Author |
Message |
Topic: Brekeke do not accept ACK message from Lancom |
lakeview
Replies: 4
Views: 4350
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Forum: Brekeke SIP Server Forum Posted: Wed Nov 29, 2017 12:10 pm Subject: Brekeke do not accept ACK message from Lancom |
> Route: <SIP:192.168.16.18:5060;lr>
The RFC3261 defines it as
SIP-URI = "sip:"
Try version 3 of Brekeke SIP Server because it accepts such ACK. |
Topic: Verification client certificate |
lakeview
Replies: 3
Views: 3317
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Forum: Brekeke SIP Server Forum Posted: Wed Nov 15, 2017 11:24 am Subject: Verification client certificate |
If you use a self-signed certificate as Client's certificate, you need to add it as a trusted certificate. Or set [Peer Certification Validation]=off at [Configuration] > [SIP] -> [TLS] page.
... |
Topic: Brekeke deletes last lines of INVITE message |
lakeview
Replies: 2
Views: 4297
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Forum: Brekeke SIP Server Forum Posted: Fri Nov 10, 2017 10:50 am Subject: Brekeke deletes last lines of INVITE message |
Hi loop,
Add the line below in the net.sip.packet.reader = false
The developer fixed this bug already so upgrading the SIP Server to the latest version will also fix the issue.
http://www.breke ... |
Topic: BSS VM Linux Template |
lakeview
Replies: 2
Views: 6537
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Forum: Brekeke SIP Server Forum Posted: Fri Jul 21, 2017 1:35 pm Subject: BSS VM Linux Template |
We have some experiences with Open vSwitch (OVS) on CentOS and Ubuntu for Brekeke SIP Server.
In the evaluation phase, we use KVM and VirtualBOX for simulating the above environments.
Are you i ... |
Topic: Websocket requests receiving 400 Bad Request based on size |
lakeview
Replies: 1
Views: 2686
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Forum: Brekeke SIP Server Forum Posted: Tue May 09, 2017 2:44 pm Subject: Websocket requests receiving 400 Bad Request based on size |
HTTP request's packet buffer size for WebSocket is 700 byte.
So resize the Cookie data to meet this limitation. |
Topic: Ringing Multiple Users from a Single Invite |
lakeview
Replies: 1
Views: 2556
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Forum: Brekeke SIP Server Forum Posted: Wed Jul 20, 2016 10:40 am Subject: Ringing Multiple Users from a Single Invite |
If you are using Brekeke SIP Server Advanced Edition, try DialPlan rule like this.
[Matching Patterns]
$request=^INVITE
To=sip:777@
[Deploy Patterns]
To = sip:30@192.168.10.222
$session = fa ... |
Topic: User export not working, just partial list (7 records) |
lakeview
Replies: 6
Views: 8446
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Forum: Brekeke SIP Server Forum Posted: Wed Dec 02, 2015 12:29 pm Subject: User export not working, just partial list (7 records) |
If you use another web browser, can you export the list? |
Topic: User export not working, just partial list (7 records) |
lakeview
Replies: 6
Views: 8446
|
Forum: Brekeke SIP Server Forum Posted: Wed Nov 25, 2015 11:30 am Subject: User export not working, just partial list (7 records) |
What kind of characters is it? |
Topic: Create CA certificate for TLS mode |
lakeview
Replies: 2
Views: 12921
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Forum: Brekeke SIP Server Forum Posted: Tue Nov 24, 2015 4:56 pm Subject: Create CA certificate for TLS mode |
Are you planning to use Self-signed certificate?
Have you looked at this page?
http://wiki.brekeke.com/wiki/create-a-self-signed-certificate-using-OpenSSL |
Topic: User export not working, just partial list (7 records) |
lakeview
Replies: 6
Views: 8446
|
Forum: Brekeke SIP Server Forum Posted: Tue Nov 24, 2015 4:52 pm Subject: User export not working, just partial list (7 records) |
Are you using any special characters in user names? |
Topic: Best way to do DID based sip routing for thousands of DIDs |
lakeview
Replies: 2
Views: 8589
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Forum: Brekeke SIP Server Forum Posted: Tue Sep 22, 2015 9:11 pm Subject: Best way to do DID based sip routing for thousands of DIDs |
It is an interesting.
I supposed Alias database meets the requirement.
You can add records with SQL from 3rd application, or CSV file.
http://wiki.brekeke.com/wiki/How-to-use-Alias-function
But ... |
Topic: Brekeke 3.4.1 Beta & WebRTC |
lakeview
Replies: 3
Views: 5737
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Forum: Brekeke SIP Server Forum Posted: Tue Oct 07, 2014 3:49 pm Subject: Brekeke 3.4.1 Beta & WebRTC |
> 1- Will Brekeke Advanced support WebRTC Video call ?
Yes.
If both caller and callee (UAC and UAS) support WbRTC, you can use any media such as audio and video between clients.
