Forwarding to SIP TRUNK with Easybell: Caller Number Not Transmitted, Only the Easybell Number

Asterisk Asterisk SIP
1

Hello everyone,

I am new to this topic, but with the help of examples and AI, I decided to tackle it, thinking it would be relatively straightforward to implement.

Currently, I want to forward a number from Easybell to a Zoom room number. To join a Zoom meeting via phone, a relatively long number, room ID, and password must be entered. I want to shorten this process by setting up call forwarding that already includes the password and room ID.

Everything works so far. Occasionally, the password is not automatically used, and the caller is asked to enter it manually, but otherwise, the call successfully lands in the Zoom room.

However, I also want the caller’s number to be transmitted so that each caller does not appear with the same number or no number at all, making it possible to distinguish who is calling. Currently, this is not working, and Easybell states that the issue must somehow be related to Asterisk.

I have a pjsip.conf and an extensions.conf file.

In pjsip.conf, I have added the following contents at the bottom:

[easybell]
type=registration
outbound_auth=easybell-auth
server_uri=sip:sip.easybell.de
client_uri=sip:0049123456@sip.easybell.de
retry_interval=60

[easybell-identify]
type=identify
endpoint=easybell-endpoint
match=sip.easybell.de
match=195.xxx.xx.xx

[easybell-auth]
type=auth
auth_type=userpass
username=0049123456
password=XpasswordX

[easybell-trunk]
type=aor
contact=sip:sip.easybell.de

[easybell-endpoint]
type=endpoint
context=incoming
disallow=all
allow=alaw
allow=ulaw
dtmf_mode=rfc4733
outbound_auth=easybell-auth
aors=easybell-trunk
from_user=004912345
from_domain=sip.easybell.de
send_pai=yes
send_rpid=yes
trust_id_inbound=yes
trust_id_outbound=yes
;callerid=asreceived

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

In extensions.conf, I have added the following contents at the bottom:

[incoming]
exten => _X.,1,NoOp(Incoming call on Easybell)
   same => n,Answer()
   same => n,Wait(2)  ; Wait time to ensure the call is stable
   same => n,Set(CALLERID(all)=<sip:${CALLERID(num)}@sip.easybell.de>)
   same => n,Set(CALLERID(num)=0170123456)
   same => n,Set(PJSIP_HEADER(add,P-Asserted-Identity)=sip:${CALLERID(num)}@sip.easybell.de)
   same => n,Set(PJSIP_HEADER(add,Remote-Party-ID)=sip:${CALLERID(num)}@sip.easybell.de)
   same => n,Dial(PJSIP/+4969987654@easybell-endpoint,,D(654654654##w7*654789##),b)
   same => n,Hangup()

exten => s,1,NoOp(Unknown incoming number, forwarding to default)
   same => n,Goto(incoming,_X.,1)

Apparently, neither PAI nor RPID is transmitted correctly. Does anyone have an idea how to fix this?

2

This is invalid.

You have Asterisk configured to send PAI and RPID itself using “send_pai” and “send_rpid”, you can’t also add a header for it.

You need to provide a SIP trace using “pjsip set logger on” with a call attempt to see what was actually sent.

3

Yes, that’s right, here is the full output once again:

asterist-pbx*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Transmitting SIP request (602 bytes) to UDP:195.185.yy.yy:5060 --->
REGISTER sip:sip.easybell.de SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPj6f92358c-90c9-4b75-ae44-8f03e63d60b1
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=8d1d9bba-94d2-4e6b-9357-2655f5707242
To: <sip:00496xxx8xxx2@sip.easybell.de>
Call-ID: 4c629280-5bbf-44ed-911f-58da44f32d17
CSeq: 16724 REGISTER
Contact: <sip:s@192.168.xxx.xxx:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Length:  0


<--- Received SIP response (642 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPj6f92358c-90c9-4b75-ae44-8f03e63d60b1;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=8d1d9bba-94d2-4e6b-9357-2655f5707242
To: <sip:00496xxx8xxx2@sip.easybell.de>;tag=95c37a12bff1a2c36d72bf8333176544.78550000
Call-ID: 4c629280-5bbf-44ed-911f-58da44f32d17
CSeq: 16724 REGISTER
P-NGCP-Auth-IP: 192.168.251.xxx
P-NGCP-Auth-UA: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
WWW-Authenticate: Digest realm="sip.easybell.de", nonce="Z7xI22e8R6/fcJH80EWNv2B8SRf4S/8Z"
Server: Sipwise NGCP Proxy mr11.5.1
Content-Length: 0


