 Thread: RE: Did not work / Offline ? |
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After switching from www3 to Server7 it was OK for 1-2 hours.
I saw now, that the config cannot be loaded also on www7. I switched to www1. It works hopefully.
Automatic exchange of servers if one fails would be appreciated at premium accounts.
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| Thread: RE: DTMF does not work |
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Hallo,
DTMF seems not to work... Our Sipgate-Trunk has the setting "auto". Each digit is twice forwarded.
Could you check it?
We have the problem for one week. It is the same both from my mobile (Nokia E72) and a normal DECT-Desktop-Phone.
Because the number is not recognised tries the system to get it called via Easybell.
Thank you for your help.
Best Regard
P.
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Jul 13 09:47:29 VERBOSE[28891] logger.c: We're at 88.198.69.250 port 40936
Jul 13 09:47:29 VERBOSE[28891] logger.c: Video is at 88.198.69.250 port 38562
Jul 13 09:47:29 VERBOSE[28891] logger.c: Adding codec 0x8 (alaw) to SDP
Jul 13 09:47:29 VERBOSE[28891] logger.c: Adding codec 0x4 (ulaw) to SDP
Jul 13 09:47:29 VERBOSE[28891] logger.c: Adding codec 0x10 (g726) to SDP
Jul 13 09:47:29 VERBOSE[28891] logger.c: Adding codec 0x400 (ilbc) to SDP
Jul 13 09:47:29 VERBOSE[28891] logger.c: Adding codec 0x200 (speex) to SDP
Jul 13 09:47:29 VERBOSE[28891] logger.c: Adding codec 0x2 (gsm) to SDP
Jul 13 09:47:29 VERBOSE[28891] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jul 13 09:47:29 VERBOSE[28891] logger.c: -- Called Easybell/000044991779551XXXXXXXXXX
Jul 13 09:47:29 VERBOSE[107012] chan_sip.c: SIP response 100 to standard invite
Jul 13 09:47:29 VERBOSE[107012] chan_sip.c: SIP response 100 to standard invite
Jul 13 09:47:29 VERBOSE[107012] chan_sip.c: SIP response 407 to standard invite
Jul 13 09:47:29 VERBOSE[107012] logger.c: We're at 88.198.69.250 port 40936
Jul 13 09:47:29 VERBOSE[107012] logger.c: Video is at 88.198.69.250 port 38562
Jul 13 09:47:29 VERBOSE[107012] logger.c: Adding codec 0x8 (alaw) to SDP
Jul 13 09:47:29 VERBOSE[107012] logger.c: Adding codec 0x4 (ulaw) to SDP
Jul 13 09:47:29 VERBOSE[107012] logger.c: Adding codec 0x10 (g726) to SDP
Jul 13 09:47:29 VERBOSE[107012] logger.c: Adding codec 0x400 (ilbc) to SDP
Jul 13 09:47:29 VERBOSE[107012] logger.c: Adding codec 0x200 (speex) to SDP
Jul 13 09:47:29 VERBOSE[107012] logger.c: Adding codec 0x2 (gsm) to SDP
Jul 13 09:47:29 VERBOSE[107012] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jul 13 09:47:29 VERBOSE[107012] chan_sip.c: SIP response 100 to standard invite
Jul 13 09:47:29 VERBOSE[107012] chan_sip.c: SIP response 100 to standard invite
Jul 13 09:47:29 VERBOSE[107012] chan_sip.c: SIP response 403 to standard invite
Jul 13 09:47:29 VERBOSE[28891] logger.c: -- SIP/Easybell-822c is circuit-busy
Jul 13 09:47:29 VERBOSE[28891] chan_sip.c: Hangup call SIP/Easybell-822c, SIP callid 6ce12885681b859f3da955b44a165490@sip.easybell.de
Jul 13 09:47:29 VERBOSE[28891] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
Jul 13 09:47:29 VERBOSE[27744] logger.c: -- Local/0000449917795XXXXXXXXX@from-internal/n-7a6f,1 answered SIP/29010XXXXX-1790
Jul 13 09:47:29 VERBOSE[28891] logger.c: -- Playing 'cannot-complete-as-dialed' (language 'de')
Jul 13 09:47:32 VERBOSE[28891] logger.c: -- Playing 'pls-try-call-later' (language 'de')
Jul 13 09:47:36 VERBOSE[27744] chan_sip.c: Hangup call SIP/29010XXXX-1790, SIP callid 534adba579c76d0f0b8b143111f4aa21@sipgate.de
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| Thread: RE: Sending recorder calls via e-mails seems not to work |
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Hi,
I have activated the option that the recorded call via mail will be sent. I have also set up the mail adress. It did not worked even though I tested it with two mail adresses.
