Re: [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

2017-01-03 Thread Dmitriy Serov
Joshua, issue has been filed. Thank you! https://issues.asterisk.org/jira/browse/ASTERISK-26689 03.01.2017 20:58, Joshua Colp пишет: On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1]

Re: [asterisk-users] Saving endpoint statuses to database with pjsip and realtime

2017-01-03 Thread Anton Teyhrib
Ok, thank you! Happy new year:) 3 янв. 2017 г. 11:13 PM пользователь "Joshua Colp" написал: > On Tue, Jan 3, 2017, at 02:12 PM, Anton Teyhrib wrote: > > Hi Joshua! > > Thank you for response. > > I want to show user availability in external web application (ruby). I > > also >

Re: [asterisk-users] Saving endpoint statuses to database with pjsip and realtime

2017-01-03 Thread Joshua Colp
On Tue, Jan 3, 2017, at 02:12 PM, Anton Teyhrib wrote: > Hi Joshua! > Thank you for response. > I want to show user availability in external web application (ruby). I > also > found that ARI can do this. What is the best way to do that? ARI or AMI? Either should be able to do what you need, but

Re: [asterisk-users] Saving endpoint statuses to database with pjsip and realtime

2017-01-03 Thread Anton Teyhrib
Hi Joshua! Thank you for response. I want to show user availability in external web application (ruby). I also found that ARI can do this. What is the best way to do that? ARI or AMI? 3 янв. 2017 г. 11:06 PM пользователь "Joshua Colp" написал: On Thu, Dec 29, 2016, at 03:43

Re: [asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?

2017-01-03 Thread Joshua Colp
On Fri, Dec 30, 2016, at 05:04 AM, Kevin Long wrote: > > > Hello, > > I am using asterisk 14.2 and PJSIP, with TLS transport. > > I’m sure I’m doing something wrong here .. > > > In 2 distinct softphone clients (Bria and Groundwire), I am able to > register successfully, and place a SIP

Re: [asterisk-users] Saving endpoint statuses to database with pjsip and realtime

2017-01-03 Thread Joshua Colp
On Thu, Dec 29, 2016, at 03:43 PM, Anton Teyhrib wrote: > Hi all, > Is there any native way to save endpoint statuses to database? > I use asterisk 13 with pjsip and realtime, and didn't found proper way. > I read that there is config parameter in sip.conf: rtupdate=yes. But how > can I do that

Re: [asterisk-users] 183 Session in Progress. Disconnecting channel for lack of RTP activity

2017-01-03 Thread Joshua Colp
On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote: > Yes, this means the remote end was not sending any audio stream. > But it shouldn't. > According to [1] before remote end send OK or ACK there is one way SDP, > no any audio stream. > PJSIP channel (option rtp_timeout) does not take this

Re: [asterisk-users] Fax faling on PJSip

2017-01-03 Thread Joshua Colp
On Tue, Dec 20, 2016, at 12:21 PM, Bryant Zimmerman wrote: > I am working on moving from version 11 to version 13 for my fax > applications. > We are bumping into an issue where the bulk of the T38 faxes are > failing. > > The sending test switch is reporting COMREC_ERR_TRANSMIT_PHASE > >

Re: [asterisk-users] how to add area code to outgoing number in Asterisk 13.13

2017-01-03 Thread Joshua Colp
On Tue, Jan 3, 2017, at 12:32 PM, Motty Cruz wrote: > Thank you Carlos, you’re right I am using PJSIP. Should I not use it? PJSIP allows the following two formats: PJSIP/@ PJSIP// Where in the second case the SIP URI can contain the number. Provided you use either of those (the first would be

Re: [asterisk-users] Find out what context is the exten from

2017-01-03 Thread Joshua Colp
On Thu, Dec 29, 2016, at 02:33 PM, Gabriel Ortiz Lour wrote: > Hi all, > Anyone with a clever ideia to find out (programmaticaly) on what > context, > within the dialplan structure, a exten is just telling the base context > (with many includes inside) ? > Att. Can you provide more detail

Re: [asterisk-users] how to add area code to outgoing number in Asterisk 13.13

2017-01-03 Thread Motty Cruz
Thank you Carlos, you’re right I am using PJSIP. Should I not use it? Thanks, Motty From: Carlos Chavez [mailto:cur...@telecomab.mx] Sent: Saturday, December 31, 2016 5:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Motty Cruz Subject: Re: [asterisk-users] how to

Re: [asterisk-users] Does HEP require PJSIP or does it also works with SIP ?

2017-01-03 Thread Annus Fictus
Hello, I'm not totally sure but HEP permit SIP signaling and RTCP data capture only on PJSIP channels. For chan_sip you have to use captagent. Regards El 03/01/2017 a las 10:04, Olivier escribió: Hello, On a newly built Asterisk 13.13.1 system, I can't make HEP work with chan_sip (though

Re: [asterisk-users] Does HEP require PJSIP or does it also works with SIP ?

2017-01-03 Thread Joshua Colp
On Tue, Jan 3, 2017, at 11:04 AM, Olivier wrote: > Hello, > > On a newly built Asterisk 13.13.1 system, I can't make HEP work with > chan_sip (though I could make it work with PJSIP on another instance). > > Looking either at [1] or hep.conf, I can't see anything meaning HEP > requires PJSIP. >

[asterisk-users] Does HEP require PJSIP or does it also works with SIP ?

2017-01-03 Thread Olivier
Hello, On a newly built Asterisk 13.13.1 system, I can't make HEP work with chan_sip (though I could make it work with PJSIP on another instance). Looking either at [1] or hep.conf, I can't see anything meaning HEP requires PJSIP. Before diging deeper, may I simply ask if HEP requires PJSIP or