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Thread: host=pbxes.org or wwwx.pbxes.com
art

Replies: 0
Views: 8134

host=pbxes.org or wwwx.pbxes.com 07.06.2013 14:13 Forum: Terminal Equipment

I have always registered all my devices to pbxes.org
A few days ago I lost registration (only from my freepbx trying to register to a subPBX on pbxes.org).
Searching in the forum I found a thread (on trixbox) which suggested registering to www2.pbxes.com (in my case that would be www3 but that's just detail). When I changed the "host=" entry and the registration string as suggested, my freepbx successfully registered.
I'm worried that this method might not always work since pbxes.org should redirect to the appropriate host.

Please clarify or point me to any documentation that does.

Thread: RE: Receive "please enter extension id" when trying to setup a mobile extension
art

Replies: 4
Views: 9167

RE: Receive "please enter extension id" when trying to setup a mobile extension 05.06.2013 18:48 Forum: Miscellaneous

{embarrassed} I was inserting the accountID prior to the extension number.

It's been a while since I've setup most of the extensions but I thought I had to do that (I know I have to do it on the extensions themselves).

Apologies for the confusion.

Thread: RE: Can't call Bouygues Telecom voicemail - "blacklisted by frederic"
art

Replies: 2
Views: 9007

RE: Can't call Bouygues Telecom voicemail - "blacklisted by frederic" 04.06.2013 21:53 Forum: Miscellaneous

Not sure how the forum can help. I just dialled that number and immediately got an answer.
Admittedly, I'm not using Google voice/or talk or anything.. just using a regular service provider.
Try visiting your log.. setting up a trace.. then dialling the number. You may get some clue.

Thread: RE: The user you are trying to use is unavailable
art

Replies: 3
Views: 8527

RE: The user you are trying to use is unavailable 04.06.2013 19:41 Forum: Miscellaneous

What "says" you can have two simultaneous calls? Do you mean your service provider?

Thread: RE: Receive "please enter extension id" when trying to setup a mobile extension
art

Replies: 4
Views: 9167

RE: Receive "please enter extension id" when trying to setup a mobile extension 04.06.2013 19:31 Forum: Miscellaneous

Originally was using Chrome. I tried I-Explorer with the same result! I've run out of browsers on PCs - Next, I tried creating the extension using iPad (Safari) again, it's the same result.
I should inform that prior to requesting help, I tried selecting personal data and just clicking "Submit and Start".

Thread: RE: Receive "please enter extension id" when trying to setup a mobile extension
art

Replies: 4
Views: 9167

Receive "please enter extension id" when trying to setup a mobile extension 03.06.2013 21:01 Forum: Miscellaneous

I have only a few mobile extensions.. today (and yesterday) I've been trying to add a mobile extension in place of a sip extension..
So I deleted the sip extension..
then tried to insert an entry for the mobile extension.

I receive "please enter extension id" - ??

ofcourse I've input an extension id.. so I'm confused. Now with the others (regular sip extensions that I should make mobile extensions) I'm afraid of deleting them and not being able to recreate them.

Thread: Asterisk subtrunk
art

Replies: 0
Views: 6719

Asterisk subtrunk 05.05.2013 22:45 Forum: Terminal Equipment

I do not know what has caused this, I was unaware that it was the case:

When I register to pbxes.org it resolves to 188.40.65.170 (www1.pbxes.com).

When I dialout from a pbxes extension to an extension on my local FreePBX, it comes from 88.198.69.250 (www3.pbxes.com) and that is refused with a:

Failed to authenticate device "home office" <sip:210@88.198.69.250:27658>;tag=as7fcc54f9

This also occurs if an inbound call on a pbxes trunk is directed to my FreePBX.

Is this new behaviour? if so, is there a solution?

your help is appreciated.

Thread: Elsewhere - posted a problem and it was resolved
art

Replies: 0
Views: 3258

Elsewhere - posted a problem and it was resolved 10.02.2013 17:43 Forum: Bugs

I get a message that I cannot post unless I'm a "paid" account (which I am), and I'm logged in (which again I am) and it shouldn't be pbxes PRO (the forum) which it wasn't..
But I could post that the problem had been solved.

I also see that all the forums are greyed out (except Bugs) - so I'm posting here.

