search for: murbl

Displaying 7 results from an estimated 7 matches for "murbl".

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2023 Jun 19
1
Multiple phones on same PJSIP account
...xtension account (username/secret)? Yes. This is one of the major advantages to using PJSIP instead of chan_sip. (Other than the quality of the code and whether it's maintained.) Antony. -- "It wouldn't be a good idea to talk about him behind his back in front of him." - murble Please reply to the list; please *don't* CC me.
2020 Oct 25
0
chan_sip doesn't authenticate on INVITE from a Dial() command
...s now changed to "how can I get Asterisk to use the credentials and successfully authenticate, then dial on to the number I need?" > Thanks for any help :) > > > Antony. -- "It wouldn't be a good idea to talk about him behind his back in front of him." - murble Please reply to the list; please *don't* CC me.
2023 Jun 19
2
Multiple phones on same PJSIP account
I am creating a dialplan where a single user (Alice) has two offices. Both of her phones should ring if her extension is called. I could use a ring group, but I'm wondering can both phones use the same PJSIP extension account (username/secret)? Thanks Brian (ast newb) -------------- next part -------------- An HTML attachment was scrubbed... URL:
2023 Jun 19
1
Multiple phones on same PJSIP account
...extension account (username/secret)? Yes. This is one of the major advantages to using PJSIP instead of chan_sip. (Other than the quality of the code and whether it's maintained.) Antony. -- "It wouldn't be a good idea to talk about him behind his back in front of him." - murble Please reply to the list; please *don't* CC me. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital....
2020 Oct 25
2
chan_sip doesn't authenticate on INVITE from a Dial() command
Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or
1998 Jun 16
7
Ethernet card addr <-> IP
-----BEGIN PGP SIGNED MESSAGE----- Hi everyone - Someone I''m working with has a requirement to map ethernet card addresses to unique IP addresses, and then have a Linux IP masquerade server know of this mapping list and not allow any data to pass from any ethernet card that a) it doesn''t know about, or b) isn''t assigned the right IP. Ideally it would also log this
2019 Feb 27
5
Asterisk - can't hear other side. Or other side does not hear us
Hello, This is not technical post, just looking for suggestions on what to check.I have asterisk for long time, no updates, just maintain OS updates. I use SPA504G phones Very rarely and randomly when we pickup a phone - other side does not hear us. Call them back and all works. Now I have couple people I'm talking to and it seems like very call like this. Someone can't hear someone.