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Thread: RE: blacklisted by dumbura
ayk

Replies: 1
Views: 6878

blacklisted by dumbura 04.01.2011 10:39 Forum: Bugs

Hi,

I have this Premium account, as well as another premium account ayking2 as backup. However, recently I've been getting this "blacklisted by dumbura" error messages when trying to make calls. I have no idea what this dumbura is, and judging from previous threads it's supposed to be duplicate account from the same user. That's not my account. Can someone please kindly check and rectify?

Thanks in advance!

EDIT: It's been quite a while and this situation has not been fixed yet. Does this mean I have to pay support to fix something that I've paid for and that's supposed to work?

Thread: HKBN 2b Call Drop with SIP Error 481
ayk

Replies: 0
Views: 8117

HKBN 2b Call Drop with SIP Error 481 20.01.2010 00:40 Forum: Bugs

I am opening a new thread in Bugs because I believe this is a bug in the software.

Zitat:
Jan 18 16:23:45 VERBOSE[18359] chan_sip.c: SIP response 200 to standard invite
Jan 18 16:32:10 VERBOSE[18359] logger.c: -- Got SIP response 481 "No matching call entry found"


Zitat:
Jan 19 16:41:30 VERBOSE[25036] chan_sip.c: SIP response 200 to standard invite
Jan 19 16:51:28 VERBOSE[25036] logger.c: -- Got SIP response 481 "No matching call entry found"


Original thread that cannot be replied to:
http://www4.pbxes.com/forum/thread.php?threadid=1779

The problem is that the call will get dropped after approximately 7-10 minutes with a SIP Response 481. This problem is back sometime late last year and after trying various methods including getting a PRO account and modify the source, Perhaps when Asterisk was upgraded the patch was overwritten?

Original Digium patch:
https://issues.asterisk.org/view.php?id=5747

New Digium patch that *may* be related since then:
http://svnview.digium.com/svn/asterisk?r...9&view=revision

Thread: RE: voip.ms: call forwarding?
ayk

Replies: 7
Views: 32806

RE: voip.ms: call forwarding? 19.01.2010 23:17 Forum: Providers

I got it from this forum as well, from none other than bobmats himself! großes Grinsen Both of us need to thank him...haha

http://www4.pbxes.com/forum/thread.php?threadid=166

Thread: RE: voip.ms: call forwarding?
ayk

Replies: 7
Views: 32806

RE: voip.ms: call forwarding? 14.01.2010 19:45 Forum: Providers

Nope, it's the same voip.ms subaccount.

Trunk1: voip.ms
username: voip.ms username
password: voip.ms password
server: sip??.us.voip.ms
register: yes

Trunk2: voip.ms CID
username: <blank>
password: password:username
server: same server as trunk1
register: no

Thread: RE: hkbn garbled voice
ayk

Replies: 5
Views: 17408

RE: hkbn garbled voice 11.01.2010 23:09 Forum: Providers

Hi Mconvey,

Are you still having this problem? Also, do your calls get disconnected after approx. 7-10 minutes?

Thanks.

Thread: RE: voip.ms: call forwarding?
ayk

Replies: 7
Views: 32806

RE: voip.ms: call forwarding? 11.01.2010 23:05 Forum: Providers

I use Voip.ms for call forwarding and they work with forwarding CID.

Make sure you have one trunk with the same username/password combination registered on the same voip.ms server. Then the username blank and password = password:username trick works with a trunk that's not set to register.

Hope this helps.

Thread: RE: Voip.ms Toronto POP ip change
ayk

Replies: 1
Views: 10464

Voip.ms Toronto POP ip change 26.11.2009 18:08 Forum: Providers

Hi,

Since voip.ms changed their IP address for the Toronto POP, I haven't been able to use my phones. The phones work if I put in IP address directly but fail if I put in hostname.

Change is as follows:
code:

=====================
What are the changes?
=====================

The IP address for the host sip.ca2.voip.ms and iax2.ca2.voip.ms will be replaced November, Wednesday 25th @ 2:00 AM EST with the following IP address:


NEW IP Address: 174.137.63.202
OLD IP Address: 24.102.60.67


Please kindly check DNS cache.

Thanks.

ayking

Thread: RE: Trunks "Registration failed - timeout"
ayk

Replies: 1
Views: 10211

Trunks "Registration failed - timeout" 16.07.2009 22:14 Forum: Bugs

Almost all of my trunks have this in the logs:
Registration for "trunk" timed out, trying again (Attempt #??)

