Introduction
I am currently trying to transition from Yate and OpenWrt on one box to FreePBX and OPNsense
firewall on two systems and struggling with the configuration somewhat. Maybe someone has any
idea what goes wrong or is missing. I am a bit overwelmed with all the options TBH.
I will try to structure my post in a way that makes it easier for someone else, or myself in
the future to use this as reference.
Current state
- Provider: Vodafone (Arcor) DSL, public IPv4 address, no IPv6
- FritzBox: Only as DSL modem, does not handle or touch telephony
- OpenWrt Router: Handles telephony via Yate
Yate telephony credentials:
These settings are tested and work. Many other settings here, I am not so sure of.
[acc_sip]
enabled=yes
protocol=sip
username=<area_code><local_number>
description=<area_code><local_number>
interval=600
password=<sip_password>
domain=<area_code>.sip.arcor.de
registrar=<area_code>.sip.arcor.de
localaddress=yes
keepalive=100
<area_code>
here has the format 01234
.
Configuration of the new system
FritzBox
- Setup OPNsense router as exposed host
- Delete any telephony settings
Opnsense configuration
NAT>Port Forwarding
Forwards the WAN SIP and media ports of OPNsense router to the FreePBX box.
-
SIP to PBX
- Interface: WAN
- TCP/IP Version: IPv4
- Protocol: UDP
- Destination: “This Firewall”
- Destination port ranges: SIP (5060) - SIP (5060)
- Redirect target IP: Single host or Network,
<static IPv4 address of FreePBX box>
- Redirect target port: SIP (5060)
- Description: SIP to PBX
-
RTP to PBX
- Interface: WAN
- TCP/IP Version: IPv4
- Protocol: UDP
- Destination: “This Firewall”
- Destination port ranges: 10000 - 20000
- Redirect target IP: Single host or Network,
<static IPv4 address of FreePBX box>
- Redirect target port: 10000
- Description: RTP to PBX
(everything not stated here, leave empty, disabled or default)
NAT>Outbound
Makes sure that RTP ports between PBX and provider are mapped to the same numbers:
-
Mode: Hybrid outbound NAT rule generation
-
PBX UDP static-port
- Interface: WAN
- TCP/IP Version: IPv4
- Protocol: UDP
- Source address: Single host or Network,
<static IPv4 address of FreePBX box>
- Source port: any
- Destination address: any
- Destination port: any
- Translation / target: Interface address
- Static-port: yes
- Description: PBX UDP static-port
(everything not stated here, leave empty, disabled or default)
FreePBX
Since the FreePBX configuration is rather complex and long, these settings might be incomplete.
Settings>Asterisk SIP Settings>General Settings
-
RTP Settings
- RTP Port Ranges: 10000 - 20000
- RTP checksums: yes
- Strict RTP: yes
- RTP Timeout: 30
- RTP Hold Timeout: 30
- RTP Keep Alive: 0
-
Media Transport Settings:
- STUN Server Address: stun.l.google.com:19302
- TURN Server Address: standard.relay.metered.ca:443
- TURN Server Username:
<metered_turn_server_username>
- TURN Server Password:
<metered_turn_server_password>
Settings>Asterisk SIP Settings>SIP Settings [chan_pjsip]
-
Misc PJSip Settings
- Keep Alive Interval: 90
- Taskprocessor Overload Trigger: pjsip_only
-
udp
- udp - 0.0.0.0 - All: yes
-
0.0.0.0 (udp)
- Port to Listen on: 5060
Connectivity>Trunks
-
Add/Edit Trunk>General
- Trunk Name:
<area_code><local_number>
- Outbound CallerID:
<+<country_code><area_code_without_leading_0><local_number>>
- CID Options: Alloy any CID
- Trunk Name:
-
Add/Edit Trunk>pjsip Settings>General
- Username:
<area_code><local_number>
- Auth username:
<area_code><local_number>
- Secret:
<sip_password>
- Authentication: Outbound
- Registration: Send
- Language Code:
