Author |
Message |
Topic: Reduce size of To INVITE packet. MTU problem. |
Tata
Replies: 3
Views: 1516
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Forum: Brekeke SIP Server Forum Posted: Tue Dec 05, 2023 8:20 pm Subject: Reduce size of To INVITE packet. MTU problem. |
Which parts of SIP packet are consuming a packet size? SDP? |
Topic: Reduce size of To INVITE packet. MTU problem. |
Tata
Replies: 3
Views: 1516
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Forum: Brekeke SIP Server Forum Posted: Tue Dec 05, 2023 5:20 pm Subject: Reduce size of To INVITE packet. MTU problem. |
Using $b2bua=true reduces the size of the SIP packet.
It seems you already put it in the DeployPatterns.
but do you still have the issue? |
Topic: Strip square brackets in Via/received for ipv6 REG response |
Tata
Replies: 3
Views: 1577
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Forum: Brekeke SIP Server Forum Posted: Fri Nov 17, 2023 6:01 pm Subject: Strip square brackets in Via/received for ipv6 REG response |
Try DialPlan rule like this.
[Matching Patterns]
$request = ^REGISTER
$addrtype = 6
$param(Via,"received") = \[(.+)\]
$removeparam(Via,"received") = (.+)
[Deploy Patterns ... |
Topic: Source IP changes when using failover rule with redundancy. |
Tata
Replies: 6
Views: 3544
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Forum: Brekeke SIP Server Forum Posted: Fri Oct 13, 2023 10:41 pm Subject: Source IP changes when using failover rule with redundancy. |
"net.bind.use" enables the interface binding.
> Would I need to set net.bind.interface explicitly before using net.bind.use=true?
It doesn't matter. |
Topic: Source IP changes when using failover rule with redundancy. |
Tata
Replies: 6
Views: 3544
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Forum: Brekeke SIP Server Forum Posted: Fri Oct 13, 2023 10:10 am Subject: Source IP changes when using failover rule with redundancy. |
Add the line below in the net.bind.use = true |
Topic: 200 OK Response logging as errorr |
Tata
Replies: 1
Views: 2446
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Forum: Brekeke SIP Server Forum Posted: Wed May 17, 2023 8:10 pm Subject: 200 OK Response logging as errorr |
It is logged in Error Log because $response is usually used for sending an error response code.
Because the SIP Server sends "200 OK" to OPTIONS, it meets the requirement.
So you don't ... |
Topic: Use problem |
Tata
Replies: 1
Views: 2812
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Forum: Brekeke SIP Server Forum Posted: Thu Sep 08, 2022 7:51 am Subject: Use problem |
How do I import User Authentication and what does the imported correct file in csv format look like when opened via excel?
Yes. A CSV file can be edited with Excel.
Is it possible to change t ... |
Topic: 3cx ad Brekeke |
Tata
Replies: 5
Views: 4411
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Forum: Brekeke SIP Server Forum Posted: Fri Jul 08, 2022 10:34 am Subject: 3cx ad Brekeke |
The DialPlan rules you wrote return "400 Bad Request" to the 3CX and the PSTN..
Try the rule below. This rule accepts an INVITE without the authentication if it is sent from 3CX or PSTN
... |
Topic: 3cx ad Brekeke |
Tata
Replies: 5
Views: 4411
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Forum: Brekeke SIP Server Forum Posted: Tue Jul 05, 2022 12:24 pm Subject: 3cx ad Brekeke |
I don't recommend you to disable the authentication for INVITE, what you did with DialPlan, if the SIP Server can be reached from the Internet.
I recommend you to check the remote IP address of 3CX ... |
Topic: Manipulating SIP Fields in an INVITE |
Tata
Replies: 1
Views: 3282
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Forum: Brekeke SIP Server Forum Posted: Thu Apr 07, 2022 6:57 pm Subject: Manipulating SIP Fields in an INVITE |
Yes, you can replace these SIP headers with DialPlan.
For example..
[Matching Patterns]
$request = ^INVITE
$getSIPuser(Remote-Party-ID) = 1(.+)
[Deploy Patterns]
From = sip:%1@192.168. ... |
Topic: sip server relays Invite Message but rewrite key information |
Tata
Replies: 2
Views: 3193
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Forum: Brekeke SIP Server Forum Posted: Tue Mar 01, 2022 3:41 pm Subject: sip server relays Invite Message but rewrite key information |
What SIP UA client product are you using?
Does the second INVITE sent to UA3 have a different Call-ID from the original INVITE sent to UA2? |
Topic: TLS client verification without IP in the name fields |
Tata
Replies: 4
Views: 3582
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Forum: Brekeke SIP Server Forum Posted: Tue Jan 04, 2022 2:43 pm Subject: TLS client verification without IP in the name fields |
Have you restarted the SIP Server after you changed the configuration?