<--- Transmitting SIP request (789 bytes) to UDP:195.185.yy.yy:5060 --->
REGISTER sip:sip.easybell.de SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPj45cabacf-84d9-4211-806f-b11e9a2ac06c
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=8d1d9bba-94d2-4e6b-9357-2655f5707242
To: <sip:00496xxx8xxx2@sip.easybell.de>
Call-ID: 4c629280-5bbf-44ed-911f-58da44f32d17
CSeq: 16725 REGISTER
Contact: <sip:s@192.168.xxx.xxx:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Authorization: Digest username="00496xxx8xxx2", realm="sip.easybell.de", nonce="Z7xI22e8R6/fcJH80EWNv2B8SRf4S/8Z", uri="sip:sip.easybell.de", response="b90a9d31c8aab0068a1a964a4443146b"
Content-Length:  0


<--- Received SIP response (589 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPj45cabacf-84d9-4211-806f-b11e9a2ac06c;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=8d1d9bba-94d2-4e6b-9357-2655f5707242
To: <sip:00496xxx8xxx2@sip.easybell.de>;tag=95c37a12bff1a2c36d72bf8333176544.78550000
Call-ID: 4c629280-5bbf-44ed-911f-58da44f32d17
CSeq: 16725 REGISTER
P-NGCP-Auth-IP: 192.168.251.xxx
P-NGCP-Auth-UA: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Server: Sipwise NGCP Proxy mr11.5.1
Contact: <sip:s@192.168.xxx.xxx:5060>;expires=90
Content-Length: 0


<--- Received SIP request (868 bytes) from UDP:195.185.yy.yy:5060 --->
INVITE sip:s@192.168.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 195.185.yy.yy;branch=z9hG4bKQ7CwRaz.kXdk7;rport
From: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
To: <sip:s@sip.easybell.de>
CSeq: 10 INVITE
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
Max-Forwards: 68
Supported: histinfo
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK
Content-Type: application/sdp
Content-Length: 346
Contact: <sip:6DEEECFA-67BC47B80003327C-2C4756C0@195.185.yy.yy;transport=udp>

v=0
o=- 19 1056709149 IN IP4 195.185.yy.yy
s=-
c=IN IP4 195.185.yy.yy
t=0 0
m=audio 28774 RTP/AVP 9 8 0 100 105
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=rtpmap:105 telephone-event/16000
a=fmtp:105 0-15
a=sendrecv
a=ptime:20
a=maxptime:40
a=direction:both

<--- Transmitting SIP response (388 bytes) to UDP:195.185.yy.yy:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.185.yy.yy;rport=5060;received=195.185.yy.yy;branch=z9hG4bKQ7CwRaz.kXdk7
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
From: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
To: <sip:s@sip.easybell.de>
CSeq: 10 INVITE
Server: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Length:  0


    -- Executing [s@incoming:1] NoOp("PJSIP/easybell-endpoint-00000002", "Unbekannte eingehende Nummer, auf Standard weiterleiten") in new stack
    -- Executing [s@incoming:2] Goto("PJSIP/easybell-endpoint-00000002", "incoming,_X.,1") in new stack
    -- Goto (incoming,_X.,1)
    -- Executing [_X.@incoming:1] NoOp("PJSIP/easybell-endpoint-00000002", "Eingehender Anruf auf Easybell") in new stack
    -- Executing [_X.@incoming:2] Answer("PJSIP/easybell-endpoint-00000002", "") in new stack
<--- Transmitting SIP response (917 bytes) to UDP:195.185.yy.yy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.185.yy.yy;rport=5060;received=195.185.yy.yy;branch=z9hG4bKQ7CwRaz.kXdk7
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
From: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
To: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
CSeq: 10 INVITE
Server: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Contact: <sip:192.168.xxx.xxx:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 19 1056709151 IN IP4 192.168.xxx.xxx
s=Asterisk
c=IN IP4 192.168.xxx.xxx
t=0 0
m=audio 14462 RTP/AVP 8 0 100
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP request (452 bytes) from UDP:195.185.yy.yy:5060 --->
ACK sip:192.168.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 195.185.yy.yy;branch=z9hG4bKQ50Cta6rAfilI;rport
From: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
To: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
CSeq: 10 ACK
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
Max-Forwards: 68
Content-Length: 0
Contact: <sip:6DEEECFA-67BC47B80003327C-2C4756C0@195.185.yy.yy;transport=udp>


