Hi alltogether,
I am struggling with an inbound routing of calls from “Toplink Xpress”.
Found great tips in this forum from Stewart1 regarding toplink
however - even with “sendrpid=pai” the inbound rules are not working (should call SIP 2001)
Only the Userdefinition is working as default (is calling SIP 2000)
The INVITE logs snipped are following shortly - thanks for the help in advance !
May 31 11:09:33 VERBOSE15459 chan_sip.c:
— SIP read from UDP:SOURCE_IPV6:5060 —
INVITE sip:P102_TOPLINK_USER AT TARGET_IPV6_ASTERISK:5060 SIP/2.0
Via: SIP/2.0/UDP SOURCE_IPV6:5060;branch=z9hG4bKac1706824423
Max-Forwards: 52
From: sip:+49172_SOURCE_MOBILE_PHONE AT PROVIDER;user=phone;id=TelNGN;tag=1c1297377107
To: sip:+4989_TARGET_FIXED_LINE_NR AT PROVIDER
Call-ID: 105970020315202110932 AT SOURCE_IPV6
CSeq: 1 INVITE
Contact: sip:+49172_SOURCE_MOBILE_PHONE AT SOURCE_IPV6:5060;line=sr-N6IAzBFwMJZLWJZfWmZfM.P-W.y6Mx1dCRsIH9WLOjZ6mjt-Pqs0.msek9YZ…pWOBWl.4eSWtpTRqfWOhq1KLssWquZM9W5pwIa.9rsW4uZk.p9NKp4K6fsN4uZCBpWkUI0.jpCp4uZahW-p6tMMueloFs.jjfAkUKuguKsTc**
Supported: from-change,100rel,timer,replaces,histinfo,sdp-anat
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
P-Preferred-Identity: sip:+49172_SOURCE_MOBILE_PHONE AT PROVIDER
User-Agent: TELES-SBC
P-Asserted-Identity: sip:+49172_SOURCE_MOBILE_PHONE AT PROVIDER;user=phone
P-Charging-Vector: TARGET_IPV6_ASTERISK
Content-Type: application/sdp
Content-Length: 451
P-Early-Media: sendrecv
P-Ie-Display:
v=0
o=- 1831960063 1040775049 IN IP6 2001:4180:3:2:213:218:12:40
s=TELES-SBC
c=IN IP6 2001:4180:3:2:213:218:12:40
t=0 0
m=audio 12030 RTP/AVP 8 0 18 2 101
c=IN IP6 2001:4180:3:2:213:218:12:40
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:on - - - -
a=rtcp:12031 IN IP6 2001:4180:3:2:213:218:12:40
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
May 31 11:09:33 VERBOSE15459 chan_sip.c: — (18 headers 17 lines) —
May 31 11:09:33 VERBOSE15459 chan_sip.c: Sending to SOURCE_IPV6:5060 (no NAT)
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Sending to SOURCE_IPV6:5060 (no NAT)
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Using INVITE request as basis request - 105970020315202110932 AT SOURCE_IPV6
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found peer ‘toplink1’ for ‘+49172_SOURCE_MOBILE_PHONE’ from SOURCE_IPV6:5060
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found RTP audio format 8
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found RTP audio format 0
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found RTP audio format 18
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found RTP audio format 2
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found RTP audio format 101
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found audio description format PCMA for ID 8
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found audio description format telephone-event for ID 101
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found audio description format PCMU for ID 0
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found audio description format G729 for ID 18
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Found audio description format G726-32 for ID 2
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|g726|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Peer audio RTP is at port 2001:4180:3:2:213:218:12:40:12030
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c: Looking for P102_TOPLINK_USER in toplink-in (domain TARGET_IPV6_ASTERISK)
May 31 11:09:33 VERBOSE15459C-0000000b sip/route.c: sip_route_dump: route/path hop: sip:+49172_SOURCE_MOBILE_PHONE AT SOURCE_IPV6:5060;line=sr-N6IAzBFwMJZLWJZfWmZfM.P-W.y6Mx1dCRsIH9WLOjZ6mjt-Pqs0.msek9YZ…pWOBWl.4eSWtpTRqfWOhq1KLssWquZM9W5pwIa.9rsW4uZk.p9NKp4K6fsN4uZCBpWkUI0.jpCp4uZahW-p6tMMueloFs.jjfAkUKuguKsTc**
May 31 11:09:33 VERBOSE15459C-0000000b chan_sip.c:
— Transmitting (no NAT) to SOURCE_IPV6:5060 —
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SOURCE_IPV6:5060;branch=z9hG4bKac1706824423;received=SOURCE_IPV6
From: sip:+49172_SOURCE_MOBILE_PHONE AT PROVIDER;user=phone;id=TelNGN;tag=1c1297377107
To: sip:+4989_TARGET_FIXED_LINE_NR AT PROVIDER
Call-ID: 105970020315202110932 AT SOURCE_IPV6
CSeq: 1 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:P102_TOPLINK_USER AT TARGET_IPV6_ASTERISK:5060
Content-Length: 0
May 31 11:09:33 VERBOSE15594C-0000000b chan_sip.c: Audio is at 17042
May 31 11:09:33 VERBOSE15594C-0000000b chan_sip.c: Adding codec ulaw to SDP
May 31 11:09:33 VERBOSE15594C-0000000b chan_sip.c: Adding codec alaw to SDP
May 31 11:09:33 VERBOSE15594C-0000000b chan_sip.c: Adding codec gsm to SDP
May 31 11:09:33 VERBOSE15594C-0000000b chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
May 31 11:09:33 VERBOSE15594C-0000000b chan_sip.c: Reliably Transmitting (no NAT) to 2001:a61:258e:be00:2e3a:fdff:febe:917b:5060:
INVITE sip:2000 AT 2001:a61:258e:be00:2e3a:fdff:febe:917b;uniq=926EAB31F80FE98695AF7ED4E317A SIP/2.0
Via: SIP/2.0/UDP TARGET_IPV6_ASTERISK:5060;branch=z9hG4bK07d8f320
Max-Forwards: 70
From: sip:+49172_SOURCE_MOBILE_PHONE AT TARGET_IPV6_ASTERISK;tag=as33071bc5
To: sip:2000 AT 2001:a61:258e:be00:2e3a:fdff:febe:917b;uniq=926EAB31F80FE98695AF7ED4E317A
Contact: sip:+49172_SOURCE_MOBILE_PHONE AT TARGET_IPV6_ASTERISK:5060
Call-ID: 04998a2a1a93a99c06732bd707f13d41 AT TARGET_IPV6_ASTERISK:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Date: Mon, 31 May 2021 09:09:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “+49172_SOURCE_MOBILE_PHONE” sip:+49172_SOURCE_MOBILE_PHONE AT TARGET_IPV6_ASTERISK
Content-Type: application/sdp
Content-Length: 347