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Thread: Native SIP client removed from Android 12?
sia

Replies: 1
Views: 796

Native SIP client removed from Android 12? 16.04.2022 02:35 Forum: Terminal Equipment

It appears native SIP client has being removed from Android 12:
https://www.xda-developers.com/android-1...ve-sip-calling/
https://android-review.googlesource.com/...3//COMMIT_MSG#9

What's current recommend Android SIP client for use with PBXes?
I've used Sipdroid in the past, but would like to check alternative options.

In the same thread, any recommendations for iOS client?
In general, i prefer simple, reliable and battery friendly applications vs full featured and complex.

Thanks!

--igor

Thread: RE: See extension status from website
sia

Replies: 12
Views: 26807

RE: See extension status from website 01.04.2017 21:13 Forum: Feature Requests

http:/pbxes.org/alias?action=state does not work for me, resulting in 404.

The WebCall page is valid, and works with no parameters or with ?action=video

p.s. I did try using specific http://wwwN.pbxes.org/ as well

--igor

Thread: RE: Calls to SIP uri's (another extension) end up with 603 Declined
sia

Replies: 2
Views: 6704

RE: Calls to SIP uri's (another extension) end up with 603 Declined 12.11.2016 20:46 Forum: Bugs

Well, in the past (since June until ~4 weeks ago) I'd no problems calling username-xxx@pbxes.com from another extension (Android native SIP); but in the last 3 weeks I'm getting 603 Declined to such calls from multiple devices to multiple extensions. This what i mean by bug :-(

Thread: RE: Calls to SIP uri's (another extension) end up with 603 Declined
sia

Replies: 2
Views: 6704

traurig Calls to SIP uri's (another extension) end up with 603 Declined 07.11.2016 13:13 Forum: Bugs

I'm trying to call from one extension to another via SIP URI, within same PBX, i.e. sip:username-NNN@pbxes.org. I do have per-extension incoming routes setup for username-NNN, routing calls properly.

This did work until about 2 weeks ago, but now i'm receiving "603 Declined". Same with the call to sip:username@pbxes.org (or sip:username@pbxes.com), which ended up in the auto-attendant before. Tried restarting PBX via "Submit & Start", no change.Tried varying clients - no difference, this appears to be server-side issue.

Thanks!

p.s. Just found approach which works - calling NNN@username.pbxes.org... Not sure if this legitimate way (i think there is a single wild-carded IP behind it), but at least it works.

--igor

Thread: RE: stuck on www7 after "Network Outage" May 12th; can't return to www2.
sia

Replies: 2
Views: 5621

RE: stuck on www7 after "Network Outage" May 12th; can't return to www2. 21.05.2013 21:45 Forum: Bugs

Confirmed, thanks!

Thread: RE: stuck on www7 after "Network Outage" May 12th; can't return to www2.
sia

Replies: 2
Views: 5621

stuck on www7 after "Network Outage" May 12th; can't return to www2. 21.05.2013 05:51 Forum: Bugs

My pbx is normally on www2 (Seattle), Pacific time zone (premium account).

After outage on May 12th it was transferred to www7 (Miami) and stayed there since. Since then i've experienced occasional issues with trunks and extensions.
At least one of my trunks uses IP range blocking and cannot talk to www7 (yet).
Also most my extensions are in California.

When i try to force switch back to www2 i get
"Please wait while your account is being transferred to another server." message, but still stay on www7, even after logout/login.

Please advise. Thanks!

--igor

Thread: RE: dialing SIP destination via specific trunk
sia

Replies: 3
Views: 14473

RE: dialing SIP destination via specific trunk 05.09.2009 20:52 Forum: Feature Requests

That's correct, usually SIP URI's are globally reachable - but it's not always the case.

I'm trying to use SIP <-> Skype gateway, available to Sipnet.ru (Tario) customers. It's possible to call skypeusername@skype.sipnet.ru (or even i believe skypeusername@skype.com) via Sipnet. Yes, it does require SIP registration; and i do not see how i can make PBX to route this URI via Sipnet trunk.

I guess if Sipnet would be my default trunk it may work; i'd test it; but this is not setup i can really use (it would require setting up full routing of numerical prefixes first, leaving Sipnet as last resort default). [LATER] I've tried using Sipnet as default trunk, but PBXes still attempts to reach SIP URI directly - as it should in general case.

I think i'm probably asking to add some support for SIP URI in dial patterns for outbound routing. It can be just regexp, or may be special magic wild-card meaning "matches any SIP URI".

Ideally it should be possible to route different SIP URI's into different trunks, but even global default route for SIP URI's would be helpful for now.

Thanks!

--igor

Thread: RE: dialing SIP destination via specific trunk
sia

Replies: 3
Views: 14473

dialing SIP destination via specific trunk 04.09.2009 22:52 Forum: Feature Requests

Hi,

I've a sip destination (user@domain.com) which is only reachable via specific trunk.

I know how to map a SIP extension to SIP destination; but how to make PBX route it via specified trunk? I do not belive outbound routing can be made applicable to SIP destinations.

Thanks!

--igor

Thread: RE: Status "Error loading configuration...."
sia

Replies: 4
Views: 10770

RE: Status "Error loading configuration...." 15.12.2008 23:47 Forum: Bugs

For me the Trunk part of Status window is empty now, both in the Firefox and Safari.
Extensions and Queues/Conferences are ok.

--igor

Thread: RE: digital receptionist -> callthru
sia

Replies: 8
Views: 21680

RE: digital receptionist -> callthru 21.11.2008 18:27 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

Thanks!

Tried that - and i'm getting reorder tone when calling this extension from PSTN or another extension:

Nov 21 09:02:05 VERBOSE[3668] logger.c: -- Called username-callthru
Nov 21 09:02:05 VERBOSE[6640] logger.c: -- Got SIP response 482 "Loop Detected"

Username above is my PBXes username; as i understand it get's routed to PBXes, then incoming route is chosen...

Aha, got it to work with dial set to SIP/username-callthru@pbxes.org.
pbxes.com works as well; if i remove it i'm getting "Loop Detected" again.

Thanks!

--igor

Thread: RE: digital receptionist -> callthru
sia

Replies: 8
Views: 21680

digital receptionist -> callthru 21.11.2008 03:31 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

I'd like to be able to have a call-thru digital receptionist option, i.e. digital receptionist menu option which would ask for pin and then provide a dialtone.

Currently you need a separate DID for this purpose, which does not always make sense.

Thanks!

--igor

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