<what ever>
- SIP server:
<area_code>.sip.arcor.de
- SIP server port: 5060
- Context: from-pstn
- Transport: 0.0.0.0-udp
- Username:
-
Add/Edit Trunk>pjsip Settings>Advanced
- DTMF Mode: Auto
- Send Line in registration: yes
- Forbidden Retry Interval: 30
- Fatal Retry Interval: 30
- General Retry Interval: 60
- Expiration: 3600
- Max Retries: 10000
- Qualify Frequency: 60
- User=Phone: yes
- Contact User:
<area_code><local_number>
- From Domain:
<area_code>.sip.arcor.de
- From User:
<area_code><local_number>
- Client URI:
sip:<area_code><local_number>@<area_code>.sip.arcor.de
- Server URI:
sip:<area_code>.sip.arcor.de
- RTP Symmetric: yes
- Media Encryption: SRTP via in-SDP
- Force rport: Yes
-
Add/Edit Trunk>pjsip Settings>Codecs
- ulaw
- alaw
- g729
- g726
- gsm
- g722
- h264
- mpeg4
Connectivity>Outbound Routes
- Add/Edit Outgoing>Route Settings
- Route Name: Outgoing
- Route Type: Emergency
- Trunk Sequence for Matched Routes:
<area_code><local_number>
Connectivity>Dial Patterns
- ( ) | [ 00. / ]
- ( ) | [ 01[5-7]. / ]
- ( ) | [ 0N. / ]
- ( <area_code> ) | [ 1. / ]
- ( <area_code> ) | [ Z. / ]
Current issues
- Trying to call my mobile phone from my SIP phone results in: “All circuits are busy now”
Log
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:27] Dial("PJSIP/101-00000014", "PJSIP/<my_mobile_phone_number>@<area_code><local_number>,300,Rb(func-apply-sipheaders^s^1,(1))U(sub-send-obroute-email^<my_mobile_phone_number>^<my_mobile_phone_number>^1^1719052869^^+49<area_code><local_number>)") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] app_stack.c: PJSIP/<area_code><local_number>-00000015 Internal Gosub(func-apply-sipheaders,s,1(1)) start
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:1] NoOp("PJSIP/<area_code><local_number>-00000015", "Applying SIP Headers to channel PJSIP/<area_code><local_number>-00000015") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:2] Set("PJSIP/<area_code><local_number>-00000015", "TECH=PJSIP") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:3] Set("PJSIP/<area_code><local_number>-00000015", "SIPHEADERKEYS=Alert-Info") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:4] While("PJSIP/<area_code><local_number>-00000015", "1") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:5] Set("PJSIP/<area_code><local_number>-00000015", "sipheader=unset") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:6] ExecIf("PJSIP/<area_code><local_number>-00000015", "1?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
[2024-06-22 12:41:09] ERROR[1586487] res_pjsip_header_funcs.c: No headers had been previously added to this session.
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:7] ExecIf("PJSIP/<area_code><local_number>-00000015", "0?Set(sipheader=<http://127.0.0.1>;info=unset)") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:8] ExecIf("PJSIP/<area_code><local_number>-00000015", "0?Set(sipheader=<http://127.0.0.1>unset)") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:9] ExecIf("PJSIP/<area_code><local_number>-00000015", "0?Set(PJSIP_HEADER(add,Alert-Info)=unset)") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:10] EndWhile("PJSIP/<area_code><local_number>-00000015", "") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:4] While("PJSIP/<area_code><local_number>-00000015", "0") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] pbx.c: Executing [s@func-apply-sipheaders:11] Return("PJSIP/<area_code><local_number>-00000015", "") in new stack
