRTC communications with sipjs client package (tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file the following : DEBUG[30891][C-] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 99.88.77.66... OK.

Re: [asterisk-users] Discrepancy between Asterisk console and Asterisk Manager DeviceStateChange

On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello I notice a major difference in what Asterisk console is telling me (which seems correct) and what Asterisk Manager is telling. A SIP user is called, and the phone does no

[asterisk-users] Discrepancy between Asterisk console and Asterisk Manager DeviceStateChange

Hello I notice a major difference in what Asterisk console is telling me (which seems correct) and what Asterisk Manager is telling. A SIP user is called, and the phone does not ring. This is the situation. On Asterisk console I see (which seems to be in line with an unreachable phone) :

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

Hello Joshua could it be a bug ? I am using asterisk-certified-13.21-cert6 Kind regards. J. Op 01-07-21 om 20:20 schreef Joshua C. Colp: On Thu, Jul 1, 2021 at 3:15 PM Jonas Kellens <mailto:jonas.kell...@telenet.be>> wrote: Hello Joshua this is the SIP REGISTER at

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

this happens only once ? And why should there *ever* be an "Expired" if there is a SIP REGISTER every 180 seconds ?! Kind regards. Op 01-07-21 om 17:41 schreef Joshua C. Colp: On Thu, Jul 1, 2021 at 12:34 PM Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello Josh

Re: [asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

   [6] => ) So there is a re-register at 11:10:45 How do you explain the "Expired" 10 minutes later ?? Op 30-06-21 om 20:32 schreef Joshua C. Colp: On Wed, Jun 30, 2021 at 3:28 PM Jonas Kellens mailto:jonas.kell...@telenet.be>> wrote: Hello I see the following

[asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

Hello I see the following event from the Asterisk Manager : 2021-06-30 11:20:55 Array (     [0] => Event: PeerStatus     [1] => Privilege: system,all     [2] => SystemName: tstv7     [3] => ChannelType: SIP     [4] => Peer: SIP/testacc7700921     [5] => PeerStatus: Unregistered     [6] =>

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

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

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

, 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/se

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

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

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

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-c

[asterisk-users] defaultexpiry & maxexpiry on peer level

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"

[asterisk-users] Asterisk manager : core show hints

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 

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

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

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

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

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

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

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

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

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

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

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

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 not

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

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

[asterisk-users] doing dnsmgr_lookup for

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]   

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

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

Re: [asterisk-users] BLF and Call Queues

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:

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

on-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 K

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

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

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

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

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

ugh 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 transf

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

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. --

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

, 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 ;

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

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.

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

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

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

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

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

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: Le

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

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

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

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

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

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(

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

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 =>

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

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...

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

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

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

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

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

ile 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 :

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

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:

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

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.kel

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

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:

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

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.kel

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

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

[asterisk-users] Asterisk 13.12.2 : strange queue behaviour

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

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

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

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

-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

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

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 v

[asterisk-users] Realtime queue & agent groups

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=

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

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

[asterisk-users] Problem setting up ssl connection

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:

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

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 M

[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.

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

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&

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

? 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_dia

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

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

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

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 li

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

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 :

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

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

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

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

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

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.kel

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

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

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

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.kel

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

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 asteri

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

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

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

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

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

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

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

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

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

: 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

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

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 Kel

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

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 anyt

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

nd 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

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

000wrtc 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 &l

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

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 he

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

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 ve

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

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

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

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

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

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

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

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

Re: [asterisk-users] Setting realm=blabla in sip.conf ignored ?

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-users] Setting realm=blabla in sip.conf ignored ?

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

[asterisk-users] function SHARED and function IMPORT : 2 questions

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,

Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

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

Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

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:

Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

gium.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'

Re: [asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

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

[asterisk-users] Calendar integration : Could not authenticate to server: rejected Basic challenge

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]:

[asterisk-users] Queue priority not respected

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

Re: [asterisk-users] Call Queues : linear strategy WITH priority

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

[asterisk-users] Call Queues : linear strategy WITH priority

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.

[asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes

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

Re: [asterisk-users] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes

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

[asterisk-users] Problem with realtime mysql I can't seem to resolve

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 =

[asterisk-users] Use dialplan variables from MySQL database and replace with value

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

[asterisk-users] park()-command always parks on default 701

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]

Re: [asterisk-users] queue log realtime mysql

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

[asterisk-users] queue log realtime mysql

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

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