All these trunks were working prior to July 15 morning EDT. Now incoming calls to these trunks from pstn may sporadically connect but usually cannot connect. Seems like the connections would go offline for most of the time, then reconnect for a while and then dead again.

I have a second account on www2 that is still working fine with the same providers.

Please kindly help. This seems to be a different problems than what others are experiencing in the other threads.

Thanks a lot.

Thread: RE: can't receive calls
ayk

Replies: 13
Views: 32779

RE: can't receive calls 15.07.2009 22:59 Forum: Bugs

ALL of my trunks are sporadically and randomly offline. They would register for a while, then it would get "Registration Time-out". This has been happening on me and a friend's account on the NY server the whole day today.

On the other hand, another account on the San Diego server seems to be fine, as long as I define the server as www2.pbxes.com instead of pbxes.org.

Pascal: I'm wondering if it's something to do with the NY network rather than the NY server. No modification was done on my account today before it went off, and if you say the server is fine, then the repeating but random errors *may* point to some routing issues. Just a suggestion.

Thread: RE: can't receive calls
ayk

Replies: 13
Views: 32779

RE: can't receive calls 15.07.2009 19:47 Forum: Bugs

Mine is on NY. It worked for about 15 minutes and now it just went offline again.

Thread: RE: can't receive calls
ayk

Replies: 13
Views: 32779

RE: can't receive calls 15.07.2009 17:33 Forum: Bugs

I have the same problem today. Which server are you on?

Thread: RE: Customization - Blocking outbound routes by time
ayk

Replies: 13
Views: 35771

RE: Customization - Blocking outbound routes by time 12.11.2008 23:52 Forum: PBXes PRO

Just like to chime in as well.

I too have requested before that premium users be granted permission to modify source. For me, as I do not need the CRM and billing at all, it does not make sense to pay more just to modify the source a few times. CRM and billing features are something PRO users use constantly and thus see their value, but I wouldn't be modifying the source code every few days.

If as pascal mentioned that account plans cannot be mixed on extension basis, then I can't have just 1 extension go PRO and the rest use premium. If I split to two accounts, then I'll still be JUST able to modify the source in the account with 1 extension, but not for the other extensions. That's not going to do me much good.

I am a single home user with 20 extensions defined (some SIP and some classic) because I use them for special spped dialout, conferences or call forwarding. I'm willing to pay for premium because I like the idea of able to set things up to my liking, and allow powerful features like callthru, callback and followme. Even if I slimmed down in terms of number of extensions, it'll cost me 40 euro a month instead of 10 and it doesn't make sense at all for home users.

If you are firm about not including the source modify function to premium accounts, how about charge a very small extra amount for this specific feature on a PER-ACCOUNT basis?

Thread: Dialing out in conference
ayk

Replies: 0
Views: 7372

Dialing out in conference 30.10.2008 00:19 Forum: Feature Requests

Hi,

Currently, this is what happen when a digit is dialed within a conference call:

Zitat:
The conference may be exited by pressing a digit. The extension number of the digit gets called.


However, it means if one of the participant has an IVR system, as soon as I press a digit the call will attempt to go to this extension, and the conference is dropped. Is it possible to have the dial-out capability also contained within the "*" menu? (e.g. * to access menu, then 9 to exit and dial-tone)

P.S. I found this problem when attempting this: http://www1.pbxes.com/forum/thread.php?t...1207855091&sid=, but I assume there'll be other times when this may be useful.

Thread: RE: Callthru Calls Recording
ayk

Replies: 10
Views: 27019

RE: Callthru Calls Recording 30.10.2008 00:13 Forum: Miscellaneous

Thanks. The callthru using loopback extension with recording on now works.

Is it true that a call-thru call does not recognize special codes such as "*1" for recording, or "##" for transfer?

Thread: RE: Callthru Calls Recording
ayk

Replies: 10
Views: 27019

RE: Callthru Calls Recording 27.10.2008 21:54 Forum: Miscellaneous

I tried this as well, but was not successful.

