[asterisk-users] Cannot get my first WebRTC experiment to work.

2015-01-28 Thread Antonio Gómez Soto
Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works. My network setup by the way: I am working from a cable modem,

Re: [asterisk-users] Dialing from phonebook, and hiding the dialed number from the user.

2015-01-27 Thread Antonio Gómez Soto
Hi AJ, Yes, interesting idea, thanks! Antonio On Tue, Jan 27, 2015 at 5:09 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Monday 26 Jan 2015, Antonio Gómez Soto wrote: Hi, does anyone have a recommendation for a SIP phone, which allows dialing from a phonebook, and hiding

[asterisk-users] Dialing from phonebook, and hiding the dialed number from the user.

2015-01-26 Thread Antonio Gómez Soto
Hi, does anyone have a recommendation for a SIP phone, which allows dialing from a phonebook, and hiding the dialed number from the end users? Also from the call history of course. It seems Mitel can do this, and I have a use case where this is a requirement. Thanks, Antonio --

[asterisk-users] Need help interpreting SDP on failing WebRTC connection

2015-01-26 Thread Antonio Gómez Soto
Hi, I am trying to setup a WebRTC connection to asterisk 1.13.0. Using Bria a regular SIP connection works, but using sipml5 on chrome, I got nothing. My network setup by the way: I am working behind a comcast cable modem, the test setup is at digital ocean, and from my laptop I also have a

[asterisk-users] Need some help interpreting SDP on a failing WebRTC connection

2015-01-26 Thread Antonio Gómez Soto
Hi, I am trying to setup a WebRTC connection to asterisk 1.13.0. Using Bria a regular SIP connection works, but using sipml5 on chrome, I got nothing. My network setup by the way: I am working behind a comcast cable modem, the test setup is at digital ocean, and from my laptop I also have a

[asterisk-users] Wiki (pjsip+realtime) says don't put the transports into realtime. Still true?

2015-01-25 Thread Antonio Gómez Soto
Hi, The asterisk wiki page says: Sorcery.conf allows you to try to configure other PJSIP objects such as transport using realtime and it currently won't stop you from doing so. However, some of these object types should not be used with realtime and this can lead to errant behavior. Which

[asterisk-users] Confused by concepts behind pjsip: endpoint, aor, contact

2015-01-04 Thread Antonio Gómez Soto
Hello, I am slightly confused by the difference between chan_sip and pjsip. Especially the new (to me) objects aor and contact. I am having trouble mapping them to the typical SIP configuration settings on a phone. Suppose I have a phone with two line buttons, for two extension numbers. Now, I

Re: [asterisk-users] Confused by concepts behind pjsip: endpoint, aor, contact

2015-01-04 Thread Antonio Gómez Soto
Thanks for responding, On Sun, Jan 4, 2015 at 5:45 PM, George Joseph george.jos...@fairview5.com wrote: On Sun, Jan 4, 2015 at 3:29 PM, Antonio Gómez Soto antonio.gomez.s...@gmail.com wrote: Hello, I am slightly confused by the difference between chan_sip and pjsip. Especially the new

Re: [asterisk-users] Confused by concepts behind pjsip: endpoint, aor, contact

2015-01-04 Thread Antonio Gómez Soto
Joshua, On Sun, Jan 4, 2015 at 6:39 PM, Joshua Colp jc...@digium.com wrote: [..snip..] Also I notice, an AOR does seem do be directly correlated with an auth record, so why are they separate in the configuration, why not unify the aor and the auth objects? They aren't at all. Auth =

Re: [asterisk-users] Confused by concepts behind pjsip: endpoint, aor, contact

2015-01-04 Thread Antonio Gómez Soto
On Sun, Jan 4, 2015 at 8:48 PM, Joshua Colp jc...@digium.com wrote: Antonio Gómez Soto wrote: snip I did not mean they are the same, I meant that there seems to be a one-to-one relationship. So I am wondering, since the auth does seem useless without an aor, but an aor can exist