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] app_stack.c: Spawn extension (from-pstn, <my_mobile_phone_number>, 1) exited non-zero on 'PJSIP/<area_code><local_number>-00000015'
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] app_stack.c: PJSIP/<area_code><local_number>-00000015 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[2024-06-22 12:41:09] VERBOSE[1666069][C-0000000b] app_dial.c: Called PJSIP/<my_mobile_phone_number>@<area_code><local_number>
[2024-06-22 12:41:09] VERBOSE[1666070] res_pjsip_logger.c: <--- Transmitting SIP response (637 bytes) to UDP:<my_sip_phone_ip>:3072 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP <my_sip_phone_ip>:3072;rport=3072;received=<my_sip_phone_ip>;branch=z9hG4bK-9a9f43f6f3so
Call-ID: 313731393035323836353231313136-vftrrjeux7gg
From: "PBX" <sip:101@<pbx_host_name>>;tag=ig3yemy4dv
To: <sip:<my_mobile_phone_number>@<pbx_host_name>;user=phone>;tag=c143e133-817c-4092-a45e-0760f4a71b62
CSeq: 2 INVITE
Server: FPBX-16.0.40.8(20.8.1)
Contact: <sip:<my_pbx_ip>:5060>
a_code>Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
P-Asserted-Identity: "CID:+49<area_code><local_number>" <sip:<my_mobile_phone_number>@<pbx_host_name>;user=phone>
Content-Length: 0
[2024-06-22 12:41:10] VERBOSE[1586487] res_pjsip_logger.c: <--- Transmitting SIP request (1217 bytes) to UDP:<arcor_sip_ip>:5060 --->
INVITE sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP <my_exteral_ip>:5060;rport;branch=z9hG4bKPja32e7f65-8245-4380-881d-f20f62254dbb
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de;user=phone>;tag=2e2c13d3-c731-4d9d-8e9a-9a63625ee2a9
To: <sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de;user=phone>
Contact: <sip:<area_code><local_number>@<my_exteral_ip>:5060>
Call-ID: 7cfa8e8f-f762-4727-a404-a5b8c2f71a18
CSeq: 31608 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-16.0.40.8(20.8.1)
Content-Type: application/sdp
Content-Length: 471
v=0
o=- 1046302118 1046302118 IN IP4 <my_exteral_ip>
s=Asterisk
c=IN IP4 <my_exteral_ip>
t=0 0
m=audio 14216 RTP/SAVP 0 8 111 9 18 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:2SjEzw/R6f7RqYKBBIpevAh6h1zAZ+GyC4q9xTCX
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
[2024-06-22 12:41:10] VERBOSE[57020] res_pjsip_logger.c: <--- Received SIP response (361 bytes) from UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <my_exteral_ip>:5060;received=<my_exteral_ip>;branch=z9hG4bKPja32e7f65-8245-4380-881d-f20f62254dbb;rport=5060
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de;user=phone>;tag=2e2c13d3-c731-4d9d-8e9a-9a63625ee2a9
To: <sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de;user=phone>
Call-ID: 7cfa8e8f-f762-4727-a404-a5b8c2f71a18
CSeq: 31608 INVITE
[2024-06-22 12:41:10] VERBOSE[57020] res_pjsip_logger.c: <--- Received SIP response (563 bytes) from UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <my_exteral_ip>:5060;received=<my_exteral_ip>;branch=z9hG4bKPja32e7f65-8245-4380-881d-f20f62254dbb;rport=5060
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de;user=phone>;tag=2e2c13d3-c731-4d9d-8e9a-9a63625ee2a9
To: <sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de;user=phone>;tag=1f9d6b7c-0018-0432-0000-0000
Call-ID: 7cfa8e8f-f762-4727-a404-a5b8c2f71a18
CSeq: 31608 INVITE
Server: SSW/0.0.0
Proxy-Authenticate: Digest realm="arcor.de",nonce="6676aa46e883e3a436db70e81fe061c6bc66e6b9",algorithm=MD5
Content-Length: 0
[2024-06-22 12:41:10] VERBOSE[1586487] res_pjsip_logger.c: <--- Transmitting SIP request (482 bytes) to UDP:<arcor_sip_ip>:5060 --->
ACK sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP <my_exteral_ip>:5060;rport;branch=z9hG4bKPja32e7f65-8245-4380-881d-f20f62254dbb
