Author |
Message |
Topic: Proxy authentication required even if UA is registered |
taitan
Replies: 15
Views: 18226
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Forum: Brekeke SIP Server Forum Posted: Wed Jul 11, 2018 11:24 am Subject: Proxy authentication required even if UA is registered |
Are you still using Brekeke SIP Server version 3.5.2.8?
If so, let you upgrade it to the latest version because there are new logging function which may help your analysis.
http://www.brekeke.c ... |
Topic: Sending calls to multiple gateways |
taitan
Replies: 3
Views: 6707
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Forum: Brekeke SIP Server Forum Posted: Wed Jul 11, 2018 11:07 am Subject: Sending calls to multiple gateways |
> Should I place each gateway with its capacity (using route.underlimit) plus $continue=true in its own dial plan entry to make it use sequentially all gateways or can I put al gateways in a single ... |
Topic: Avoid SRTP in BSS |
taitan
Replies: 1
Views: 2547
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Forum: Brekeke SIP Server Forum Posted: Mon Mar 26, 2018 2:39 pm Subject: Avoid SRTP in BSS |
SRTP is defined in SDP so it is negotiable.
It means the gateway simply ignores SDP attributes about SRTP and sends back non-SRTP SDP with 200 OK response.
For TLS, it is not negotiable generally ... |
Topic: Control the number of channels allowed in a SIP trunk |
taitan
Replies: 7
Views: 5085
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Forum: Brekeke SIP Server Forum Posted: Thu Oct 26, 2017 7:31 pm Subject: Control the number of channels allowed in a SIP trunk |
Matching Patterns
$request = ^INVITE
$getHost(Contact) = ^1\.1\.1\.1$|^2\.2\.2\.2$
To = sip:(1212.+)@
$route.underlimit("gateway1","30") = true |
Topic: Control the number of channels allowed in a SIP trunk |
taitan
Replies: 7
Views: 5085
|
Forum: Brekeke SIP Server Forum Posted: Mon Aug 21, 2017 11:16 am Subject: Control the number of channels allowed in a SIP trunk |
Try this.
Matching Patterns
$request = ^INVITE
$addr = ^1\.1\.1\.1$|^2\.2\.2\.2$
To = sip:(1212.+)@
$route.underlimit("gateway1","30") = true
Deploy Patterns
To = sip:%1 ... |
Topic: Delete the record in Push Notification when USER logout |
taitan
Replies: 1
Views: 15113
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Forum: Brekeke PBX Forum Posted: Wed Aug 09, 2017 9:09 pm Subject: Delete the record in Push Notification when USER logout |
Which SIP client product are you using? |
Topic: sip server change contact ip address to its private address |
taitan
Replies: 5
Views: 5394
|
Forum: Brekeke SIP Server Forum Posted: Wed Aug 09, 2017 9:00 pm Subject: sip server change contact ip address to its private address |
Generally an order of these SIP headers are not important.
Is it the same sipphone 2 at both server1 and server 2 testings?
If so, is it the same IP address from server 1 and server 2? |
Topic: sip server change contact ip address to its private address |
taitan
Replies: 5
Views: 5394
|
Forum: Brekeke SIP Server Forum Posted: Tue Aug 08, 2017 5:26 pm Subject: sip server change contact ip address to its private address |
Which firewall product are you using for SIP Server 1?
Can you go to [Dial Plan]->[History] page after you make a test call over the Server 1?
And then click the latest field number which ind ... |
Topic: sip server change contact ip address to its private address |
taitan
Replies: 5
Views: 5394
|
Forum: Brekeke SIP Server Forum Posted: Mon Aug 07, 2017 9:03 am Subject: sip server change contact ip address to its private address |
Go to [Status]->[SIP Server] status page and find the [interface] filed.
Does it show the SIP Server's public IP address?
If not, set the public IP address at [Configuration]->[System] page ... |
Topic: 3G/4G Connection Unstable or Failed to Register |
taitan
Replies: 7
Views: 5500
|
Forum: Brekeke SIP Server Forum Posted: Fri Jan 06, 2017 12:17 pm Subject: 3G/4G Connection Unstable or Failed to Register |
You can ignore "407" in the error log. It will happen if you use SIP authentication in the SIP Server. SIP client will retry REGISTER with the credential after 407. That's why you found the ... |
Topic: Setting source IP on outgoing packets |
taitan
Replies: 2
Views: 9724
|
Forum: Brekeke SIP Server Forum Posted: Fri Jan 06, 2017 12:05 pm Subject: Setting source IP on outgoing packets |
Use Linux's "route" command. |
Topic: Client on 4G can't get INVITE |
taitan
Replies: 3
Views: 4400
|
Forum: Brekeke SIP Server Forum Posted: Thu May 26, 2016 7:17 pm Subject: Client on 4G can't get INVITE |
Which SIP client products are you using? |
Topic: Brekeke MT cannot receive incoming call. |
taitan
Replies: 25
Views: 36135
|
Forum: Brekeke PBX Forum Posted: Thu Apr 28, 2016 11:57 am Subject: Brekeke MT cannot receive incoming call. |
Can you find "404" in the SIP Server's [Logs]->[Error Logs] or [Session Logs]?
