Re: [asterisk-users] ODBC freezing Asterisk 13

Saint Michael wrote: ​Many people are reporting the same issue, so it is not my imagination. Asterisk 13 above 13.1 is useless for anybody who ​relies on res_odbc.so. As you know, after that version, the dropped the complexity of Pooling onto unix_odbc itself. Not so simple, it seems. I noticed

[asterisk-users] ODBC freezing Asterisk 13

​Many people are reporting the same issue, so it is not my imagination. Asterisk 13 above 13.1 is useless for anybody who ​relies on res_odbc.so. As you know, after that version, the dropped the complexity of Pooling onto unix_odbc itself. Not so simple, it seems. I noticed that after a few hours

[asterisk-users] 1 way audio but audio+video is fine

Sporadically we get 1 way audio when one party is outside our firewall. The caller is on NAT, and it works fine most of the time. Caller can hear the called party, same thing going the other direction. Caller can hear called party. Asterisk 13.9 on Debian chan_sip with two identical Grandstream

Re: [asterisk-users] Asterisk and Yealink T21P E2

No problems with authentication during invite after reboot? I'm using insecure=no in SIP configuration. Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres

Re: [asterisk-users] Asterisk and Yealink T21P E2

On 07/14/2016 02:14 PM, Marcelo Terres wrote: Hello. Anybody in the list is using this IP phone? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres

Re: [asterisk-users] Compile of smsq.c failed on Ubuntu Xenial (16.04LTS)

On 2016-07-13 17:09, Ernie Dunbar wrote: Hi everyone. I'm trying to compile Asterisk with the smsq utility on Ubuntu 16.04 LTS, and while most things are compiling fine, smsq fails with the following output: root@test25:/usr/src/asterisk-certified-13.1-cert7/utils# make smsq [CC] smsq.c ->

[asterisk-users] Asterisk and Yealink T21P E2

Hello. Anybody in the list is using this IP phone? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres --

[asterisk-users] Voicemail Mailboxes + Cassandra

Hi List I have two questions: 1- Mailbox on the Asterisk Voicemail Server are created automatically? 2- Is there any support on the code to put the voice records on a Cassandra NoSQL database? BR Joaquin This email is confidential and may be subject to privilege. If you are

[asterisk-users] Certified Asterisk 13.8-cert1 Now Available

The Asterisk Development Team has announced the release of Certified Asterisk 13.8-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.8-cert1 resolves several issues reported by the

Re: [asterisk-users] Force out-bond call to specific CIC

Yes, as far as I remember, in your dial string, simply use a Dial(DAHDI/X/1234567) where X is the dahdi device channel number. Hope that helps. Matthew Fredrickson On Wed, Jul 13, 2016 at 5:22 AM, Mehdi Shirazi wrote: > Hi > > How is it possible to use Dial application

[asterisk-users] CDR replacement with CEL

hi, i'm trying replace CDR with CEL reasons: - minimize Stasis listeners (CDR) - CEL, CDR produces "similar" data - own logic of CDR meaning like "calldate,src,dst,direction,.." dst is always first connected point in PBX - real user or IVR/queue etc., numbers are only attributes of object

Re: [asterisk-users] PJSIP defaults for endpoints when using realtime

On Thu, Jul 14, 2016 at 6:45 AM, A J Stiles wrote: > On Thursday 14 Jul 2016, Joshua Colp wrote: > > Carlos Chavez wrote: > > > Until Asterisk 11 I could use sip.conf to set defaults for all phones > > > (language, dtmf, vmexten, etc) and just leave many fields in

Re: [asterisk-users] PJSIP defaults for endpoints when using realtime

On Thursday 14 Jul 2016, Joshua Colp wrote: > Carlos Chavez wrote: > > Until Asterisk 11 I could use sip.conf to set defaults for all phones > > (language, dtmf, vmexten, etc) and just leave many fields in the > > database as NULL. What would be the proper way to do this for Asterisk > > 13 and

Re: [asterisk-users] PJSIP defaults for endpoints when using realtime

Carlos Chavez wrote: Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? Kia ora, PJSIP doesn't have the ability in it to

Re: [asterisk-users] Asterisk 13 MWI

On Wed, Jul 13, 2016 at 3:44 PM, Carlos Chavez wrote: > On 7/12/16 9:27 PM, George Joseph wrote: > > > > On Tue, Jul 12, 2016 at 11:55 AM, Carlos Chavez > wrote: > >> I am still a little confused about how to activate MWI with PJSIP on >>

Re: [asterisk-users] PJSIP defaults for endpoints when using realtime

with templates. Regards El 13/07/2016 a las 23:49, Carlos Chavez escribió: Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13