    -- Executing [_X.@incoming:3] Wait("PJSIP/easybell-endpoint-00000002", "2") in new stack
    -- Executing [_X.@incoming:4] Set("PJSIP/easybell-endpoint-00000002", "PJSIP_HEADER(add,P-Asserted-Identity)=sip:0170xxxxxxx@sip.easybell.de") in new stack
    -- Executing [_X.@incoming:5] Set("PJSIP/easybell-endpoint-00000002", "PJSIP_HEADER(add,Remote-Party-ID)=sip:0170xxxxxxx@sip.easybell.de") in new stack
    -- Executing [_X.@incoming:6] Dial("PJSIP/easybell-endpoint-00000002", "PJSIP/+496950123456@easybell-endpoint,,D(80802020808##w7*123450##),b") in new stack
    -- Called PJSIP/+496950123456@easybell-endpoint
    -- PJSIP/easybell-endpoint-00000002 requested media update control 26, passing it to PJSIP/easybell-endpoint-00000003
<--- Transmitting SIP request (1184 bytes) to UDP:195.185.yy.yy:5060 --->
INVITE sip:6DEEECFA-67BC47B80003327C-2C4756C0@195.185.yy.yy;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPj08b253d5-0849-4a64-8fe3-d40a319320c9
From: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
To: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
Contact: <sip:192.168.xxx.xxx:5060>
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
CSeq: 16844 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:_X.@sip.easybell.de>
Remote-Party-ID: <sip:_X.@sip.easybell.de>;party=called;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 19 1056709152 IN IP4 192.168.xxx.xxx
s=Asterisk
c=IN IP4 192.168.xxx.xxx
t=0 0
m=audio 14462 RTP/AVP 8 0 100
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (404 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPj08b253d5-0849-4a64-8fe3-d40a319320c9;received=109.90.4x.14x
From: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
To: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
CSeq: 16844 INVITE
Content-Length: 0


<--- Transmitting SIP request (1145 bytes) to UDP:195.185.yy.yy:5060 --->
INVITE sip:+496950123456@sip.easybell.de SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPj41401aac-cf2e-4a13-b159-6796c0d4aa1e
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>
Contact: <sip:00496xxx8xxx2@192.168.xxx.xxx:5060>
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21506 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:0170xxxxxxx@sip.easybell.de>
Remote-Party-ID: <sip:0170xxxxxxx@sip.easybell.de>;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Type: application/sdp
Content-Length:   265

v=0
o=- 1577565478 1577565478 IN IP4 192.168.xxx.xxx
s=Asterisk
c=IN IP4 192.168.xxx.xxx
t=0 0
m=audio 17404 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (359 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPj41401aac-cf2e-4a13-b159-6796c0d4aa1e;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21506 INVITE
Content-Length: 0


<--- Received SIP response (659 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPj41401aac-cf2e-4a13-b159-6796c0d4aa1e;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=95c37a12bff1a2c36d72bf8333176544.78550000
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21506 INVITE
P-NGCP-Auth-IP: 192.168.251.xxx
P-NGCP-Auth-UA: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Proxy-Authenticate: Digest realm="sip.easybell.de", nonce="Z7xI5me8R7q0QSAnxvmP39dr1krP/Zpd"
Server: Sipwise NGCP Proxy mr11.5.1
Content-Length: 0


<--- Transmitting SIP request (482 bytes) to UDP:195.185.yy.yy:5060 --->
ACK sip:+496950123456@sip.easybell.de SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPj41401aac-cf2e-4a13-b159-6796c0d4aa1e
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=95c37a12bff1a2c36d72bf8333176544.78550000
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21506 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Length:  0