I was able to download the file thru the webinterface neither
Thank you for checking the issue.
I have used the option, that the call will be recorded if I use "*1"...
P.
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| Thread: Backup-Account for some additional Fee |
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Hi,
I would appreciate, if PBXes would change automatically the account onto an additional one (backup-account) being hosted on an other server. I would mean some additional security.
It could be a mirrored account with the same settings. I can imagine to pay 50% for this additional account. It wouöld worth it.
Thank you.
P.
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| Thread: Reporting the problems |
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Hi,
I need sometimes the following features:
1. If some of my friends are not accessable via VoIP. (Router must be restarted etc.) it would be good if they would be informed via e-mail about this fact. It would allow them the check the devices locally. I do have neither the time nor the physical access to check them.
2. It would be also good if I - as the administrator - would receive a report on a daily basis if everything is Ok.
-- How many clients are logged in
-- was ist necessary to use a backup trunk
-- How many call were failed
etc.
It would allow me e.g. to check the prepaid trunks if money is OK.
because of the system of backup-trunks I am not informed by the user if anythink went wrong. They can call 99,9% of the time.
Thank you very much.
P.
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| Thread: RE: Sub PBXes |
pbx
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21 |
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199689 |
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Hi,
we have an AVM 7270 as PBX in the office.
We have 8 Devices (6 DECT and 2 Analog Lines)... The six DECT-Phones are registered on the AVM as VoIP-Phones and use our PBXes-System directly.
What would be the advantage If I registered the Fritzbox as SubPBX of PBXes? Also now are the phones registered - thru the VoIP-Login of the Fritzbox . in the PBXes-System.
Could you tell me some advantages where this feature can be used for?
Thank you very much.
P.
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| Thread: RE: PBXese down - SOHO account |
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There are for some days outages at PBXes. Today morning it was pretty chaotic.
Just now it seems to be ok... (www3, Amsterdam)
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| Thread: RE: PBXes failes and must be regularly restarted |
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The servers are really not in sync... (Forumpostings, Logs etc)
After some minutes the system do work no longer. It must be restarted again and again.
Edit: now it works for some minutes on www3 (america / virgin)... I have SIP-Trace activated... I will send it to PBXes if any failures occur.
11:47 /CET/ Edit: 75% of extensions is not registered (shown in Fritzbox)...
P.
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| Thread: RE: PBXes failes and must be regularly restarted |
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The same here today (actually once eaxh day...)
Fritzbox shows the the VoIP Extensions are not connected correctly. Sometimes is only a part of the handsets (DECT) connected.
PBXes Status shows no problem...
After switching from once server to other the call-logs have disappeared and we received the following error if we wanted to open status :
"Error loading configuration file variables.txt?aldope=373632
We are currently on WWW4.
I.
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| Thread: RE: PBXes failes and must be regularly restarted |
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The same here on www1 today morning.
Once a day must the system be saved and restarted... Without it the "status" shows everything would be OK, but it actually does not work.
After restart it works again...
Our DECT-clients are loggen in vie a Fritzbox 7270...
Gr. I.
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| Thread: RE: Backup-Trunks: how do they work? |
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Hello,
I have experienced many times that the backup trunks do not work...
I have two trunks:
1. european mobiles calls --> sipservice.de
2. everything else --> sipgate.de (also as backup for sipservice)
I would think that a call would be forwarded to sipgate if sipservice does not work, is deactivated or send a error back (e. g. resetCDR) to pbxes....
How must they be configured to achieve this effect?
Sipservice is sometimes blocked because their firewalls closes. (Hacking attack) In this case Pbxes sees the trunk, everything seems to be OK, the calls are however not forwarded to the called person.
Is it actually the way how the backup-trunks of Pbxes should work? Should Pbxes automatically use the backup-trunk in this case? (ResetCDR is sent back...)
Thank you for confirming the way how the backup trunks should really work.
Regards,
I.
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| Thread: RE: PBXes failes and must be regularly restarted |
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Hi,
yesterday and today we had regularly the problem that PBXes cannot be connected correctly. It has to be restarted twice a day was absolutely horrible because a lot of people use the system.
After "save and restart" was it OK again...
Could you check it?
Thank you..
Regards,
I.
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