If you found "404" in the [Error Logs], you have a problem in DialPlan.
If you found "404 ... |
Topic: Brekeke MT cannot receive incoming call. |
taitan
Replies: 25
Views: 36135
|
Forum: Brekeke PBX Forum Posted: Tue Apr 26, 2016 9:10 pm Subject: Brekeke MT cannot receive incoming call. |
Have you edited DialPlan rules?
If you use ARS, you don't have to edit DialPlan.
Also let me know the name of ITSP if possible.. |
Topic: Brekeke MT cannot receive incoming call. |
taitan
Replies: 25
Views: 36135
|
Forum: Brekeke PBX Forum Posted: Tue Apr 26, 2016 11:33 am Subject: Brekeke MT cannot receive incoming call. |
Ha,
Have you read this?
http://wiki.brekeke.com/wiki/Setup-for-Connecting-ITSP |
Topic: Brekeke MT cannot receive incoming call. |
taitan
Replies: 25
Views: 36135
|
Forum: Brekeke PBX Forum Posted: Mon Apr 25, 2016 7:56 pm Subject: Brekeke MT cannot receive incoming call. |
Can you capture INVITE packet if you make a incoming call?
If not, the ISP didn't send calls to the Brekeke PBX. |
Topic: Brekeke MT cannot receive incoming call. |
taitan
Replies: 25
Views: 36135
|
Forum: Brekeke PBX Forum Posted: Mon Apr 25, 2016 9:59 am Subject: Brekeke MT cannot receive incoming call. |
If there are no records which indicate incoming calls, it seems ISP didn't send calls to the Brekeke PBX.
Which ISP is it?
Does the issue happen always? |
Topic: Brekeke MT cannot receive incoming call. |
taitan
Replies: 25
Views: 36135
|
Forum: Brekeke PBX Forum Posted: Thu Apr 21, 2016 10:16 am Subject: Brekeke MT cannot receive incoming call. |
Are there any records in [SIP Server]->[Logs]->[Session Logs] and [Error Logs] which indicate incoming calls? |
Topic: How to Generate CDR/Call Logs every hour |
taitan
Replies: 1
Views: 7261
|
Forum: Brekeke SIP Server Forum Posted: Tue Nov 10, 2015 10:53 pm Subject: How to Generate CDR/Call Logs every hour |
Have you looked at this API doc? It will meet the requirement.
http://www.brekeke.com/doc/sip/sip_accounting_plugin.txt |
Topic: Cannot turn to P2P when ICE enabled |
taitan
Replies: 11
Views: 14707
|
Forum: Brekeke SIP Server Forum Posted: Tue Nov 10, 2015 10:52 pm Subject: Cannot turn to P2P when ICE enabled |
Are INVITE and UPDATE using a same Call-ID? |
Topic: Cannot turn to P2P when ICE enabled |
taitan
Replies: 11
Views: 14707
|
Forum: Brekeke SIP Server Forum Posted: Mon Nov 09, 2015 9:32 pm Subject: Cannot turn to P2P when ICE enabled |
Are you sure the above DialPlan rule is executed?
It seems $rtp=false is not called.
Check it at the DialPlan History page.
> c=117.22.xx.xx
Is it the caller side's global IP address? ... |
Topic: Proxy authentication required even if UA is registered |
taitan
Replies: 15
Views: 18226
|
Forum: Brekeke SIP Server Forum Posted: Mon Nov 02, 2015 6:30 pm Subject: Proxy authentication required even if UA is registered |
You don't have to worry about 407 because it happens every time if the SIP Server authenticates SIP requests.
For 481, can you find it in the Error logs page?
Which SIP request method was it? INV ... |
Topic: Cannot CANCEL a re-INVITE request |
taitan
Replies: 2
Views: 8541
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Forum: Brekeke SIP Server Forum Posted: Mon Nov 02, 2015 6:09 pm Subject: Cannot CANCEL a re-INVITE request |
Since re-INVITE is not initial request, "100 Trying" will not be sent.
"any requests" means "any initial request".
Refer the Brekeke SIP Server's document about "1 ... |
Topic: SIP NOTIFY (Event: check-sync) not relaying to UA |
taitan
Replies: 21
Views: 17838
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Forum: Brekeke SIP Server Forum Posted: Thu Jun 18, 2015 12:42 pm Subject: SIP NOTIFY (Event: check-sync) not relaying to UA |
As I said, Call-ID of NOTIFY should be unique if you want to catch it with DialPlan. It means DialPlan doesn't catch a SIP packet if it is sent within an existing dialog.
Are you sure NOTIFY does ... |
Topic: SIP NOTIFY (Event: check-sync) not relaying to UA |
taitan
Replies: 21
Views: 17838
|
Forum: Brekeke SIP Server Forum Posted: Thu Jun 18, 2015 11:27 am Subject: SIP NOTIFY (Event: check-sync) not relaying to UA |
What does "passes through" mean?
Do you mean that the SIP Server forwards NOTIFY without an issue but can not catch it with DialPlan? |
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