 Thread: RE: Account blocked |
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Thanks again for the agility. I'll remember it for the next time.
This no refund policy is a drawback of the service, which is otherwise very good.
Best regards.
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| Thread: RE: Account blocked |
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Thank you very much for the swift reply, i-p-tel.
I wasn't aware of this limitation, so I canceled the subscription to avoid unnecessary charges in the forthcoming month.
So, it brings me to 2 questions:
1 - is it safe now to cancel the subscription made today? Or should I wait till the end of the month?
2 - Is it possible for you to refund one of these payments?
Best regards.
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| Thread: RE: Account blocked |
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For the record, I am the reseller of cacaushow account.
The first two subscriptions were bought using the discounted reseller rate. The third one I bought directly under cacaushow's account (no reseller discount).
Could you please check why the second subscription was not counted to his monthly minute allowance?
In any case, I would like to be refunded for the extra subscription.
Thanks for the concern.
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| Thread: RE: What´s happening with www4 Server ? |
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Tijuca,
I am so sorry to hear it.
The last www4 issue reported was this:
[10-25-2010 20:04:01] SERVICE ALERT: www4.pbxes.com;SIP test;CRITICAL;SOFT;3;SIP timeout: No response from SIP server after 15 seconds
(You can see it here https://www1.pbxes.com/status/)
My account was fully restored that day right after the first successful ping.
Sometimes, as we can see on the forum, the system does not recover itself automatically. In these cases we need the moderators' help (as far as I know, only i-p-tel and Diafora have this status).
If the problem is a trunk outage, I spent 7 days without access to my main trunk last month (see: http://www1.pbxes.com/forum/thread.php?t...1283187314&sid=). The final solution, after the system was restored, was to un-register and re-register the trunk (as can be seen in the topic). Although I am not sure about the cause OR the solution.
Abraços de Niterói,
Marcel
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| Thread: RE: What´s happening with www4 Server ? |
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www4 was really having some problems (as the Nagios monitoring showed).
However, it is running fine since Monday for me. Have you restarted your account since then?
Regards
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| Thread: RE: DID mismatch in PBXes |
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Moderators,
the DID provider asked us to verify if PBXes is accepting incoming calls from IP 204.11.194.38
Also, since cacaushow seems to be the only account affected (marcelcalil is using another DID from the same company, and it is working properly), they did not reject the hypothesis of misconfiguration at PBXes' side.
Could you please take a look?
Thank you
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| Thread: RE: DID mismatch in PBXes |
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Temporary solution: cacaushow's DID is now pointing to my account, and I created a rule which redirects all calls back to his account.
Dirt and poor solution, but it works. Since I am cacaushow's PBXes reseller, I felt like I should do something.
Those out-of-nowhere errors, however, are mining my confidence in the service.
UPDATE
service restored on the same day, close to midnight (local time).
No idea, however, on what caused the disruption or what solved it.
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| Thread: RE: Trunk sequence problem |
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I am facing a similar problem.
There are 2 trunks in my primary route, and the first one is skipped time and again.
First trunk is a VoipSwitch reseller, second trunk is a Betamax Clone (Rynga).
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| Thread: RE: change audio message on digital receptionist |
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Fantini,
first of all you must upload a recorded message in the System Recordings.
After you upload it, go to the Digital Receptionist and either create a new menu or change an existing one. The recording uploaded will be shown in a drop down menu.
Regards
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| Thread: RE: SIP Trunk Outage |
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Thanks for the reply, Diafora.
There is one advantage of keeping trunks registered: they appear in Status Panel. ; )
Anyway, all is working fine now.
Regards
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| Thread: RE: SIP Trunk Outage |
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i-p-tel,
Thanks for the reply.
Yes, you are half-right. Extension 1000 seems to be with this callshop problem. Strange thing is that I've cancelled the Pro account several months ago, and only now the extension stopped working.
However, my real problem is the trunk outage. My SIP trunk is shown as time-out in System Log and grayed-out in Status screen.
It has been offline for a complete week now.
As a reminder, it connects fine with an ATA or softphone, so the trunk is not really in an outage.
Regards
UPDATE
the following set of actions fixed the problem:
Go to Trunk -> Account -> Register
Set to No (outbound calls only)
Submit and Apply
Make a testcall (failed)
Go to Trunk -> Account -> Register (again)
Set to Yes (inbound and outbound calls)
Submit and Apply
make a testcall (which finally worked)
Prior to that, I have tried to change trunk name, language, DTMF mode, username and password. I even bought a new trunk account from the same voipswitch reseller and registered it in my PBXes account. All these trials resulted in failure; unregistering and re-registering the trunk was the solution.
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| Thread: RE: SIP Trunk Outage |
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Ron,
any help is appreciated.
Last Friday PBXes operation was full of errors (my DID was connected to the Echo Test, for example). The were all solved, except the trunk problem. So, I strongly believe this is a system error.
Audio bypass is off.
Yes, I had only one trunk and only one outbound route when the issue started. The simplest possible scenario.
The Betamax problem is mainly related to the Freedays (calls not charged during a given period of time), not to the connection itself. As all PBXes extensions use the same IP (for each www zone), Betamax considers it as the same user, and charges for all calls - what is fine for me, as it is still very cheap for my country.
The reason I am asking for other users to look at it is because sometimes two accounts can be bound to the same problem (check this post to understand what I'm talking about : http://www1.pbxes.com/forum/thread.php?t...1281973198&sid=).
Anyway, thank you very much for your concern.
Regards,
Marcel
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| Thread: RE: call forward from one extension to another |
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Example on call forwarding:
Create extension NN1
Create extension NN2
Make sure they both work (make and receive a couple testcalls)
Now, if you want NN2 to receive calls when NN1 is unavailable (or busy, or ringing too much):
Go to extension NN1
Choose the proper forwarding field (all, busy, no answer, etc)
Enter only the extension number (in this example, NN2)
Click Submit and Apply (red bar)
Now calls to NN1 are forwarded to NN2
......................