Do you still have the issue? |
Topic: TLS client verification without IP in the name fields |
Tata
Replies: 4
Views: 3582
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Forum: Brekeke SIP Server Forum Posted: Mon Jan 03, 2022 11:19 am Subject: TLS client verification without IP in the name fields |
Let you set [Peer Certification Validation] = "off" in [Configuration]->[SIP] page. |
Topic: Switching from Java to AdoptOpenJDK 11 (HotSpot) |
Tata
Replies: 8
Views: 6154
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Forum: Brekeke SIP Server Forum Posted: Wed Dec 15, 2021 4:13 pm Subject: Switching from Java to AdoptOpenJDK 11 (HotSpot) |
Do you need to replace Oracle Java 8 with AdoptOpenJDK 11?
The wiki topic below will help you.
https://docs.brekeke.com/sip/upgrade-java-version-to-java-11-or-later
But, to avoid downtime, I reco ... |
Topic: Which RFC defines how VAD is treated in SIP Signalling |
Tata
Replies: 2
Views: 3841
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Forum: Brekeke SIP Server Forum Posted: Wed Mar 10, 2021 5:50 pm Subject: Which RFC defines how VAD is treated in SIP Signalling |
It seems there is no RFC which defines " vad=no". |
Topic: WEBRTC Sip server - PBX |
Tata
Replies: 6
Views: 11095
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Forum: Brekeke PBX Forum Posted: Mon Feb 08, 2021 8:52 am Subject: WEBRTC Sip server - PBX |
Hi h.fabien and o.mahmoud,
Did you add new DialPlan rules or modify default rules?
You need to keep the default "From PBX" and "To PBX" rules applied to bridge SIP calls for Web ... |
Topic: CDR TIMEZONE FORMAT |
Tata
Replies: 2
Views: 4067
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Forum: Brekeke SIP Server Forum Posted: Mon Feb 01, 2021 5:25 pm Subject: CDR TIMEZONE FORMAT |
Try GMT+4 and restart the SIP Server. |
Topic: Avaya CM 5.2.1 connection with BSS |
Tata
Replies: 5
Views: 5022
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Forum: Brekeke SIP Server Forum Posted: Fri Nov 20, 2020 9:15 am Subject: Avaya CM 5.2.1 connection with BSS |
As a SIP proxy, BSS passes SDP what Avaya sent. so BSS doesn't care even if there are multiple codecs. |
Topic: Avaya CM 5.2.1 connection with BSS |
Tata
Replies: 5
Views: 5022
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Forum: Brekeke SIP Server Forum Posted: Thu Nov 19, 2020 11:22 am Subject: Avaya CM 5.2.1 connection with BSS |
I recommend you to talk to Avaya admin to configure the trunk because the settings are depending on versions and deployment. |
Topic: Avaya CM 5.2.1 connection with BSS |
Tata
Replies: 5
Views: 5022
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Forum: Brekeke SIP Server Forum Posted: Tue Nov 17, 2020 9:33 am Subject: Avaya CM 5.2.1 connection with BSS |
FYI:
https://docs.brekeke.com/interop/configure-brekeke-sip-server-with-avaya-pbx-cm-5-2 |
Topic: Brekeke listener is not listening (though service starts) |
Tata
Replies: 2
Views: 5194
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Forum: Brekeke SIP Server Forum Posted: Fri Jul 17, 2020 4:05 pm Subject: Brekeke listener is not listening (though service starts) |
If a firewall is running, disable it or open the UDP/TCP port 5060.
Also the SIP Server process should have a privilege to open the port. |
Topic: Database error and TLS 1.2 questions |
Tata
Replies: 2
Views: 4875
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Forum: Brekeke SIP Server Forum Posted: Fri Jul 17, 2020 3:56 pm Subject: Database error and TLS 1.2 questions |
> 1. Brekeke Product Name and Version: 3.3.5.8 Advanced
Update it to the latest version because 3.3.5.8 is pretty old.
https://docs.brekeke.com/sip/sip-history
> 1. When we click on th ... |
Topic: BSS is updating SDP with $rtp=false |
Tata
Replies: 2
Views: 4946
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Forum: Brekeke SIP Server Forum Posted: Fri Jun 26, 2020 3:43 pm Subject: BSS is updating SDP with $rtp=false |
set &net.nat.force=false in Deploy Patterns. |
Topic: One way Audio |
Tata
Replies: 15
Views: 16927
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Forum: Brekeke SIP Server Forum Posted: Wed May 27, 2020 2:13 pm Subject: One way Audio |
Do you have the one-way audio issue with other calls too? |
Topic: NATted calls not working anymore |
Tata
Replies: 25
Views: 27748
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Forum: Brekeke SIP Server Forum Posted: Mon May 11, 2020 9:14 am Subject: NATted calls not working anymore |
Let you look at the SIP Server's log... |
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