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Setup Trunk without authentication (Vodafone IP)


#1

Hey guys,

I need to config a sip-trunk for vodafone “IP Anlagenanschluss R.4a”.
Vodafone description (in german)

Actually I have no clue how to set it up…
I configured my firewall so that incomming traffic on a specific public IP is forwarded to the freepbx-server. I also took care that all outgoing traffic to the vodafone SBC is routed through the right public IP.
(so far so good).

Right now I have absolutely no idea what I should enter in my pjsip-trunk config…
I received the following data from vodafone:

Test Phone-No: +49XXXXXXXXX RNB 00-99
Realm-SIP Domain:xxxxxx.ngn.vodafone.de
Customer PBX: MY_PUBLIC_IP Port: 5060 UDP
VF-SBC IP + Port: 88.79.204.9 Port: 5060 UDP

I tried:

Authentication: NONE
Registration: NONE
SIP Server: [Realm-SIP Domain]
SIP Server Port: 5060
Qualify Frequency: 0
Outbound Proxy: 88.79.204.9:5060/;lr
From Domain: [Realm-SIP Domain]
Match (Permit): 88.79.204.9
Trust RPID/PAI: YES
Send RPID/PAI: Send P-Asserted-Identity header

Thats not working…

Is there anyone who can help me?

THX
mts


#3

I activated pjsip debug and I’m getting this messages:
There is one error " SIP/2.0 400 Bad Request" “Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)”

Any Idea whats wrong here?
I have an any / any inbound route but I don’t receive any signaling…

<— History Entry 3580 Received from 88.79.204.9:5060 at 1563185367 —>
INVITE sip:+49LOCALNUMBER@pbx1.nucleus.ngn.vodafone.de:5060 SIP/2.0
Via: SIP/2.0/UDP 88.79.204.9:5060;received=88.79.204.9;branch=z9hG4bKebq34h00983et2ekci50.1;origin=172.19.116.80
To: sip:+49LOCALNUMBER@nbgsx001.ngn.vodafone.de;user=phone
From: sip:+49EXTERNALNUMBER@ims.vodafone.de;user=phone;tag=SD0r0b001-2239645f
Call-ID: SD0r0b001-c395cdbe96ad36f5ddb5dfbfa3ea4f60-ct4u830040
CSeq: 1 INVITE
Max-Forwards: 60
Contact: sip:+49EXTERNALNUMBER@88.79.204.9:5060;transport=udp
Date: Mon, 15 Jul 2019 12:09:27 GMT
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Supported: resource-priority
P-Asserted-Identity: sip:+49EXTERNALNUMBER@ims.vodafone.de;user=phone
P-Asserted-Identity: tel:+49EXTERNALNUMBER
Accept: application/sdp
P-Early-Media: supported
Content-Type: application/sdp
Content-Length: 289
Content-Type: application/sdp
Content-Length: 289

v=0
o=- 0 0 IN IP4 88.79.204.9
s=IMSS
c=IN IP4 88.79.204.9
t=0 0
m=audio 55000 RTP/AVP 96 9 8 101 102
a=rtpmap:101 telephone-event/8000
a=rtpmap:102 telephone-event/16000
a=ptime:20
a=maxptime:30
a=fmtp:96 mode-set=0,1,2; mode-change-period=2; mode-change-neighbor=1; max-red=0

<— History Entry 3581 Sent to 88.79.204.9:5060 at 1563185367 —>
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 88.79.204.9:5060;rport=5060;received=88.79.204.9;branch=z9hG4bKebq34h00983et2ekci50.1;origin=172.19.116.80
Call-ID: SD0r0b001-c395cdbe96ad36f5ddb5dfbfa3ea4f60-ct4u830040
From: sip:+49EXTERNALNUMBER@ims.vodafone.de;user=phone;tag=SD0r0b001-2239645f
To: sip:+49LOCALNUMBER@nbgsx001.ngn.vodafone.de;user=phone;tag=z9hG4bKebq34h00983et2ekci50.1
CSeq: 1 INVITE
Warning: 399 SIP “Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)”
Server: FPBX-14.0.13.4(15.4.0)
Content-Length: 0

<— History Entry 3582 Received from 88.79.204.9:5060 at 1563185367 —>
ACK sip:+49LOCALNUMBER@pbx1.nucleus.ngn.vodafone.de:5060 SIP/2.0
Via: SIP/2.0/UDP 88.79.204.9:5060;received=88.79.204.9;branch=z9hG4bKebq34h00983et2ekci50.1;origin=172.19.116.80
CSeq: 1 ACK
To: sip:+49LOCALNUMBER@nbgsx001.ngn.vodafone.de;user=phone;tag=z9hG4bKebq34h00983et2ekci50.1
From: sip:+49EXTERNALNUMBER@ims.vodafone.de;user=phone;tag=SD0r0b001-2239645f
Call-ID: SD0r0b001-c395cdbe96ad36f5ddb5dfbfa3ea4f60-ct4u830040
Max-Forwards: 60
Content-Length: 0
Content-Length: 0


#4

Several things seem fishy:

The SDP is indeed malformed. RTP Payload Type 96 is in the ‘dynamic’ range but there is no ‘a’ line to specify the codec. Based on the fmtp parameters, the line that is missing is probably:
a=rtpmap:96 AMR-WB/16000
Vodafone likely supports AMR-WB, so calls from mobiles with VoLTE can connect in HD (wideband).

However, possibly Vodafone is sending valid SDP, but a SIP ALG in a router, firewall or SBC in the path is corrupting the packet. If your PBX is not directly on a public IP address, post what networking devices you have between internet and PBX.

Also, I’d expect pjsip to be robust enough to ignore the malformed PT 96 and just choose another codec. In your trunk settings, what do you have in pjsip General for Context? On an inbound call attempt, does the Asterisk log show that context being run? If so, post what gets logged.