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Thread: RE: blocked account
rda

Replies: 2
Views: 7187

RE: blocked account 08.12.2016 14:20 Forum: Bugs

I am sorry but I am confused.

I don't think I have any other account, with the exception of the one I created this morning trying to put things at work and the one that you blocked.

This rises my serious security concerns.

Could you please send me the list of "my" accounts?

Feel free to use my e-mail address corresponding to my paid account.
Thank you

EDIT
On a second thought, please remove ALL "my" free accounts with the exceptions of my ONLY paid account. .. so that my paid account will become eligible in future to become the ONLY free account.

I would do it myself, but I do not have knowledge of other free accounts under my name that made my "rdarioc" account not more suitable to become a free account.

Thank you

Thread: RE: blocked account
rda

Replies: 2
Views: 7187

blocked account 08.12.2016 10:38 Forum: Bugs

Last night my account was blocked (instead of downgraded from SOHO to FREE).

I had no way to contact you to inform of the inconvenience.

I paid again for the SOHO subscription but I am not going to use those additional services.

Is there a way next month I won't get my account blocked again and simply downgraded to a free account?

In the meanwhile there are other two issues:

(1) how can I download the credentials of the trunks I am using?
(2) While the account was blocked I tried to create another FREE account BUT every time I was trying to connect some trunks (with correct credentials) it kept answering me with a pop up error: This trunk is being used by <account-blocked>

The trunks were actually available because I connected to them separately without issues.

THank you for your support

Thread: RE: Missing/Inconsistent log
rda

Replies: 4
Views: 10766

RE: Missing/Inconsistent log 02.10.2016 05:29 Forum: Bugs

Thank you. I saw the other thread. I still wonder how is it possible that my log registered my extension 100 as source of the call, while the other guy, Tom, got a log with "sipvicious" as source of the call.

Apart from that, yes, I also think that was the problem.

Was the source still the IP mentioned in my log? It belongs to an hosting company in Germany. Have you notified them that some of their clients ran this sort of attack toward your server and people lost money because of it?

Yes, reconfiguring the firewall to block that IP is one option, but won't last long. How about supporting TLS to connect to your server? VPN would not work, because I have clients on my cellphone and while those clients should use the VPN, the phone should still be able to access other services outside your VPN.

Thank you for in advance for your reply.

Thread: RE: Missing/Inconsistent log
rda

Replies: 4
Views: 10766

RE: Missing/Inconsistent log 01.10.2016 13:34 Forum: Bugs

Perhaps I am misunderstanding something:

Allow me to summarize my understanding on how a webcall works on your service:

1. someone gets to the following web address:

www.pbxes.org/<username>

Where <username> was the field defined by me in one of my extensions configuration page and

2. this someone fills-in the field available on such page for a "call-back" and

3. pbxes.org first start the call toward the above "call-back" number and upon successful connection, only then,

4. pbxes.org starts the call toward the extension corresponding to the <username> as referred in the above step #1.

Now, if this is what should happen when a webcall is triggered, I have a couple of doubts:

A. The call was directed toward a ring-group which does not have any <username> configured and reachable from the web. Yes, the extension 302 has a <username> that could have triggered such webcall, but as you can see from the below log, the call went toward a ring-group (which included extension 302)


2016-10-01 02:14:27 +49xxxxxxxx +49xxxxxxxx ­ VoipJumperDE from-internal-cont Dial 00:00:00
2016-10-01 02:14:22 "R.D. Contarino" <100> 302 ­ from-internal-cont Dial 00:00:00
2016-10-01 02:13:42 "R.D. Contarino" <100> 1 89.163.­242.161 ext-group Hangup


and

B. Where can I find the number that this someone put in the field mentioned in the above step #2 that could have, upon successful connection, triggered the web call? In my experience, but I could be wrong, the webcall is activated ONLY after a successful connection to such number, but I have no record of it.

I am still concerned about some hacking issues. I hope you will help me to clarify those doubts.

Thank you.

p.s. I noted that now on my Call Monitor, I read the IP address you wrote (see my above cut&paste), while tonight the field was filled in with its FQDN. How come that tonight I see (and cut&paste) one thing while now I see (and cut&paste) another thing? Did you perhaps change my log?