But, wait a minute, this must be a bug as well !!? :-(

Thread: sub-trunk not receiving
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Replies: 0
Views: 3184

sub-trunk not receiving 09.02.2013 20:45 Forum: Miscellaneous

Need to decipher a log entry that's stopping communications:

Feb 9 22:33:50 VERBOSE[109272] logger.c: Capabilities: us - 0x161e (gsm|ulaw|alaw|g726|speex|ilbc|g722), peer - audio=0x11d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x1c (ulaw|alaw|g726)
Feb 9 22:33:50 VERBOSE[109272] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Feb 9 22:33:50 VERBOSE[87148] chan_sip.c: Hangup call SIP/artarzi-1236700-b018, SIP callid 5352f4ad6b3db497201b5c8264566b45@10.0.1.80
Feb 9 22:33:55 VERBOSE[86697] chan_sip.c: Hangup call SIP/artarzi-24028959-0e22, SIP callid 71e9a8305832265718feff776f1a373c@88.198.69.250
Feb 9 22:33:55 VERBOSE[86697] chan_sip.c: Hangup call SIP/artarzi-110-c343, SIP callid ifCJcFfSoK2.iMydLwgDFc13.Mu6jVIj

This call is received from an ATA (spa3000) and will never be able to do video. I'm interpreting this as a refusal of the call because there's no video codec in common. Am I mistaken? if so, what is the reason the call is not passing through?

Thread: new FreePBX - unable to call out (resolved)
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Replies: 0
Views: 7205

new FreePBX - unable to call out (resolved) 14.05.2012 00:39 Forum: Terminal Equipment

On the trunk CID Options forced Trunk CID and inserted Outbound CallerID as trunk name. This may not be the most elegant solution but it works.
I am still unable to receive calls into the local PBX

Have upgraded my (ancient) Trxibox to FreePBX Distro. Now trying to port over services. Did not make any changes to PBXES.
When I call out the call is redirected to the default inbound route (of PBXES)

Did a SIP trace and extracted the "area" of the redirection as:


May 14 02:05:53 VERBOSE[69298] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
May 14 02:05:53 VERBOSE[69298] logger.c: Looking for 00202YYYYYYYY in from-pstn (domain pbxes.org)
May 14 02:05:53 VERBOSE[69298] logger.c: list_route: hop: <sip:204@77.69.186.124:5060>
May 14 02:05:53 VERBOSE[69298] logger.c: Transmitting (NAT)
May 14 02:05:53 VERBOSE[119579] logger.c: -- Called 0097339XXXXXX@from-internal/n
May 14 02:05:53 VERBOSE[119614] logger.c: We're at 88.198.69.250 port 40218
May 14 02:05:53 VERBOSE[119614] logger.c: Video is at 88.198.69.250 port 41916
May 14 02:05:53 VERBOSE[119614] logger.c: Adding codec 0x1000 (g722) to SDP
May 14 02:05:53 VERBOSE[119614] logger.c: Adding codec 0x4 (ulaw) to SDP
May 14 02:05:53 VERBOSE[119614] logger.c: Adding codec 0x8 (alaw) to SDP
May 14 02:05:53 VERBOSE[119614] logger.c: Adding codec 0x10 (g726) to SDP
May 14 02:05:53 VERBOSE[119614] logger.c: Adding codec 0x400 (ilbc) to SDP
May 14 02:05:53 VERBOSE[119614] logger.c: Adding codec 0x200 (speex) to SDP
May 14 02:05:53 VERBOSE[119614] logger.c: Adding codec 0x2 (gsm) to SDP
May 14 02:05:53 VERBOSE[119614] logger.c: 13 headers, 14 lines
May 14 02:05:53 VERBOSE[119614] logger.c: Reliably Transmitting (NAT)
May 14 02:05:53 VERBOSE[119614] logger.c: -- Called 13338700/39XXXXXX
May 14 02:05:53 VERBOSE[69298] logger.c:
May 14 02:05:53 VERBOSE[69298] logger.c: --- (17 headers 0 lines)May 14 02:05:53 VERBOSE[69298] logger.c: --- (17 headers 0 lines)---

Where:
00202YYYYYYYY is the number I'm trying to call
sip:204@77.69.186.124:5060 is the extension (local)
39XXXXXX is the number call is redirected to

Warning: I'm also facing this problem with another service (i.e. I KNOW this problem is on my side, but I don't know how to solve it)

Thread: RE: Sub PBXes
art

Replies: 21
Views: 166979

RE: Sub PBXes 26.12.2011 17:38 Forum: News

I do not see any of my sub-PBX on the status screen. Could you please help with that? also will they ever show on the call monitor as Trunk?

Thread: RE: Problem logging in
art

Replies: 2
Views: 6944

RE: Problem logging in 05.06.2009 23:27 Forum: Bugs

Thank you, I stand corrected. Seems I was too engrossed and mistakenly inserted something in one of the password fields.. then the system had to have all password fields filled (as it should) ..

Thread: pbxes.org DNS record (resolved)
art

Replies: 0
Views: 5444

pbxes.org DNS record (resolved) 03.06.2009 21:30 Forum: Bugs

I was registering to pbxes.org and calls were not being accepted lately. When I changed to 78.46.102.143 calls are now accepted. This is from my Asterisk server.
pbxes.org resolves to 213.133.110.43 using my ISP's DNS. I am also registering a standalone telephone with no problems (only using port 80) from within the same network.

