Thread: RE: Limit outbound routes? |
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Hmmm. By dummy trunk, I guess you mean either a SIP trunk with no account, or with an invalid account that doesn't really go anywhere..?
Also, is there a way to allow calling local extensions as well, whether they are classic/PSTN numbers or SIP extensions? So the whole thing would be "This extension can call US numbers or local extensions (even if they happen to be via international PSTN)".
Thanks so much!
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Thread: RE: Limit outbound routes? |
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Hi,
Let's say extension 1234 can only call US numbers. I'd setup an outbound route for +1, 001, 0111 for ext 1234, then another "catchall" outbound route for it with no trunks defined. Would that work?
Thanks!
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Thread: RE: Screechy Digital Receptionist ring indicator |
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Hi all,
Whenever I enter an extension number to call within a Digital Receptionist menu, I get a harsh/screechy ring indicator (the extension does ring). I've read the other threads re indicator.conf but it's not even giving the nice US ringtone.
I've tried logging into the extension using Sipdroid and Groundwire, same thing. Any ideas?
Thanks!
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Thread: Option to show passwords |
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Hello,
The passwords are now shown as 3 asterisks. Is there some way to show them again over SSL?
Thanks!
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Thread: RE: Unable to remove Google Voice trunk |
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Hi,
I deleted my Google Voice trunk -- it's no longer on the Trunks list -- but when I call the GV number, I see in the Call Log that pbxes is still receiving the call, so it's still registered to GV. I already restarted via Personal Data.
Please check. Thanks!
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I can't reply to this message, so I'm editing in a quick note that this problem seems to have fixed itself after a few hours. Probably after some sort of registration timeout.
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Thread: RE: Already exists fail on DB |
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I'm also experiencing this problem... my Sub PBX entry disappeared.
Note that when I manually changed from www6 to www4, the entry reappeared. @dragonhills, are you also on www6? Perhaps that server is having problems again.
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Thread: RE: White label/API |
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SOAP would be nice, although a simple POST interface would probably be good enough. Actually anything over standard HTTP(S) would be cool.
It might be easier to just support virtual hosts (allowing multiple entries per reseller, e.g. client1.example.com, client2.example.com) and use the templates of the reseller account.
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Thread: RE: White label/API |
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I wish there was some way we could sell Premium services under our own brand/website, i.e. provisioning, configuration, maintenance, monitoring and "Status" access via an API. Any plans..?
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Thread: RE: Way more than fall-over to mobile phone |
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Ah, no wonder it works great for me... I do have a WiFi+GSM phone.
(OT) Hmmm. Maybe that's why I've been having problems with hunt groups. I guess I have to recheck how total ring time for the group is affected by the "Number of seconds to ring phones before sending callers to voicemail" General Setting...
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Thread: RE: pfingo incoming calls |
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Ok, I almost fell off my chair when I saw sip.pfingo.com in the list of recommended inbound providers. Very funny, guys. :-P
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Thread: RE: Not reachable by ENUM from SIPBroker, etc. |
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Hello,
1. Map a DID to a *non*-PBXes account at e.g. e164.org
2. Call via SIPBroker ENUM (e.g. **275*013... from FWD) - Call succeeds
3. Call via OpenSER ENUM - Call succeeds
4. Change the mapping to a PBXes account
5. Call via SIPBroker ENUM - Call fails
6. Call via OpenSER ENUM - Call fails
If you change it back, calls succeed again.
Apparently, PBXes is giving a 404 to the INVITE, even though the address in the SIP Request-Line is correct, and can be reached directly by a SIP phone.
If anyone at PBXes is interested in the Wireshark captures of the OpenSER sessions, please email me. This might be applicable to a lot of other ENUM-enabled switches trying to connect to PBXes accounts.
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Thread: RE: pfingo incoming calls |
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Arrrgh. This is absolutely frustrating. How come my itsy-bitsy Trixbox 2.6.1 running in VMware Player (in a Linux box, behind a DSL router with dynamic IP address) can handle incoming pfingo calls perfectly and even forward it to PBXes with no problems whatsoever, but PBXes in all its glory... *can't*?
No special setup:
Trunk name: pfingo
Peer details:
username=3xxxxxxx
type=peer
secret=yyyyyy
insecure=very
host=sip.pfingo.com
Register string:
3xxxxxxx:yyyyyy@sip.pfingo.com/3xxxxxxx
Don't get me wrong, I really like PBXes. That's why this thing is bugging me like crazy (I have a paid account at pfingo, btw).
If I get paid support, can this bug get squished once and for all..?
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Thread: Problems calling .org but not .com |
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I'm getting errors calling (from another SIP service, e.g. FWD) my username(-ext)@pbxes.org address ("403 Forbidden" or "407 Proxy Authentication Required"). If I call username(-ext)@pbxes.com, then there's no problem.
I guess I could use .com instead, but I've specified .org in a number of places and don't want to have to change it everywhere. Is there some maintenance etc. going on?
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Thread: RE: Webcall can't handle "+" prefix |
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Hi,
Thanks! It's not quite done though... the outgoing number is missing the + sign, e.g. I enter +63291xxxxx, but PBXes attempts to route 63291xxxxx, so it fails.
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Thread: RE: Webcall can't handle "+" prefix |
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The webcall form converts the + sign (i.e. international prefix) into %2B. Although the correct caller id shows up when the extension rings, the target phone obviously does not ring.
Some browsers e.g. Konqueror may have rendering problems (somehow) with this %2B-prefixed caller ID, and end up with a truncated Call Monitor list.
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Thread: RE: Weird routing on secondary call |
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I rewrote the rules to not have an outgoing trunk prefix and used these instead:
Trunk:
0063+0|N.
00++|Z.
Outgoing route: (starts with)
+63
0063
0N
Seems to work fine.
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