Outgoing calls with a AVM FritzBox 7490
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@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@192.168.0.130>;tag=j2K0XQKgXBFmN To: <sip:0123456789@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@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@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@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@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@192.168.0.130>;tag=j2K0XQKgXBFmN To: <sip:0123456789@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@192.168.0.130] [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216] [I removed most Call-IDs for brevity]
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@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@192.168.0.130>;tag=j2K0XQKgXBFmN To: <sip:0123456789@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@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@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@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@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@192.168.0.130>;tag=j2K0XQKgXBFmN To: <sip:0123456789@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@192.168.0.130] [Generated Call-ID: ba99ae97-4de7-123a-09b4-52540059d216] [I removed most Call-IDs for brevity]
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participants (2)
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Denis Papes
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Janne Heß