<--- Transmitting SIP request (1352 bytes) to UDP:195.185.yy.yy:5060 --->
INVITE sip:+496950123456@sip.easybell.de SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPjfb069171-b26c-4f1d-9a19-ea3ff507d2c6
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>
Contact: <sip:00496xxx8xxx2@192.168.xxx.xxx:5060>
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21507 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Proxy-Authorization: Digest username="00496xxx8xxx2", realm="sip.easybell.de", nonce="Z7xI5me8R7q0QSAnxvmP39dr1krP/Zpd", uri="sip:+496950123456@sip.easybell.de", response="8d9216597c88881a564cc9b3cb612d76"
P-Asserted-Identity: <sip:0170xxxxxxx@sip.easybell.de>
Remote-Party-ID: <sip:0170xxxxxxx@sip.easybell.de>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   265

v=0
o=- 1577565478 1577565478 IN IP4 192.168.xxx.xxx
s=Asterisk
c=IN IP4 192.168.xxx.xxx
t=0 0
m=audio 17404 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (359 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPjfb069171-b26c-4f1d-9a19-ea3ff507d2c6;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21507 INVITE
Content-Length: 0


<--- Received SIP response (836 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPj08b253d5-0849-4a64-8fe3-d40a319320c9;received=109.90.4x.14x
From: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
To: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
CSeq: 16844 INVITE
Supported: histinfo, x-diversion
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK
Contact: <sip:6DEEECFA-67BC47B80003327C-2C4756C0@195.185.yy.yy;transport=udp>
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 19 1056709150 IN IP4 195.185.yy.yy
s=-
c=IN IP4 195.185.yy.yy
t=0 0
m=audio 28774 RTP/AVP 8 100
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=sendrecv
a=ptime:20
a=maxptime:40
a=direction:both

<--- Transmitting SIP request (514 bytes) to UDP:195.185.yy.yy:5060 --->
ACK sip:6DEEECFA-67BC47B80003327C-2C4756C0@195.185.yy.yy;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPj885df1a5-71c7-4a5f-b4ec-9febda20bc42
From: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
To: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
CSeq: 16844 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Length:  0


<--- Received SIP response (544 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPjfb069171-b26c-4f1d-9a19-ea3ff507d2c6;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=484E4F7E-67BC47BA000C632A-134F46C0
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21507 INVITE
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, UPDATE
Contact: <sip:68D0FFCE-67BC47BA000B5C6C-ECECE6C0@195.185.yy.yy;transport=udp>
Content-Length: 0


    -- PJSIP/easybell-endpoint-00000003 is making progress passing it to PJSIP/easybell-endpoint-00000002
<--- Received SIP response (535 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPjfb069171-b26c-4f1d-9a19-ea3ff507d2c6;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=484E4F7E-67BC47BA000C632A-134F46C0
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21507 INVITE
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, UPDATE
Contact: <sip:68D0FFCE-67BC47BA000B5C6C-ECECE6C0@195.185.yy.yy;transport=udp>
Content-Length: 0


    -- PJSIP/easybell-endpoint-00000003 is ringing
<--- Received SIP response (871 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPjfb069171-b26c-4f1d-9a19-ea3ff507d2c6;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=484E4F7E-67BC47BA000C632A-134F46C0
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21507 INVITE
Supported: from-change, histinfo, replaces, x-diversion
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, UPDATE
Contact: <sip:68D0FFCE-67BC47BA000B5C6C-ECECE6C0@195.185.yy.yy;transport=udp>
Content-Type: application/sdp
Content-Length: 251

v=0
o=- 572190800 1 IN IP4 195.185.yy.yy
s=-
c=IN IP4 195.185.yy.yy
t=0 0
m=audio 36592 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -
a=direction:both

<--- Transmitting SIP request (508 bytes) to UDP:195.185.yy.yy:5060 --->
ACK sip:68D0FFCE-67BC47BA000B5C6C-ECECE6C0@195.185.yy.yy;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPj7db96b9d-bab7-4977-a98c-d6e0b2de2e12
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=484E4F7E-67BC47BA000C632A-134F46C0
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21507 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Length:  0