Thread: RE: Missing/Inconsistent log
rda

Replies: 4
Views: 10766

Missing/Inconsistent log 01.10.2016 07:01 Forum: Bugs

Hi there,

my system log reports the following lines:

Sep 30 23:57:26 VERBOSE[103123] logger.c: -- Registered SIP 'rdarioc-200' expires 1800
Oct 1 02:14:22 VERBOSE[8028] logger.c: -- Called 302@from-internal/n


while my Call Monitor reports the following lines:

2016-10-01 02:13:42 "R.D. Contarino" <100> 1 sa413.saturn.f­astwebserver.de ext-group Hangup (00:04:27)


In other words there are 40 seconds of System Log missing and I am even unable to find out who called me in the middle of the night.

What worries me is that I cannot even track if my account has been hacked or not.

Any suggestion?

Thread: Trunk name for Inbound routing
rda

Replies: 0
Views: 6027

Trunk name for Inbound routing 03.03.2016 20:47 Forum: Miscellaneous

Dear all,

I have a north american DID provider which allows me to receive call on its registered number and route onto a voip client of my choice, as long as I register it using user-id, pwd and their provided sip register.

Similarly, I have just received a FritzBox 7412 from 1und1 which works as sip server and allows me to register a softphone to receive and place calls using their service.

In both cases I register both accounts as trunks and I am normally able to receive calls either from the north american DID number or calls directed to my German number received with the 1und1 subscription.

Unfortunately I am unable to trap those calls in the inbound routing process as I am unable to identify the name of the trunk.

I tried either with the name that I associate to the trunk when I create it or the name of the user-id used to register to those servers.

Just to give an idea, every time I receive a call on my 1und1 number I get a log similar to:

Mar 3 20:33:53 VERBOSE[62491] chan_sip.c: Hangup call SIP/620-XXXX, SIP callid C9525D99E2B6E792@aa.bbb.ccc.ddd

Where XXXX changes at every call and aaa.bbb.ccc.ddd is the public IP of my 1und1 router which I conveniently hidden for privacy.

I cannot configure the north american DID service to automatically route all the calls to my SIP URI address, nor I can configure the FritzBox 7412 to route all the calls to my SIP URI address.

How can I trap all the calls coming through those trunks so that I can use the incoming route mechanism?

Thank you in advance for any help

Thread: RE: Multiple DID on the same trunk (MyDivert)
rda

Replies: 3
Views: 17817

RE: Multiple DID on the same trunk (MyDivert) 24.03.2015 08:32 Forum: Providers

I believe what VOI means is the following:

* * *

1. Check with your trunk provider IF he allows
multiple log-in with same credentials (or even with
different credentials) and IF he can associate
different DID to different log-in sessions

2. Register several trunks on PBXes, with the same provider
one trunk per DID. Use different trunk names on PBXes

3. Check if the in-bound trunk mechanism on PBXes
recognize the incoming trunk based on the name
you provided at above step #2. If so
then you can now set up your in-bound rule based
on the trunk name which will allow you to
distinguish between different DIDs

I had the same problem and, to handle it, since my DID
provider did not allow the option I described at above step 1, I had to create several accounts, thus several trunks, one per DID, to solve this issue.

Therefore, it is not possible to create an inbound rule, one per DID, as VOI says, but it is possible to create an inbound rule, one per TRUNK, and if you associate only one DID to each TRUNK then you can handle this.

Hope this helps.

Thread: Wrong Trunk Number Shown on Status
rda

Replies: 0
Views: 4890

Wrong Trunk Number Shown on Status 20.11.2014 10:59 Forum: Bugs

Dear Sirs,

I have an account with a soho subscription.

this soho subscription is currently active BUT few months ago it was not as for a number of months I did not use it.

I just noted in the Status page, even though I have enabled more than 5 Trunks, only the first 5 are shown.

The unshown trunk is working anyway.

Thank you for your attention.

Best regards
Rosario

Thread: CSIPSimple/ZRTP support and PBXES
rda

Replies: 0
Views: 12850

CSIPSimple/ZRTP support and PBXES 03.02.2014 15:54 Forum: Terminal Equipment

Dear all,

the subject says it all:

Do you know if CSIPSimple on Android is able to establish ZRTP/SRTP secure calls between two PBXes users?

Thank you in advance.
Rosario

Thread: RE: Partial CID match in Inbound routes
rda

Replies: 14
Views: 49162

RE: Partial CID match in Inbound routes 14.06.2013 08:09 Forum: Feature Requests

Dear I-P-Tel,

I am having troubles on setting up inbound rules intercepting CIDs.

