Re: [asterisk-users] Fwd: Backport fix

2016-08-17 Thread Joshua Colp
Saint Michael wrote: There is a big bug https://issues.asterisk.org/jira/browse/ASTERISK-24768 It affects version 11, but it was fixed only from 13.20 onwards. However, millions of people still use version 11. This bug makes Asterisk crash

[asterisk-users] Fwd: Backport fix

2016-08-17 Thread Saint Michael
There is a big bug https://issues.asterisk.org/jira/browse/ASTERISK-24768 It affects version 11, but it was fixed only from 13.20 onwards. However, millions of people still use version 11. This bug makes Asterisk crash every few hours under any load that has RTP going through Asterisk. For example,

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

2016-08-17 Thread George Joseph
On Wed, Aug 17, 2016 at 1:40 PM, Jonas Kellens wrote: > On 16-08-16 17:45, George Joseph wrote: > > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > wrote: > >> On 16-08-16 04:38, George Joseph wrote: >> >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> wrote: >> >>> Hello >>> >>>

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

2016-08-17 Thread Jonathan H
On 17 August 2016 at 20:40, Jonas Kellens 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 ot

Re: [asterisk-users] SIP 603 response when call is not answered

2016-08-17 Thread Matthew Jordan
On Wed, Aug 17, 2016 at 8:29 AM, Hooman Fazaeli wrote: > On 2016-08-16 12:10, Joris Engbers wrote: >> >> Hooman Fazaeli writes: >> >>> Hi >>> >>> I have noticed that asterisk returns 'SIP 603' when the called party does >>> not answer. >>> >>> My test setup is simple: two SIP phones (extensions: 1

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 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 mailto:jonas.kell...@telenet.be>> wrote: Hello

[asterisk-users] SIP client open URL upon answer (was : Re: SIP client able to handle Access-URL: header)

2016-08-17 Thread Bertrand LUPART - Linkeo.com
Hello, > I'm playing with the optional URL parameter of the Dial() command, which > "will also be sent to the called party upon successful connection, if the > channel technology supports the sending of URLs in this way."[1] > > Basically, the following asterisk dialplan directive : > > - - 8

Re: [asterisk-users] Farewell

2016-08-17 Thread Josh Reynolds
Congrats! Enjoy the time away, you've earned it :) --- Josh Reynolds josh@engineered.online On Wed, Aug 17, 2016 at 8:37 AM, Vincent Medina wrote: > I just wanted to wish all of you good luck I'm officially retired and will > be removing my name from the list. I can attest that this list has be

[asterisk-users] Farewell

2016-08-17 Thread Vincent Medina
I just wanted to wish all of you good luck I'm officially retired and will be removing my name from the list. I can attest that this list has been a great help throughout my career. I have deployed probably over 100 installations over a 10-year period.  Any of you newcomers this list the m

Re: [asterisk-users] SIP 603 response when call is not answered

2016-08-17 Thread Hooman Fazaeli
On 2016-08-16 12:10, Joris Engbers wrote: Hooman Fazaeli writes: Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds

Re: [asterisk-users] How to remove unused custom hints?

2016-08-17 Thread Tomáš Holý
Perfect. That is exactly what I need. Thank you very much. Nice day! S pozdravem Tomáš Holý INTERCONNECT s.r.o. Zákaznická linka: +420 61333 TEL: +420 61321 FAX: +420 246063179 h...@interconnect.cz Dne úterý 16. srpna 2016 12:01:20 CEST, John Kiniston napsal(a): You can delete them

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 >: Hello after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of my realtime

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

2016-08-17 Thread John Novack
Remove yourself READ - Included with every message - asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dario Estupinan wrote: asterisk-users mailing list

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

2016-08-17 Thread Dario Estupinan
REMOVE ME please. 2016-08-15 15:16 GMT-05:00 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: >

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 s