    -- PJSIP/easybell-endpoint-00000003 answered PJSIP/easybell-endpoint-00000002
    -- Sending DTMF '80802020808##w7*123450##' to the called party.
    -- Channel PJSIP/easybell-endpoint-00000003 joined 'simple_bridge' basic-bridge <2ee5a921-3710-42a5-8afd-4b83cd3c0e8c>
    -- Channel PJSIP/easybell-endpoint-00000002 joined 'simple_bridge' basic-bridge <2ee5a921-3710-42a5-8afd-4b83cd3c0e8c>
<--- Transmitting SIP request (1159 bytes) to UDP:195.185.yy.yy:5060 --->
INVITE sip:6DEEECFA-67BC47B80003327C-2C4756C0@195.185.yy.yy;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPjb680e12e-ce39-49ce-b9e2-74746be47c61
From: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
To: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
Contact: <sip:192.168.xxx.xxx:5060>
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
CSeq: 16845 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:_X.@sip.easybell.de>
Remote-Party-ID: <sip:_X.@sip.easybell.de>;party=called;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Type: application/sdp
Content-Length:   232

v=0
o=- 19 1056709153 IN IP4 192.168.xxx.xxx
s=Asterisk
c=IN IP4 195.185.yy.yy
t=0 0
m=audio 36592 RTP/AVP 8 100
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Transmitting SIP request (1192 bytes) to UDP:195.185.yy.yy:5060 --->
INVITE sip:68D0FFCE-67BC47BA000B5C6C-ECECE6C0@195.185.yy.yy;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPjd8dc632f-09db-420a-b1e9-0077364a4513
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=484E4F7E-67BC47BA000C632A-134F46C0
Contact: <sip:00496xxx8xxx2@192.168.xxx.xxx:5060>
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21508 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:0170xxxxxxx@sip.easybell.de>
Remote-Party-ID: <sip:0170xxxxxxx@sip.easybell.de>;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Type: application/sdp
Content-Length:   240

v=0
o=- 1577565478 1577565479 IN IP4 192.168.xxx.xxx
s=Asterisk
c=IN IP4 195.185.yy.yy
t=0 0
m=audio 28774 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (398 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPjd8dc632f-09db-420a-b1e9-0077364a4513;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=484E4F7E-67BC47BA000C632A-134F46C0
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21508 INVITE
Content-Length: 0


<--- Received SIP response (404 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPjb680e12e-ce39-49ce-b9e2-74746be47c61;received=109.90.4x.14x
From: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
To: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
CSeq: 16845 INVITE
Content-Length: 0


<--- Received SIP response (878 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPjd8dc632f-09db-420a-b1e9-0077364a4513;received=109.90.4x.14x
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=484E4F7E-67BC47BA000C632A-134F46C0
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21508 INVITE
Supported: from-change, histinfo, replaces, timer, x-diversion
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, UPDATE
Contact: <sip:68D0FFCE-67BC47BA000B5C6C-ECECE6C0@195.185.yy.yy;transport=udp>
Content-Type: application/sdp
Content-Length: 251

v=0
o=- 572190800 2 IN IP4 195.185.yy.yy
s=-
c=IN IP4 195.185.yy.yy
t=0 0
m=audio 36592 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -
a=direction:both

<--- Transmitting SIP request (508 bytes) to UDP:195.185.yy.yy:5060 --->
ACK sip:68D0FFCE-67BC47BA000B5C6C-ECECE6C0@195.185.yy.yy;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPj7d9af00f-6406-401e-b524-7e4e7c39c78d
From: <sip:00496xxx8xxx2@sip.easybell.de>;tag=9df569a7-d57c-4b15-affb-efecb549bac0
To: <sip:+496950123456@sip.easybell.de>;tag=484E4F7E-67BC47BA000C632A-134F46C0
Call-ID: 919aa524-db6f-431a-8672-f540cef6de1a
CSeq: 21508 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Length:  0


<--- Received SIP response (836 bytes) from UDP:195.185.yy.yy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport=62206;branch=z9hG4bKPjb680e12e-ce39-49ce-b9e2-74746be47c61;received=109.90.4x.14x
From: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
To: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
CSeq: 16845 INVITE
Supported: histinfo, x-diversion
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK
Contact: <sip:6DEEECFA-67BC47B80003327C-2C4756C0@195.185.yy.yy;transport=udp>
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 19 1056709151 IN IP4 195.185.yy.yy
s=-
c=IN IP4 195.185.yy.yy
t=0 0
m=audio 28774 RTP/AVP 8 100
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=sendrecv
a=ptime:20
a=maxptime:40
a=direction:both