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de;user=phone>;tag=2e2c13d3-c731-4d9d-8e9a-9a63625ee2a9
To: <sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de;user=phone>;tag=1f9d6b7c-0018-0432-0000-0000
Call-ID: 7cfa8e8f-f762-4727-a404-a5b8c2f71a18
CSeq: 31608 ACK
Max-Forwards: 70
User-Agent: FPBX-16.0.40.8(20.8.1)
Content-Length: 0
[2024-06-22 12:41:10] VERBOSE[1586487] res_pjsip_logger.c: <--- Transmitting SIP request (1456 bytes) to UDP:<arcor_sip_ip>:5060 --->
INVITE sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP <my_exteral_ip>:5060;rport;branch=z9hG4bKPj01d2759a-29c2-43db-974e-6dbd79a14542
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de;user=phone>;tag=2e2c13d3-c731-4d9d-8e9a-9a63625ee2a9
To: <sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de;user=phone>
Contact: <sip:<area_code><local_number>@<my_exteral_ip>:5060>
Call-ID: 7cfa8e8f-f762-4727-a404-a5b8c2f71a18
CSeq: 31609 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-16.0.40.8(20.8.1)
Proxy-Authorization: Digest username="<area_code><local_number>", realm="arcor.de", nonce="6676aa46e883e3a436db70e81fe061c6bc66e6b9", uri="sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de:5060;user=phone", response="47edfb402f6f2946acec6dfa12a65fad", algorithm=MD5
Content-Type: application/sdp
Content-Length: 471
v=0
o=- 1046302118 1046302118 IN IP4 <my_exteral_ip>
s=Asterisk
c=IN IP4 <my_exteral_ip>
t=0 0
m=audio 14216 RTP/SAVP 0 8 111 9 18 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:2SjEzw/R6f7RqYKBBIpevAh6h1zAZ+GyC4q9xTCX
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
[2024-06-22 12:41:10] VERBOSE[57020] res_pjsip_logger.c: <--- Received SIP response (361 bytes) from UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <my_exteral_ip>:5060;received=<my_exteral_ip>;branch=z9hG4bKPj01d2759a-29c2-43db-974e-6dbd79a14542;rport=5060
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de;user=phone>;tag=2e2c13d3-c731-4d9d-8e9a-9a63625ee2a9
To: <sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de;user=phone>
Call-ID: 7cfa8e8f-f762-4727-a404-a5b8c2f71a18
CSeq: 31609 INVITE
[2024-06-22 12:41:10] VERBOSE[57020] res_pjsip_logger.c: <--- Received SIP response (811 bytes) from UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <my_exteral_ip>:5060;received=<my_exteral_ip>;branch=z9hG4bKPj01d2759a-29c2-43db-974e-6dbd79a14542;rport=5060
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de;user=phone>;tag=2e2c13d3-c731-4d9d-8e9a-9a63625ee2a9
To: <sip:<my_mobile_phone_number>@<area_code>.sip.arcor.de;user=phone>;tag=1f9d6b7c-0018-0432-0000-0000
Call-ID: 7cfa8e8f-f762-4727-a404-a5b8c2f71a18
CSeq: 31609 INVITE
Supported: resource-priority,timer
Require: timer
Accept: application/sdp
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Contact: <sip:<my_mobile_phone_number>.iIiIiI.ac1f0a9c.@<arcor_sip_ip>:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 92
v=0
o=- 1046302118 1046302118 IN IP4 172.23.134.25
s=IMSS
c=IN IP4 172.23.134.25
t=0 0
[2024-06-22 12:41:10] VERBOSE[1666069][C-0000000b] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
- Calling my number from my mobile phone returns not available
Log
[2024-06-22 13:34:46] VERBOSE[2976][C-00000001] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/101-00000000'
[2024-06-22 13:35:54] VERBOSE[1215] res_pjsip_logger.c: <--- Received SIP request (1042 bytes) from UDP:<arcor_sip_ip>:5060 --->
INVITE sip:<area_code><local_number>@<my_exteral_ip>:5060;line=cjmezgd SIP/2.0
Via: SIP/2.0/UDP <arcor_sip_ip>:5060;branch=z9hG4bKtepaov2000lgdaiv9kr0.1