For JsSIP, e ... |
Topic: FYI:RFC6598 for Carrier Grade NAT(CGN)/Large-scale NAT(LSN) |
lakeview
Replies: 1
Views: 5926
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Forum: Brekeke SIP Server Forum Posted: Mon Aug 25, 2014 12:36 pm Subject: FYI:RFC6598 for Carrier Grade NAT(CGN)/Large-scale NAT(LSN) |
FYI:
Brekeke SIP Server version 3.4 and later support RFC6598 without any additional settings.
http://wiki.brekeke.com/wiki/BSS_History |
Topic: Does Brekeke SIP Server rev.286.3 Support RFC6598??? |
lakeview
Replies: 2
Views: 4649
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Forum: Brekeke SIP Server Forum Posted: Mon Aug 25, 2014 12:32 pm Subject: Does Brekeke SIP Server rev.286.3 Support RFC6598??? |
tjtsaitsai,
Why are you using such a very old version?
The rev 286.3 was released 4 years ago and has some issues.
> the configuration by adding net.internal-addr.pattern =
> ^100. ... |
Topic: "register.contact.remote" works correctly ? |
lakeview
Replies: 5
Views: 9195
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Forum: Brekeke SIP Server Forum Posted: Tue Jun 17, 2014 4:25 pm Subject: "register.contact.remote" works correctly ? |
Brekeke's developer team will add new GUI filed to tune the variable "net.internal-addr.pattern" in a future release.
If so, an administrator doesn't have to set it manually.
Some intern ... |
Topic: "register.contact.remote" works correctly ? |
lakeview
Replies: 5
Views: 9195
|
Forum: Brekeke SIP Server Forum Posted: Mon Jun 16, 2014 11:05 am Subject: "register.contact.remote" works correctly ? |
Another way to fix the issue is ..
Add this in the net.internal-addr.pattern = ^100
Brekeke SIP Server will treat 100.x.x.x as an internal (private) IP address.
so you don't need any DialPlan ... |
Topic: RTP Relay failed |
lakeview
Replies: 11
Views: 14781
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Forum: Brekeke SIP Server Forum Posted: Tue Dec 17, 2013 2:16 pm Subject: RTP Relay failed |
Are you using own SIP client developed on SDK?
Does the problem happen with both Xlite and your SIP client ? |
Topic: Reply OPTIONS to SBC |
lakeview
Replies: 6
Views: 12958
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Forum: Brekeke SIP Server Forum Posted: Fri Dec 13, 2013 11:27 pm Subject: Reply OPTIONS to SBC |
> Because I would like to check if there is DDoS attack to my server which does not logging in error log.
But you can see them in the Blocked Log even if you are using ver3.2.
Add some hon ... |
Topic: RTP Relay failed |
lakeview
Replies: 11
Views: 14781
|
Forum: Brekeke SIP Server Forum Posted: Fri Dec 13, 2013 11:18 pm Subject: RTP Relay failed |
Did it happen during a phone call?
What SIP client are you using?
> It Seems that there is a problem with the connection authentication.
SIP Auth and RTP are not associated. |
Topic: RTP Relay failed |
lakeview
Replies: 11
Views: 14781
|
Forum: Brekeke SIP Server Forum Posted: Tue Dec 10, 2013 10:10 am Subject: RTP Relay failed |
Did it happen during a phone call?
What SIP client and audio codec are you using? |
Topic: Admin Tool - INACTIVE- show error cannot open Database ... |
lakeview
Replies: 4
Views: 7068
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Forum: Brekeke SIP Server Forum Posted: Sun Dec 01, 2013 11:49 am Subject: Admin Tool - INACTIVE- show error cannot open Database ... |
> but Brekeke never send me an update.
Signup the eNews letter.
http://www.brekeke.com/company/enews.php |
Topic: Auto Sync in Version 3 Advanced ? |
lakeview
Replies: 4
Views: 8707
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Forum: Brekeke SIP Server Forum Posted: Wed Oct 30, 2013 1:09 pm Subject: Auto Sync in Version 3 Advanced ? |
Added the Note.
http://wiki.brekeke.com/wiki/Set-up-Auto-Sync-Brekeke-SIP-Server |
Topic: FYI: Block friendly-scanner and crawler |
lakeview
Replies: 2
Views: 7417
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Forum: Brekeke SIP Server Forum Posted: Mon Sep 23, 2013 3:55 pm Subject: FYI: Block friendly-scanner and crawler |
1. Brekeke Product Name and Version:
Brekeke SIP Server ver3.2
Add following DialPlan rules in the [Dial Plan] -> [Preliminary] page.
---------------------------------------
[Matching ... |
Topic: No voice Behind NAT |
lakeview
Replies: 17
Views: 23465
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Forum: Brekeke SIP Server Forum Posted: Thu Apr 11, 2013 2:36 pm Subject: No voice Behind NAT |
bensly,
For your case, both SIP clients must enable RTP/AVPF instead of RTP/AVP. |
Topic: No voice Behind NAT |
lakeview
Replies: 17
Views: 23465
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Forum: Brekeke SIP Server Forum Posted: Thu Apr 04, 2013 10:57 am Subject: No voice Behind NAT |
How about the [Port mapping] setting in the [RTP] page?
It should be "source port" in your case.
If you try another SIP client such as Linphone..
Does the same issue happen?
Also, ... |
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