What is the valid syntax acceptable in that specific field?

Thank you in advance.

Thread: RE: Voicetrading - Outbound trunk issue
rda

Replies: 53
Views: 167554

RE: Voicetrading - Outbound trunk issue 07.04.2013 08:31 Forum: Providers

Dear all,

I have been recently experiencing again the same annoying problem with Betamax trunks:

Got SIP response 500 "Internal server error"

Now, it is evident for some reasons they blocked some IP and I am now trying to get them unblocked again.

This is, in my humble opinion, an issue that should be addressed sooner or later in a more effective and definitive way.

Assuming we cannot grant all people to always "behave" properly, we cannot rely on a solution that purely assumes some users will not try to abuse the offered services.

That being said the only way a traffic provider has to double check the authority of a user to spend his credit is either with his credentials but also with his source IP.

Now, couldn't be possible for you guys to offer, as a separate option, the possibility to have a unique static IP assigned to every customers of yours who expressly requests it?

I don't mean it for free but paying.

Besides now it may also be time for you to consider the idea of routing your traffic back and forth to Betamax servers through a private peering, thus sparing a lot of Internet bandwith, increasing your client's perceived quality and you could even use private IP classes for the internal routing, so that you should not even ask RIPE to assign you any new IP class at all.

In other words: you could reduce your costs (peering with Betamax), increase your revenue (fee for each private IP) and increase the perceived quality of your service from your clients.

Any other suggestion or idea you may think could work better with less effort for you please let me know and I will be more than happy to try to convince Dellmont/Betamax of that new possiblity.

Thank you in advance for your reply.
Best regards
Rosario

Thread: Sip Uri Call From Pbxes To Pbxes Issues
rda

Replies: 0
Views: 4725

Sip Uri Call From Pbxes To Pbxes Issues 29.11.2012 19:04 Forum: Bugs

Dear all,

I have a friend of mine with an account over here.

Randomly, when he tries to call me using <username>@pbxes.org

Where username is my account name of course

his log correctly reports an outbound call with destination-id as <username>

On my side, I receive a call from <my-friend-username> but destination ID is "s".

What does destination "s" mean?

Anyway I don't get any call and he gets an automatic message telling that the called destination is invalid.

Any help on how to fix it?

Best regards
Rosario

Thread: RE: PBXes PRO -- Multichannel Customer Interaction
rda

Replies: 4
Views: 39573

RE: PBXes PRO -- Multichannel Customer Interaction 04.12.2011 18:48 Forum: News

I understand, but I need to discuss about your service in person as I am integrating a solution carrier grade for over 10k users and I cannot manage this size of project just online.

I am willing to come visiting your offices or premises at your headquarter but I need first to arrange a meeting in person.

Is that possible?

If you want you can send me contact details in private at rdcontarino@gmail.com

If you need to keep details confidential I am also willing to sign a mutual NDA.

Looking forward for your kind reply.
Best regards
Rosario

p.s. I must say when I saw on GoogleMaps your spot as an empty lot, it didn't give me much confidence then fortunately German Yellow pages have an updated image. Anyway I need to discuss in person the details of our project, hopefully under NDA and a forum or a paid-online assistance service is definitely not the appropriate place.

Thread: RE: PBXes PRO -- Multichannel Customer Interaction
rda

Replies: 4
Views: 39573

RE: PBXes PRO -- Multichannel Customer Interaction 04.12.2011 08:42 Forum: News

Dear Sirs,

Is there any way to contact you directly to ask for a sales offer/business opportunity other than driving to your office in Germany? I am in Frankfurt but I would like at least show up with an appointment.

Best regards
Rosario D. Contarino

Thread: RE: Google Voice Peering
rda

Replies: 206
Views: 1018537

RE: Google Voice Peering 03.10.2011 06:58 Forum: News

Hi there.. checked right now. Inbound is working .. outbound actually never did.

But, hold on, when I talk about outbound I mean, through pbxes calling a googletalk account, NOT using GV trunk to make PSTN calls.

Have you ever be able to call googletlak accounts from PBXes ?

As well, when I talk abount INBOUND I mean I use another googletalk account to call the googletalk used in PBXes to register its trunk. And it works, at least now.