1. Callthru using a loopback extension
- Loopback was detected by system and the loop was disconnected.

Zitat:

Oct 27 15:17:17 VERBOSE[3786] logger.c: -- Called Loopback_URI
Oct 27 15:17:17 VERBOSE[20014] logger.c: -- Got SIP response 482 "Loop Detected"
Oct 27 15:17:17 VERBOSE[3792] chan_sip.c: Hangup call SIP/xxxxx-53bb, SIP callid 46200a15295b72264fd39a5e696b4015@64.118.93.76
Oct 27 15:17:17 VERBOSE[3786] logger.c: -- Now forwarding SIP/xxxxxxxxxxx-1f37 to 'Local/xxxxxxxx@from-pstn' (thanks to SIP/pbxes.org-9977)
Oct 27 15:17:17 VERBOSE[3786] chan_sip.c: Hangup call SIP/pbxes.org-9977, SIP callid 63a723cf7c087171162bbb4d76c1ee43@64.118.93.76


2. Using conference to call one of the recording extension
- anytime a digit is pressed in a conference, the conference is dropped and goes to that dialed extension. However, this means I can NOT dial digits to the receiving end's IVR system.
- i.e. Only works for non-IVR phone system on the other side

At the end, I had to create a specific classic extension to call this certain company, and enable recording on the extension. This obviously has 2 limitations: a) it can only be outgoing call, and b) it can only be this company.

Are there any other options/suggestions? Thanks a lot!

Thread: RE: Sort Inbound Routes
ayk

Replies: 5
Views: 15188

RE: Sort Inbound Routes 09.10.2008 09:36 Forum: Feature Requests

Thanks! Now it's much easier to look at them!

Thread: RE: Sort Inbound Routes
ayk

Replies: 5
Views: 15188

RE: Sort Inbound Routes 08.10.2008 23:06 Forum: Feature Requests

Could this be done for trunks too? Thanks!

Thread: RE: HKBN 2b disconnect problem - with link to Digium fix
ayk

Replies: 2
Views: 14566

RE: HKBN 2b disconnect problem - with link to Digium fix 08.10.2008 01:22 Forum: Providers

The disconnection is back, at least on the New York server. Here is the log for a test call to HK MTR IVR line:

Oct 7 19:06:58 VERBOSE[8770] logger.c: Capabilities: us - 0x41e (gsm|ulaw|alaw|g726|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Oct 7 19:06:58 VERBOSE[8770] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Oct 7 19:06:58 VERBOSE[19675] logger.c: -- SIP/2b-fa73 answered Local/01185228818888@from-internal/n-c233,2
Oct 7 19:06:58 VERBOSE[19648] logger.c: -- Local/01185228818888@from-internal/n-c233,1 answered SIP/username-fcf2
Oct 7 19:17:23 VERBOSE[19675] chan_sip.c: Hangup call SIP/2b-fa73, SIP callid 4114b57c4175b64e6e60ae6958a46cce@s2hkbntel.net
Oct 7 19:17:23 VERBOSE[19648] chan_sip.c: Hangup call SIP/username-fcf2, SIP callid 4041333-3432409594-811353@msw-01.voip.primus.ca
Oct 7 19:17:24 VERBOSE[8770] logger.c: -- Got SIP response 481 "No matching call entry found"

[EDIT]
Weird. Today it's fine, with 2 calls one at 18 minutes and the other at 6 minutes. Thanks.

[EDIT2]
Today I have this problem again, disconnect after 7 minutes with the same SIP response 481 error. Argh!!

Thread: RE: Per-trunk codec setting
ayk

Replies: 3
Views: 12227

RE: Per-trunk codec setting 22.09.2008 22:23 Forum: Feature Requests

I understand that it's possible for PBXes PRO user, but if I'm not running a call centre or using CRM, then a premium account is all I need (especially for a home user).

Can the admin please confirm if the 14-day trial method would work? Is there any chance that the source will be open for edit for premium users as well?

Thanks a lot.

Thread: Longer field length for "Outbound caller ID"
ayk

Replies: 0
Views: 7269

Longer field length for "Outbound caller ID" 16.09.2008 00:36 Forum: Feature Requests

Hi,

I would like to know if it's possible to extend the length of the "Outbound Caller ID" field in the trunk setting page. The longer length is used for some logic in setting the outbound callerID being forwarded to mobile phones according to the incoming callerID/trunk.

Thanks a lot.

[Edit]
To illustrate what I'm trying to achieve, this is what I would like to put into one of my call forwarding trunk:
{DOLLAR}${CALLERID(NAME)}<1${CALLERID(NUM):1:9}>

Showing posts 1 to 20 of 54 results Pages (3): [1] 2 3 next »

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