Thread: RE: Calls to SIP uri's (another extension) end up with 603 Declined |
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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
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Thread: RE: stuck on www7 after "Network Outage" May 12th; can't return to www2. |
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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
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Thread: RE: dialing SIP destination via specific trunk |
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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
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Thread: RE: dialing SIP destination via specific trunk |
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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
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Thread: RE: digital receptionist -> callthru |
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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
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Thread: RE: digital receptionist -> callthru |
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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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