similar to: PJSIP defaults for endpoints when using realtime

I use realtime for my Asterisk configuration and are now making the transition to Asterisk 13 and PJSIP. I used alchemy to set up my databases and I can now configure my endpoints. While trying to configure a trunk I can see that there is a database table called ps_registrations that should be used to make the registration to the provider but there is no corresponding entry in the

Has anyone successfully used Mysql realtime PJSIP with Asterisk 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the following error now: Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 12 14:42:35] WARNING[24498]:

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> Is it possible to use serveral protocols for a single transport section >> in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you >> cound use webrtc along with your phones but if I try: >> >> [transport-udp] >> type=transport >> protocol=udp,ws,wss >> bind=0.0.0.0 >

On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> > wrote: > >> Has anyone successfully used Mysql realtime PJSIP with Asterisk >> 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the >> following error now: >>

Is it possible to use serveral protocols for a single transport section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0.0.0 I get an error that transport-udp is not found. Do I need a dedicated interface for WebRTC? [Feb 15 12:42:10] ERROR[3308]:

On Mon, Sep 12, 2016 at 3:01 PM, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 9/12/16 3:39 PM, George Joseph wrote: > > > > On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> >> wrote: >> >>> Has

I just finished installing Asterisk 13 on our test server and I can now use PJSIP to register phones and make and receive calls. The only problem I am having is that when I register multiple phones to a single account only one of them rings. The AOR for the account has maxcontacts at 3. If I do a pjsip show endpoints I can see two "Contact" entries which I take to mean that

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any suggestions as to what I am doing wrong? The SIP provider says the latest version of Asterisk they have

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 think that means two 'endpoints' in pjsip right? But what exactly is the difference between

Howdy, I'm trying to re-write my voicemail check extension. I formerly used the SIPPEER function to get the mailbox for a peer with ${SIPPEER(${peer},mailbox)} Is there a way to do this with PJSIP now that I've converted over? I see a function PJSIP_ENDPOINT and it has a mailboxes subset but I'm not retrieving any data from it when I query it. -- A human being should be

Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162

Thank you for the speedy reply. My originate string is something like the following where xxxxx is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's... SIP/outbound.vitelity.net/1234567890 Have a great day! Dan -----Original Message----- From:

Hello, I'm currently evaluating asterisk 13 (Currently on 11). We're testing the migration from SIP to PJSIP. Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everyone, I am starting to work with PJSIP on release 12.1.0.rc3. I used to have Asterisk 1.8 with the regular sip channel. I was using the usereqphone settings in order to set user=phone on from and to URIs. Is there a similar config in PJSIP? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there some documentation for all the available sorcery.conf mappings for realtime? Asterisk already includes some tables in the database that are not enabled by default on the sorcery.conf like transports and outbound registrations. There are no examples in the file on how to enable them. Where can I find some documentation to enable those mappings? -- Telecomunicaciones Abiertas de

I am still a little confused about how to activate MWI with PJSIP on Asterisk 13.9.1. I use realtime for configuration. So far I have tried setting both the mailboxes field on ps_endpoints and the mailboxes field in ps_aors but I cannot get the indicator lamp to blink on any of my phones (Digium, Aastra and Yealink). I have tried just the number of the mailbox and also adding the context.

Greetings everyone, I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried setting x and y to -30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10, and 100 and there appears to be no change on the phone volume. I can see that the

I use realtime on my asterisk installation. I have always used mysql for my realtime connection but as mysql seems to be on the "soon to be deprecated" list of asterisk features I am trying to move to ODBC (still using MariaDB/Mysql on backend). I find ODBC support in Asterisk very unstable. Just today my asterisk server (a test server and a production server) has crashed often