<--- Transmitting SIP request (514 bytes) to UDP:195.185.yy.yy:5060 --->
ACK sip:6DEEECFA-67BC47B80003327C-2C4756C0@195.185.yy.yy;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPjf1f9f595-cb5a-4962-b25f-18b2ebe5e6db
From: <sip:s@sip.easybell.de>;tag=142b0809-696d-4a56-8ff9-e80611455e34
To: <sip:0170xxxxxxx@sip.easybell.de>;tag=1393A137-67BC47B8000325DD-228C16C0
Call-ID: SD3guqa01-f108e441bb4bb788e6e851244edb27fd-ct68312_b2b-1
CSeq: 16845 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0~dfsg+~cs6.13.40431414-2build5
Content-Length:  0
4

And here is an update of extensions.conf:
I removed real Numbers

[incoming]
exten => _X.,1,NoOp(Eingehender Anruf auf Easybell)
   same => n,Answer()
   same => n,Wait(2)  ; Wartezeit, um sicherzustellen, dass der Anruf stabil ist
   same => n,Set(PJSIP_HEADER(add,P-Asserted-Identity)=sip:${CALLERID(num)}@sip.easybell.de)
   same => n,Set(PJSIP_HEADER(add,Remote-Party-ID)=sip:${CALLERID(num)}@sip.easybell.de)
   same => n,Dial(PJSIP/+4969123123@easybell-endpoint,,D(87654321##w7*123456##),b)
   same => n,Hangup()

exten => s,1,NoOp(Unbekannte eingehende Nummer, auf Standard weiterleiten)
   same => n,Goto(incoming,_X.,1)
5

There is both a P-Asserted-Identity and Remote-Party-ID header in the INVITE to them.

6

This isn’t a particularly sensible thing to do. It should not match any extension you hav.

I don’t understand how _X. got into the PAI and RPID.

Does easybell actually allow you to do this. Most respectable European operators will only allow you to present caller IDs that you have proved to them that you control, as custom caller IDs have been overused by spammers and scammers.

7

I know Easybell and at least with their business trunks you can set your own user provided number on outgoing calls. But you have to activate it in the settings of the sip trunk, by default its not on. Also there you can choose in which header / field you wish to place the number (From-Display, Remote-Party-ID, Prefered Identity oder Asserted Identity).

But you seem to have a different product maybe. Because I see that you are using your phonenumber as from-user. With my easybell trunks it only works when I use the trunk identifier (which is no phone number) as from-user.
So maybe with your product it’s different.

Also keep in mind that you defined a from_user in your pjsip.conf. If you do that, asterisk will always use this user when talking to easybell, regardless what you set as callerid in your dialplan.

8

So that I can understand this better, maybe someone can help me in a simple way with my specific call forwarding case:

I have a number block with Easybell, let’s say 0123-456-00 to 99.
Now, I want to set up call forwarding for the number 0123-456-20 to Zoom and include the room number and password in the Zoom dial-in sequence so that when calling the forwarded number, I am directly connected to Zoom.

Zoom provides the following speed dial number, which can be dialed from mobile devices, for this purpose. However, due to the commas, this does not work with a regular telephone. That’s why I want to set up this forwarding to enable direct access to the conference without any major detours. The speed dial number looks like this:

+496950502596,444555666777#,*777888#

The first part is the official Zoom dial-in number, followed by the room number and the password.

How would you set up this call forwarding? So far, I also have the problem that it does not work completely reliably. I do get forwarded, but sometimes I still have to enter the code manually.

Maybe someone can provide the correct forwarding setup and explain how it is structured.

9

Make sure that the channel is capable of sending DTMF (e.g. don’t try to send inband DTMF over g.729). Then

Dial(PJSIP/+496950502596@trunk,,D(w444555666777#w*777888#)

Also check Zoom’s policies on automated dialling. I believe some services do not allow it.

Adjust the number of w’s to ensure that digits are not sent too early.