For now I have changed Asterisk to the fixed IP but could you check this please?

Resolution:
Two devices using the same port don't seem to work behind one nat device (at least the one I have).

By using the different IP addresses (one resolved by DNS and one fixed) I was able to temporarily circumvent that. Now my Asterisk is registering (and normally operating) using pbxes.org as the host.

Thread: RE: Problem logging in
art

Replies: 2
Views: 6944

Achtung Problem logging in 03.06.2009 10:35 Forum: Bugs

Not really bug but major problem.

I'm unable to login. Also some of my phones not able to register (did not check all phones).
account is artarzi and all devices configured to register with pbxes.org (not a static IP).
Would appreciate help

update:
I had restarted my "pbx" using personal data which apparently needed me to enter a "new" password. I did, then I reverted to the old one. The second step did not happen. Good thing I used a password that I remembered. So now I'm good.
Now for a question: Do I have to change my password every time I need to restart the pbx?

Thread: RE: Presence and Location of Mobile Extensions
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Replies: 16
Views: 59650

RE: Presence and Location of Mobile Extensions 16.10.2008 22:45 Forum: News

Small questions:
First
Which "alias" should one use ? My account is artarzi.. the extension I want to use Locator for is 101.. and I've assigned the web page pbxes.org/Abdo to it..
So do I use pbxes.org/Abdo or pbxes.org/artarzi OR (which I need to assume otherwise how can I use more than one "alias") do I use pbxes.org/artarzi-101 ??

The application (Locator) works with the www.viking.com/gpsloc.php but not when I set it to any of the above.

Second
Where can I set the parameters: to disable callback or magnify the map or whatever... Is that in the GPS Locator application sending the location (doesn't seem to make sense) or somewhere in the pbxes.org/Abdo statement in the extension setting.. ??

Thread: RE: Registration expiry 20 secs is that normal?
art

Replies: 1
Views: 7995

Registration expiry 20 secs is that normal? 31.08.2008 00:01 Forum: Terminal Equipment

In the system log I receive
Aug 30 20:56:30 VERBOSE[23425] logger.c: -- Registered SIP 'artarzi-200' expires 20

This is an Asterisk server registering as an extension (well, a trunk on my server) and I do not understand (or have control) over the expiry of registrations..

Thread: RE: Source Code
art

Replies: 1
Views: 16478

RE: Source Code 30.08.2008 19:00 Forum: News

You guys rock !!

If ever you wish to setup a server somewhere in the Middle East, I live in Bahrain (where VoIP is legal) and would love to cooperate with you.
Between you and fring.. if it weren't for needing distribution at home (with a bit more control) I would have no need for my Asterisk server.

Thread: RE: Incoming calls not passing
art

Replies: 2
Views: 9384

RE: Incoming calls not passing 04.08.2008 08:04 Forum: Providers

It's working now!!
I don't think it's anything I did (didn't expect a reaction from their support on a Sunday) but I'll check over the next few days just to make sure.

Thread: RE: Incoming calls not passing
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Replies: 2
Views: 9384

Incoming calls not passing 03.08.2008 20:46 Forum: Providers

I've been using sipphone as my outgoing and incoming provider (for most international calls).
Everything was working fine. Recently, I added a "feature" in sipphone (or Gizmorpoject) of having a DID with a 775 area code.
I signed up for the service.. and then tested it (worked fine) ..

Recently (frankly I was only made aware today) I am unable to receive incoming calls from sipphone.. I've checked the log and the call does come through (at least from the American number) but pbxes does not pass it through.

If you could check this, I would appreciate it.

user is artarzi
incoming call shows several numbers calling through a +1775 area code which is the incoming number. The context appears as from-pstn (something or other) which is different to the normal.

I'm sure that sipphone have changed something (or else this would not have happened) but I do not understand why the call is not delivered.

Thread:
art

Replies: 9
Views: 18572

24.06.2008 21:43 Forum: Terminal Equipment

Thank you for the explanation. Mine is a C450 IP and it is both a PSTN and an IP phone. I must admit, it is not the best phone I have but it does serve a purpose. I was quite glad when it worked with pbxes on the first try. Am also glad that yours work too.

I still don't understand why the same software (firmware) does not work the same way but I've seen that on the Sipuras before. I cannot dedicate any time to trying to perfect the way the Siemens works because they do not give enough of an interface to manage the device. It is unfortunate because the concept is great but I would never try to standardize on them Siemens devices (SIP performance cannot be made optimal).

Showing posts 1 to 20 of 36 results Pages (2): [1] 2 next »

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