To: <sip:<area_code><local_number>@muesx001.ngn.vodafone.de;user=phone>
From: <sip:[email protected];user=phone>;tag=133267fe
Call-ID: [email protected]
CSeq: 1 INVITE
Max-Forwards: 56
Contact: <sip:<my_mobile_phone_number>.iIiIiI.ac1f0a99.@<arcor_sip_ip>:5060;transport=udp>
Date: Sat, 22 Jun 2024 13:35:54 GMT
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Supported: resource-priority,timer,100rel
P-Asserted-Identity: <sip:[email protected];user=phone>
P-Asserted-Identity: <tel:+491781477370>
Session-Expires: 1800;refresher=uac
Min-SE: 90
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 176
v=0
o=- 0 0 IN IP4 <arcor_sip_ip>
s=IMSS
c=IN IP4 <arcor_sip_ip>
t=0 0
m=audio 14150 RTP/AVP 9 8 98
b=AS:80
a=rtpmap:98 telephone-event/8000
a=ptime:20
a=maxptime:30
[2024-06-22 13:35:54] VERBOSE[1215] res_pjsip_logger.c: <--- Received SIP request (1042 bytes) from UDP:<arcor_sip_ip>:5060 --->
INVITE sip:<area_code><local_number>@<my_exteral_ip>:5060;line=cjmezgd SIP/2.0
Via: SIP/2.0/UDP <arcor_sip_ip>:5060;branch=z9hG4bKue6fsh300glsulh3cr50.1
To: <sip:<area_code><local_number>@muesx001.ngn.vodafone.de;user=phone>
From: <sip:[email protected];user=phone>;tag=806d5289
Call-ID: [email protected]
CSeq: 1 INVITE
Max-Forwards: 56
Contact: <sip:<my_mobile_phone_number>.iIiIiI.ac1f0a99.@<arcor_sip_ip>:5060;transport=udp>
Date: Sat, 22 Jun 2024 13:35:54 GMT
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Supported: resource-priority,timer,100rel
P-Asserted-Identity: <sip:[email protected];user=phone>
P-Asserted-Identity: <tel:+491781477370>
Session-Expires: 1800;refresher=uac
Min-SE: 90
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 176
v=0
o=- 0 0 IN IP4 <arcor_sip_ip>
s=IMSS
c=IN IP4 <arcor_sip_ip>
t=0 0
m=audio 10018 RTP/AVP 9 8 98
b=AS:80
a=rtpmap:98 telephone-event/8000
a=ptime:20
a=maxptime:30
[2024-06-22 13:35:54] VERBOSE[1216] res_pjsip_logger.c: <--- Transmitting SIP response (387 bytes) to UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <arcor_sip_ip>:5060;rport=5060;received=<arcor_sip_ip>;branch=z9hG4bKtepaov2000lgdaiv9kr0.1
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=133267fe
To: <sip:<area_code><local_number>@muesx001.ngn.vodafone.de;user=phone>
CSeq: 1 INVITE
Server: FPBX-16.0.40.8(20.8.1)
Content-Length: 0
[2024-06-22 13:35:54] ERROR[1216] res_pjsip_session.c: <area_code><local_number>: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
[2024-06-22 13:35:54] VERBOSE[1216] res_pjsip_logger.c: <--- Transmitting SIP response (441 bytes) to UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP <arcor_sip_ip>:5060;rport=5060;received=<arcor_sip_ip>;branch=z9hG4bKtepaov2000lgdaiv9kr0.1
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=133267fe
To: <sip:<area_code><local_number>@muesx001.ngn.vodafone.de;user=phone>;tag=7614ba79-ab52-4ca8-b016-415e9d79a369
CSeq: 1 INVITE
Server: FPBX-16.0.40.8(20.8.1)
Content-Length: 0
[2024-06-22 13:35:54] VERBOSE[1216] res_pjsip_logger.c: <--- Transmitting SIP response (387 bytes) to UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <arcor_sip_ip>:5060;rport=5060;received=<arcor_sip_ip>;branch=z9hG4bKue6fsh300glsulh3cr50.1
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=806d5289
To: <sip:<area_code><local_number>@muesx001.ngn.vodafone.de;user=phone>
CSeq: 1 INVITE
Server: FPBX-16.0.40.8(20.8.1)
Content-Length: 0
[2024-06-22 13:35:54] ERROR[1216] res_pjsip_session.c: <area_code><local_number>: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