Best
Rosario

Thread: RE: SMS <> Landlines
rda

Replies: 2
Views: 11799

RE: SMS <> Landlines 30.09.2011 10:04 Forum: Miscellaneous

Hello Bazmercer,

I was looking for something similar. I don' t know if you already solved the issue but, from what I can understand of, it seems to me there are a number of issues that need to be addressed:

Sending SMS.
From what I have understood from your message you're using a betamax clone.
Therefore you send your SMS using their software application which by the way allows you to set up sending phone number (among those validated by you).

In this respect it could be nice if you could do this through pbxes, but that's another story.

Receiving SMS
If someone sends you back an SMS at the number which was previously set by betamax the first thing is: who provided you that number?

In case it is a virtual DID number provide by a SIP operator, the thing is that service has to include also any kind of SMS-to-something gateway otherwise you are not getting back anything.

Then, assuming your DID number service provider gives you such kind of gateway, again it may be nice if that kind of gateway could include also conversion into SIP protocol so that it could eventually be understood by pbxes and then eventually forwarded to a SIP client, but this is science fiction as for now.

Anyway, have you solved your issue in some ways?

Best
Dario

Thread: O2 Germany Mobile closes VoIP SIP access
rda

Replies: 0
Views: 10172

O2 Germany Mobile closes VoIP SIP access 15.09.2011 07:28 Forum: Providers

Dear all,

I have been experiencing this issue since weeks now.

I own a German O2 prepaid card on which I have been used to enable a monthly internet flat option.

This till a few weeks ago when SIP service is no more working. At least I can place calls but I cannot receive any, although my client apparently properly register itself on PBXes switch.

Unfortunately I have been able to try only with applications running on my Nokia N70 (Nimbuzz, Fring) which do not allow to change standard SIP port.

Nonetheless O2 Germany proudly announced this agreement:

http://www.telephonyworld.com/news/telef...-voip-services/

In which evidently they now changed their policy.

If anybody could check with different devices or software that could be great, otherwise as they are still claiming to offer VoIP access spread the voice as it is untrue.

Best
Rosario

p.s. XMPP seems still working, but I haven't found a way to route calls from PBXes switch to any of my google or yahoo accounts. If anybody has any viable solution I'll really appreciate any feedback.

Thread: RE: Google Talk issues: when I receive a gchat, call drops
rda

Replies: 1
Views: 7451

RE: Google Talk issues: when I receive a gchat, call drops 12.09.2011 17:02 Forum: Bugs

hello there,

I'm actually stuck before that... have you been able to call a gmail account from your pbxes ?

after setting up a google talk trunk I'm capable to receive calls on my gmail account, then routed to my pbxes extension but every time I try to make a call from my pbxes to something like name@gmail.com ... I always receive a google message "your call cannot be completed" or something like that.. any idea?

Best
Rosario

Thread: RE: Google Voice Peering
rda

Replies: 206
Views: 1018537

RE: Google Voice Peering 12.09.2011 08:05 Forum: News

Hi there,

I've just set up a GTALK trunk with my @gmail.com account.

Inbound works fine when trying to call my <username>@gmail.com

Outbound doesn't. I receive a voice message saying "We cannot complete your call, please try again". While from the log I get:

Gtalk/+<username>@voice.google.com-f512 answered SIP/<username>-199-f646

So apparently the outbound trunk is selected properly but then "+<username>@voice.google.com" is dialed instead of "<username>@gmail.com"

Is that a bug or a feature?

Any idea? Thanks in advance for your help

Thread: RE: NIMBUZZ SIP client doesn't ring after passing from free to paid account
rda

Replies: 2
Views: 12948

RE: NIMBUZZ SIP client doesn't ring after passing from free to paid account 12.09.2011 06:45 Forum: Terminal Equipment

it seems like neither Nimbuzz nor Fring (on Nokia N70) support different SIP port (In the meanwhile I've also contacted their customer service, just in case).

Are you aware of any other client running on Nokia N70 (Symbian S60 rev 2) which allows me to set up a different port to try and see if it solves the issue?

Best
Rosario

FURTHER UPDATE
I have had no chances to find a VOIP client which lets me set up the server port other than 5060 for SIP support on my Nokia N70

At the same time I found out that actually other VOIP clients based on XMPP seems to still work fine

Is there a way from PBXES to reach an XMPP client? (either GTALK or anyother?)

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