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

Denis Papes denis.papes at shishko.eu
Wed Jun 23 06:44:20 UTC 2021


> 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.


But it is codec issue. In your case, FreeSWITCH calls using L16, while
FritzBox does not support that

FreeSWITCH

> Media Attribute (a): rtpmap:102 L16/8000
FritzBox
> 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

On 22/06/2021 15:55, Janne Heß wrote:
> 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]
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
A non-text attachment was scrubbed...
Name: OpenPGP_signature
Type: application/pgp-signature
Size: 840 bytes
Desc: OpenPGP digital signature
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210623/1362700f/attachment-0001.sig>


More information about the FreeSWITCH-users mailing list