[2024-06-22 13:35:54] VERBOSE[1216] res_pjsip_logger.c: <--- Transmitting SIP response (441 bytes) to UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP <arcor_sip_ip>:5060;rport=5060;received=<arcor_sip_ip>;branch=z9hG4bKue6fsh300glsulh3cr50.1
Call-ID: [email protected]
From: <sip:[email protected];user=phone>;tag=806d5289
To: <sip:<area_code><local_number>@muesx001.ngn.vodafone.de;user=phone>;tag=9a75f49a-e484-4f41-be1c-e709958368a5
CSeq: 1 INVITE
Server: FPBX-16.0.40.8(20.8.1)
Content-Length: 0
[2024-06-22 13:35:54] VERBOSE[1215] res_pjsip_logger.c: <--- Received SIP request (416 bytes) from UDP:<arcor_sip_ip>:5060 --->
ACK sip:<area_code><local_number>@<my_exteral_ip>:5060;line=cjmezgd SIP/2.0
Via: SIP/2.0/UDP <arcor_sip_ip>:5060;branch=z9hG4bKtepaov2000lgdaiv9kr0.1
CSeq: 1 ACK
To: <sip:<area_code><local_number>@muesx001.ngn.vodafone.de;user=phone>;tag=7614ba79-ab52-4ca8-b016-415e9d79a369
From: <sip:[email protected];user=phone>;tag=133267fe
Call-ID: [email protected]
Max-Forwards: 56
Content-Length: 0
[2024-06-22 13:35:54] VERBOSE[1215] res_pjsip_logger.c: <--- Received SIP request (416 bytes) from UDP:<arcor_sip_ip>:5060 --->
ACK sip:<area_code><local_number>@<my_exteral_ip>:5060;line=cjmezgd SIP/2.0
Via: SIP/2.0/UDP <arcor_sip_ip>:5060;branch=z9hG4bKue6fsh300glsulh3cr50.1
CSeq: 1 ACK
To: <sip:<area_code><local_number>@muesx001.ngn.vodafone.de;user=phone>;tag=9a75f49a-e484-4f41-be1c-e709958368a5
From: <sip:[email protected];user=phone>;tag=806d5289
Call-ID: [email protected]
Max-Forwards: 56
Content-Length: 0
[2024-06-22 13:36:01] VERBOSE[1216] res_pjsip_logger.c: <--- Transmitting SIP request (473 bytes) to UDP:<arcor_sip_ip>:5060 --->
OPTIONS sip:<area_code><local_number>@<area_code>.sip.arcor.de:5060 SIP/2.0
Via: SIP/2.0/UDP <my_exteral_ip>:5060;rport;branch=z9hG4bKPjd4a8c17d-d03a-4462-b871-7abea81bcf00
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de>;tag=acdc9839-6e72-4d54-94d5-a8af5accdea2
To: <sip:<area_code><local_number>@<area_code>.sip.arcor.de>
Contact: <sip:<area_code><local_number>@<my_exteral_ip>:5060>
Call-ID: 93467a68-4726-4ce8-9694-e33fba1b9cc3
CSeq: 33982 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.8(20.8.1)
Content-Length: 0
[2024-06-22 13:36:01] VERBOSE[1215] res_pjsip_logger.c: <--- Received SIP response (579 bytes) from UDP:<arcor_sip_ip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <my_exteral_ip>:5060;received=<my_exteral_ip>;branch=z9hG4bKPjd4a8c17d-d03a-4462-b871-7abea81bcf00;rport=5060
From: <sip:<area_code><local_number>@<area_code>.sip.arcor.de>;tag=acdc9839-6e72-4d54-94d5-a8af5accdea2
To: <sip:<area_code><local_number>@<area_code>.sip.arcor.de>;tag=df8e40d1-0015-028b-0000-0000
Call-ID: 93467a68-4726-4ce8-9694-e33fba1b9cc3
CSeq: 33982 OPTIONS
Accept: application/sdp
Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE
Contact: <sip:[email protected]:5060>
Content-Length: 0
[2024-06-22 13:36:22] VERBOSE[1216] res_pjsip_logger.c: <--- Transmitting SIP request (438 bytes) to UDP:<my_sip_phone_ip>:3072 --->
OPTIONS sip:101@<my_sip_phone_ip>:3072;line=lamztina SIP/2.0
Via: SIP/2.0/UDP <my_pbx_ip>:5060;rport;branch=z9hG4bKPj1111b6f8-5f87-472c-b03f-d9eb19e2c9b1
From: <sip:101@<my_pbx_ip>>;tag=0a85410b-9b85-422a-90a3-38968fdb4a71
To: <sip:101@<my_sip_phone_ip>;line=lamztina>
Contact: <sip:101@<my_pbx_ip>:5060>
Call-ID: 8fe6c56a-2244-426b-b3ce-589af424a943
CSeq: 9851 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.8(20.8.1)
Content-Length: 0
[2024-06-22 13:36:22] VERBOSE[1215] res_pjsip_logger.c: <--- Received SIP response (654 bytes) from UDP:<my_sip_phone_ip>:3072 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <my_pbx_ip>:5060;rport=5060;branch=z9hG4bKPj1111b6f8-5f87-472c-b03f-d9eb19e2c9b1