About the "Dial" field in extensions, it is used for inbound routing.
Example: if you want your extensions to ring from 7 am to 23 pm only, you can do it using Inbound Routing.
Step-by-step:
Go to your extension (say NN1)
In "Dial" field, replace SIP/username-NN1 by SIP/username-anynameyoulike@pbxes.org
Submit and Apply
Go to Inbound Routing
Creat route username-anynameyoulike (important, it must be the same name used on the previous step)
On Regular Hours Field, click Extension (so calls during regular hours will ring your extension)
On Special Services, leave blank (for now)
On After Hours, click Voicemail or Hang Up (so calls placed after hours won`t ring your phone at all).
On Regular Hours, write 7:00-23:00 (this is the time of the day you are willing to take calls, according to the example)
On Days, write which days of the week you want these rules to apply (since this information was omitted in the example, write *)
Select No Override
Submit and Apply
Other very useful features of Inbound Routing are Callback and Callthru. Take some further reading on these features on the forum and, if it is still not clear, you may call me at SIP:marcelcalil@pbxes.org (the prompt is in Portuguese, but please press 1 and leave a message on my voicemail).
Hope it helps.
Best Regards.
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| Thread: RE: How To Change a number from what is dialed to what is sent to trunk |
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Interesting...
Is there any way to implement digit replacement as well?
For example, the main trunk uses syntax
0ZZNXXXXXXX
(where ZZ is the area code and NXXXXXXX is the subscriber number)
While the backup uses
00XXZZNXXXXXXX
(where XX is the country code, and the other numbers follow the previous example)
The idea is to have a rule applied to the back-up trunk which strips the first digit and adds 00XX (replaces the first zero by double-zero and country code), so that a number dialed as:
0ZZNXXXXXXX
would have the 0 stripped, becoming:
ZZNXXXXXXX
and then added 00XX, resulting in:
00XXZZNXXXXXXX
This way, both trunks would accept the same input from users.
Using rule 00XX+0|ZXNXXXXXXX does not work.
This is not a must-have feature, since there are ways to bypass it (creating rules on both the trunks and the outbound routes, for example). But it would make life simpler, and the simpler the more error-free it is.
Regards.
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| Thread: RE: Inbound Calls notworking |
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It happened to me too.
Inbound calls were being sent to echo test last Friday. Today I called a customer's DID (in Brazil) from a cell phone (which should be directed to his Digital Receptionist) and some random guy in USA picked up the call!
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| Thread: RE: SIP Trunk Outage |
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Hello,
despite the problems described in http://pbxes.org/forum/thread.php?threadid=1282873282, I am having a trunk outage.
I've read this post (http://www1.pbxes.com/forum/thread.php?threadid=144&sid=), where luxapo reports the same problem I am experiencing (no connection from PBXes, normal connection with ATA and softphones).
The solution presented to luxapo, however, doesn't apply to me (my trunk config is already as described in the topic). Also, I don't remember changing anything.
Has anyone else had issues with trunks?
UPDATE: (Aug, 31)
I registered a second trunk (Betamax clone Rynga). It is shown as active in Status screen, but calls are not routed to it (yes, I have added the trunk to the outbound route).
Outbound calls do not appear in Call Monitor.
When I try to make a call:
Aug 31 09:02:20 VERBOSE[49149] logger.c: -- Playing 'not-enough-credit' (language 'pt_BR')
Aug 31 09:02:23 VERBOSE[49149] chan_sip.c: Hangup call SIP/marcelcalil-1000-7650, SIP callid fd0f33426e2a511e@TkJfTUNOT0dVRUlSQS5pdGF1dGVjLmludHJhbmV0
Aug 31 09:02:31 VERBOSE[52626] logger.c: Ignoring this INVITE request
Aug 31 09:02:31 VERBOSE[52626] chan_sip.c: Got a SIP re-invite for call 3901d058c5098351@TkJfTUNOT0dVRUlSQS5pdGF1dGVjLmludHJhbmV0
Aug 31 09:02:32 VERBOSE[52626] logger.c: Ignoring this INVITE request
Aug 31 09:02:32 VERBOSE[52626] chan_sip.c: Got a SIP re-invite for call 3901d058c5098351@TkJfTUNOT0dVRUlSQS5pdGF1dGVjLmludHJhbmV0
What System Log says about the main trunk:
Aug 31 08:55:36 NOTICE[52626] chan_sip.c: -- Registration for '7003001@187.45.221.164' timed out, trying again (Attempt #1284)
Aug 31 08:56:26 NOTICE[42685] cdr.c: CDR simple logging enabled.
Aug 31 08:56:26 VERBOSE[42685] logger.c: == MySQL RealTime reloaded.
Aug 31 08:56:26 NOTICE[42685] indications.c: Removed default indication country 'us'
Aug 31 08:56:26 VERBOSE[42685] logger.c: == Extension state: Watcher for hint 1000 deactivated. Notify User marcelcalil-1000
Aug 31 08:56:26 NOTICE[42685] cdr_addon_mysql.c: MySQL database table not specified. Assuming "cdr"
Other messages from the log:
Aug 31 09:27:22 NOTICE[52626] chan_sip.c: Unknown SIP command 'PUBLISH' from '67.231.245.210'
Trunks are offline for 4 days now!
PLEASE HELP!!!
UPDATE (Sep, 1)
Trunks offline for 5 days now.
I used to read forum topics and think that people exaggerated the moderator's unresponsiveness. Now it feels I was too complacent.
UPDATE (Sep, 2)
Trunks offline for 6 days now.
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