[Freeswitch-users] Outgoing calls with a AVM FritzBox 7490

Tue Jun 22 14:55:15 UTC 2021

Hello everyone,

I'm kind of lost with setting up FS to connect to my FritzBox 7490.
The goal is to use FS with spandsp to send and receive Faxes.
Now, most things work in my setup. FS can successfully register as a SIP client
and I can call the configured number from my mobile phoneand the call is
routed to the FS0 virtual modem.

The problem is outgoing calls. Using the ATD command on the virtual modem
to call any external phone does not work. Internal FritzBox-configured numbers
don't work either. Using Wireshark I found that the FritzBox replies with
488 Not Acceptable Here. Looking around the internet, this seems to be
related to the codec configuration. I doubt this is the problem in my case
since enabling more codecs doesn't help and incoming calls with the same
codec restrictions work.

So I'm assuming FS sends an INVITE package that the FritzBox does not like for some reason.
I played around with the From field but that just results in the FritzBox not replying at all.
Does anyone know what might be going on here or is there someone with a working example?
I have attached the INVITE package and the response package. 192.168.0.130 is the FritzBox,
192.168.0.133 is the FS host (with firewall disabled).

Thank you in advance and best regards
Janne

Frame 183405: 1278 bytes on wire (10224 bits), 1278 bytes captured (10224 bits) on interface -, id 0
Ethernet II, Src: RealtekU_ff:ff:ff (52:54:00:ff:ff:ff), Dst: AVMAudio_ff:ff:ff (7c:ff:4d:ff:ff:ff)
Internet Protocol Version 4, Src: 192.168.0.133, Dst: 192.128.0.130
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:0123456789 at 192.168.0.130:5060 SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 192.168.0.133;rport;branch=z9hG4bKcU2Qc6yc0cZ3p
        Max-Forwards: 70
        From: "FSModem" <sip:hylafaxtel at 192.168.0.130>;tag=j2K0XQKgXBFmN
        To: <sip:0123456789 at 192.168.0.130:5060>
        Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216
        [Generated Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216]
        CSeq: 37632110 INVITE
        Contact: <sip:gw+fritzbox at 192.168.0.133:5060;transport=udp;gw=fritzbox>
        User-Agent: FreeSWITCH-mod_sofia/1.10.6-release.12~64bit
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
        Supported: timer, path, replaces
        Allow-Events: talk, hold, conference, refer
        Authorization: Digest username="hylafaxtel", realm="fritz.box", nonce="XXXXX", algorithm=MD5, uri="sip:0123456789 at 192.168.0.130:5060", response="XXXXX"
        Content-Type: application/sdp
        Content-Disposition: session
        Content-Length: 225
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "FSModem" <sip:FS0 at 129.143.6.130>;party=calling;screen=yes;privacy=off
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): FreeSWITCH 1624344691 1624344692 IN IP4 192.168.0.133
            Session Name (s): FreeSWITCH
            Connection Information (c): IN IP4 192.168.0.133
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 27112 RTP/AVP 102 101
            Media Attribute (a): rtpmap:102 L16/8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-16
            Media Attribute (a): ptime:20
            [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216]
            [Generated Call-ID: 751E5AC1C59CE6B0 at 192.168.0.130]
			[I removed most Call-IDs for brevity]

Frame 183406: 837 bytes on wire (6696 bits), 837 bytes captured (6696 bits) on interface -, id 0
Ethernet II, Src: AVMAudio_ff:ff:ff (7c:ff:4d:ff:ff:ff), Dst: RealtekU_ff:ff:ff (52:54:00:ff:ff:ff)
Internet Protocol Version 4, Src: 192.168.0.130, Dst: 192.168.0.133
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (488)
    Status-Line: SIP/2.0 488 Not Acceptable Here
    Message Header
        Via: SIP/2.0/UDP 192.168.0.133;rport=5060;branch=z9hG4bKcU2Qc6yc0cZ3p
        From: "FSModem" <sip:hylafaxtel at 192.168.0.130>;tag=j2K0XQKgXBFmN
        To: <sip:0123456789 at 192.168.0.130:5060>;tag=CDE9429F6B0A3BC3
        Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216
        [Generated Call-ID: 3deaaab4-4e08-123a-ce81-52540059d216]
        CSeq: 37632110 INVITE
        Warning: 399 0.0.0.0 "successful but result empty"
        User-Agent: FRITZ!OS
        Content-Type: application/sdp
        Content-Length:   361
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): user 12041099 12041099 IN IP4 129.143.6.130
            Session Name (s): call
            Connection Information (c): IN IP4 192.168.0.130
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 7080 RTP/AVP 8 0 2 102 100 99 97 101
            Media Attribute (a): sendrecv
            Media Attribute (a): rtpmap:2 G726-32/8000
            Media Attribute (a): rtpmap:102 G726-32/8000
            Media Attribute (a): rtpmap:100 G726-40/8000
            Media Attribute (a): rtpmap:99 G726-24/8000
            Media Attribute (a): rtpmap:97 iLBC/8000
            Media Attribute (a): fmtp:97 mode=30
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15
            Media Attribute (a): rtcp:7081
            [Generated Call-ID: 596BDCD5EDA427D7 at 192.168.0.130]
            [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216]
			[I removed most Call-IDs for brevity]