From: <sip:101@<my_pbx_ip>>;tag=0a85410b-9b85-422a-90a3-38968fdb4a71
To: <sip:101@<my_sip_phone_ip>;line=lamztina>;tag=x8i51qhwvp
Call-ID: 8fe6c56a-2244-426b-b3ce-589af424a943
CSeq: 9851 OPTIONS
User-Agent: snom821/8.7.5.35
Contact: <sip:101@<my_sip_phone_ip>:3072;line=lamztina>;reg-id=1
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0
[2024-06-22 13:36:34] VERBOSE[1215] res_pjsip_logger.c: <--- Received SIP request (744 bytes) from UDP:<my_sip_phone_ip>:3072 --->
REGISTER sip:<pbx_host_name> SIP/2.0
Via: SIP/2.0/UDP <my_sip_phone_ip>:3072;branch=z9hG4bK-8jwnucom7wlz;rport
From: "PBX" <sip:101@<pbx_host_name>>;tag=g5c4cklunv
To: "PBX" <sip:101@<pbx_host_name>>
Call-ID: 313731393034333530373238363332-epmgnwbvkwvk
CSeq: 19 REGISTER
Max-Forwards: 70
User-Agent: snom821/8.7.5.35
Contact: <sip:101@<my_sip_phone_ip>:3072;line=lamztina>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:3fd31d7e-061a-42a5-8db4-000413452EEF>";audio;mobility="fixed";duplex="full";description="snom821";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
Allow-Events: dialog
X-Real-IP: <my_sip_phone_ip>
Supported: path, gruu
Expires: 3600
Content-Length: 0
[2024-06-22 13:36:34] VERBOSE[1216] res_pjsip_logger.c: <--- Transmitting SIP response (520 bytes) to UDP:<my_sip_phone_ip>:3072 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <my_sip_phone_ip>:3072;rport=3072;received=<my_sip_phone_ip>;branch=z9hG4bK-8jwnucom7wlz
Call-ID: 313731393034333530373238363332-epmgnwbvkwvk
From: "PBX" <sip:101@<pbx_host_name>>;tag=g5c4cklunv
To: "PBX" <sip:101@<pbx_host_name>>;tag=z9hG4bK-8jwnucom7wlz
CSeq: 19 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1719056194/5ddf609a89d2605e378aaab364e20c19",opaque="118c948807ad7f2d",algorithm=MD5,qop="auth"
Server: FPBX-16.0.40.8(20.8.1)
Content-Length: 0
[2024-06-22 13:36:34] VERBOSE[1215] res_pjsip_logger.c: <--- Received SIP request (1005 bytes) from UDP:<my_sip_phone_ip>:3072 --->
REGISTER sip:<pbx_host_name> SIP/2.0
Via: SIP/2.0/UDP <my_sip_phone_ip>:3072;branch=z9hG4bK-1o497lglx584;rport
From: "PBX" <sip:101@<pbx_host_name>>;tag=g5c4cklunv
To: "PBX" <sip:101@<pbx_host_name>>
Call-ID: 313731393034333530373238363332-epmgnwbvkwvk
CSeq: 20 REGISTER
Max-Forwards: 70
User-Agent: snom821/8.7.5.35
Contact: <sip:101@<my_sip_phone_ip>:3072;line=lamztina>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:3fd31d7e-061a-42a5-8db4-000413452EEF>";audio;mobility="fixed";duplex="full";description="snom821";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO"
Allow-Events: dialog
X-Real-IP: <my_sip_phone_ip>
Supported: path, gruu
Authorization: Digest username="101",realm="asterisk",nonce="1719056194/5ddf609a89d2605e378aaab364e20c19",uri="sip:<pbx_host_name>",qop=auth,nc=00000001,cnonce="539dbebd",response="13aa35358c60ab09c5310739e6692ad6",opaque="118c948807ad7f2d",algorithm=MD5
Expires: 3600
Content-Length: 0
[2024-06-22 13:36:34] VERBOSE[1216] res_pjsip_logger.c: <--- Transmitting SIP response (480 bytes) to UDP:<my_sip_phone_ip>:3072 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <my_sip_phone_ip>:3072;rport=3072;received=<my_sip_phone_ip>;branch=z9hG4bK-1o497lglx584
Call-ID: 313731393034333530373238363332-epmgnwbvkwvk
From: "PBX" <sip:101@<pbx_host_name>>;tag=g5c4cklunv
To: "PBX" <sip:101@<pbx_host_name>>;tag=z9hG4bK-1o497lglx584
CSeq: 20 REGISTER
Date: Sat, 22 Jun 2024 11:36:34 GMT
Contact: <sip:101@<my_sip_phone_ip>:3072;line=lamztina>;expires=3599
Expires: 3600
Server: FPBX-16.0.40.8(20.8.1)
Content-Length: 0