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JuanFrancisco Brekeke Newbie
Joined: 17 Sep 2018 Posts: 3
Location: Colombia
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Posted: Mon Sep 17, 2018 12:55 pm Post subject: How to replace Contact at dial plan |
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1. Brekeke Product Name and Version: version 2
2. Java version: jre 7
3. OS type and the version: windows server 2008
4. UA (phone), gateway or other hardware/software involved:
5. Your problem:
I am using at deploy pattern of dial plan for outgoing calls:
_Contact=<sip:%1@xxx.xxx.xxx.xxx:5060>
$ifdst=xxx.xxx.xxx.xxx
&net.sip.replacecontact=false
&net:rtp.ifdst=xxx.xxx.xxx.xxx
Where xxx.xxx.xxx.xxx is the IP address i want to be the calls`s one.
But Sip server does not make the replace.
Thanks and best regrads _________________ Juan Manuel Acevedo
acevedojaramillo@une.net.co |
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Haddas Brekeke Guru
Joined: 17 Jan 2008 Posts: 170
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Posted: Mon Sep 17, 2018 2:07 pm Post subject: |
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For outgoing calls, the definition should be "Contact" instead of "_Contact".
For response packet, you can use "_Contact". |
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JuanFrancisco Brekeke Newbie
Joined: 17 Sep 2018 Posts: 3
Location: Colombia
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Posted: Tue Sep 18, 2018 1:48 pm Post subject: |
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Thanks for your answer.
I still have problems with the outgoing call, I do no understand what happens
I attach Dial plan matching and deploy patterns, ARS matching and deploy patterns, dialed number and trace of the outgoing call.
It is not according with the system must to be, waht is my mistake?
Sip server dial plan:
Matching Patterns | $port = ^15062 $localhost = ^true $request = ^INVITE To = sip:(.+)@172.17.179.166
| Deploy Patterns | $auth = false &net.sip.hide.loopback = true $b2bua = true X-PBX-Param = $ifdst = 172.22.24.110 &net.rtp.ifdst = 172.22.24.110 &net.sip.replacecontact = false Contact = <sip:%1@172.22.24.110:5060>
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ARS:
Matching Pattern
To sip:3(.+)@
To sip:3$1@172.17.179.166
Dialed Number: 3348800
Outgoing call trace
Frame 5807: 839 bytes on wire (6712 bits), 839 bytes captured (6712 bits) on interface 0
Ethernet II, Src: HuaweiTe_2e:5e:82 (00:18:82:2e:5e:82), Dst: HewlettP_a7:28:20 (9c:b6:54:a7:28:20)
Internet Protocol Version 4, Src: 172.17.179.166, Dst: 172.22.24.110
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (180)
Status-Line: SIP/2.0 180 Ringing
Message Header
Via: SIP/2.0/UDP 172.24.250.182:5060;received=172.22.24.110;branch=z9hG4bK14445b1c91b4a8ca3ee-d94933b1-adaa0e94;rport=5060
From: "44037020" <sip:44037020@172.22.24.110:5060>;tag=b64443044p
To: <sip:3348800@172.17.179.166>;tag=3xvz0e2v-CC-23
Call-ID: 9d5fcad4-615458b1-724780c1-cfbddb6@172.24.250.182
CSeq: 1 INVITE
Contact: <sip:3348800@172.17.179.166:5060;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 248
Content-Type: application/sdp
Thanks _________________ Juan Manuel Acevedo
acevedojaramillo@une.net.co |
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Haddas Brekeke Guru
Joined: 17 Jan 2008 Posts: 170
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Posted: Tue Sep 18, 2018 4:07 pm Post subject: |
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It seems you are using version 2... it is too old (almost 10 years ago..)
Let you upgrade it to the latest version. |
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