[Spce-user] newbie question about Rewrite Rule Sets

Tue Dec 30 09:20:49 EST 2014

Hi,
general speaking you have the following scenarios:

== Call between subscribers ==
Subscriber A call B, both are under the same Domain

1. A send INVITE to SPCE
2. INBOUND REWRITE RULES of A's Domain is used
3. SPCE lookup B as callee
4. OUTBOUND REWRITE RULES of B's Domain is used
5. SPCE send INVITE to B

== Call from subscribers to PEER ==
Subscriber A call outbound.

1. A send INVITE to SPCE
2. INBOUND REWRITE RULES of A's Domain is used
3. SPCE Select the peer based on Caller/Callee
4. OUTBOUND REWRITE RULES of PEER is used
5. SPCE send INVITE to the peer

== Call from peer ==
Incoming call from peer to Subscriber A

1. PEER send INVITE to SPCE
2. INBOUND REWRITE RULES of PEER is used
3. SPCE lookup A as callee
4. OUTBOUND REWRITE RULES of A's Domain is used
5. SPCE send INVITE to A

NOTE: You can set Rewrite rules set in DOMAIN level. If you set another
Rewrite rules set on SUBSCRIBER's preferences level, this will overwrite
the Domain set.

Regards,
Daniel

On 12/30/2014 02:30 PM, mig at gmx.ch wrote:
> Hi all,
> First of all sorry for my stupid questions but I'm a newbie and
> installed a couple of days/weeks SIP:Provider CE Version mr3.6.2 only
> for test purpose in my LAB.
> On chapter 6.6 Configuring Rewrite Rule Sets (Handbook) it is written
> that on the NGCP every number is treated in E.164 format. Does that
> mean with every phone number From/To (caller/callee) ?
> I'm in switzerland and have a DDI 0435444390 - 99 assigned to me. I
> receive from the provider all the number with leading 0, that means
> for national 0 and international 00. The provider expects to receive
> the whole number from my DDI, equal 0435444390 (10 digits) and if it
> is a international call the callee number with 00.
>  
>  
> How are the Rewrite Rule Sets processed? I mean they can be assigned
> either to a peering,domain or subscriber. If a call is passing a
> peering server which has a rewrite rule set assigned I guess this one
> is processed, but what if the domain has also one set? are both of
> them proccessed or only that one from the peering server?
> How can I troubleshoot the rewrite rules what has been processed on a
> call so that I can check if the regex works as desired?
>  
>  
> Example of an in and outgoing call:
>  
> Swiss numbering plan:
> http://www.bakom.admin.ch/themen/telekom/00479/00604/index.html?lang=en&download=NHzLpZeg7t,lnp6I0NTU042l2Z6ln1ad1IZn4Z2qZpnO2Yuq2Z6gpJCDdIN5f2ym162epYbg2c_JjKbNoKSn6A--
>
> <cc> = 41
> <ac> = 43
> <sn> = 5444390 - 99
> <IP Phone> IP address 172.16.1.14
> <NGCP> IP address 172.16.1.7
> <PSTN> IP address 62.2.46.12
>  
> originating call <IP Phone> , terminating <PSTN>
> <IP Phone> ---> <NGCP> ---> <Peering> ---> <PSTN>
>  
> INVITE sip:0800800800 at sip.migmig.lan SIP/2.0
> Via: SIP/2.0/UDP 172.16.1.14:5060;branch=z9hG4bK-8d69118c
> From: "0435444395" <sip:0435444395 at sip.migmig.lan>;tag=c553358ee74d6b93o2
> To: <sip:0800800800 at sip.migmig.lan>
> Call-ID: c821dd6a-2a6d8567 at 172.16.1.14
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: "0435444395" <sip:0435444395 at 172.16.1.14:5060>
> Expires: 240
> User-Agent: Linksys/SPA941-5.1.8
> Content-Length: 208
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: replaces
> Content-Type: application/sdp
> v=0
> o=- 22782282 22782282 IN IP4 172.16.1.14
> s=-
> c=IN IP4 172.16.1.14
> t=0 0
> m=audio 16462 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
>  
>  
>  
> originating call <PSTN> , terminating <IP Phone>
> <PSTN> ---> <Peering> ---> <NGCP> ---> <IP Phone>
>  
> INVITE sip:0435444395 at 172.16.1.7:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 62.2.46.12:5060;branch=z9hG4bK18f4ok00eg51m0n9j0m0.1
> From: "0445777593"
> <sip:0445777593 at 212.55.198.150;user=phone>;tag=528762297
> To: 0435444395 <sip:0435444395 at vbcOTF005.cablecom.net:5060;user=phone>
> Call-ID: 4aa9fda56b9308d037bd213252ff18d4 at 212.55.198.150
> CSeq: 1 INVITE
> Max-Forwards: 67
> Supported: timer
> Session-Expires: 1800
> Min-SE: 1800
> Contact: <sip:0445777593 at 62.2.46.12:5060;transport=udp>
> Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
> Date: Sun, 21 Dec 2014 20:54:08 GMT
> Content-Type: application/sdp
> Content-Length: 260
> v=0
> o=root 1858387754 1858387754 IN IP4 62.2.46.12
> s=Asterisk PBX 1.6.1.24
> c=IN IP4 62.2.46.12
> t=0 0
> m=audio 16626 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=sendrecv
> a=ptime:20
>
> Regards,
> Miguel
>
>
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