[Freeswitch-users] Incoming call ignored by SIP client

Florent Krieg flokrrr at gmail.com
Fri Oct 10 16:03:24 MSD 2014


Hi Gavin,

Could this be a firewall issue on the client side (is 5060 open)?
When doing tcpdump, do you see any provisional reply from the client? No
packet filtering could help.

Do you see any log on the client side? (if no, then the INVITE might not
reach the application layer, if yes, it could help to investigate)

Florent


2014-10-10 12:11 GMT+02:00 Gavin Hamill <gavin at bashton.com>:

> Hi all,
>
> I'm running FS 1.4.9 on a CentOS 6.5 machine with a single NIC. The
> machine is behind NAT, and I have a working VPN so that internal calls
> work. The local IP of the machine is 192.168.196.101, and the remote VPN
> subnet (no NAT - clear routing) is 192.168.41.0/24.
>
> I am using port 5080 for internal calls (auth-calls=true) and port 5060
> for incoming calls from a SIP provider. The machine has an internal IP and
> port-forwarding handles public IP -> private IP.
>
> When I dial from a mobile phone to a number on the external SIP provider,
> I do get activity on the console:
>
> 2014-10-10 10:52:20.655471 [NOTICE] switch_channel.c:1055 New Channel
> sofia/external/07951357XXX at 87.238.73.181
> [78cc73d0-e9d1-4d7d-9f22-f7adac8ba0c0]
> 2014-10-10 10:52:21.259045 [NOTICE] switch_ivr.c:1844 Transfer
> sofia/external/07951357xxx at 87.238.73.181 to XML[106 at default]
> 2014-10-10 10:52:21.259045 [WARNING] mod_dptools.c:1628 incoming call from
> [07951357XXX] to [106]
> 2014-10-10 10:52:21.259045 [NOTICE] mod_sofia.c:2237 Pre-Answer
> sofia/external/07951357XXX at 87.238.73.181!
> 2014-10-10 10:52:21.259045 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal/sip:106 at 192.168.41.198:5060
> [a7aaa63b-fd85-4b1f-8855-180a218a388e]
>
> At this point there is a pause and a timeout after 30 seconds.  There is a
> phone registered to extension '106' listening at that IP address on port
> 5060:
>
> freeswitch at internal> sofia status profile internal reg
>
> Registrations:
>
> =================================================================================================
> Call-ID:     1572302163 at 192.168.41.198
> User:       106 at freeswitch.bashton.eu
> Contact:     "" <sip:106 at 192.168.41.198:5060>
> Agent:       qutecom/rev-g-trunk
> Status:     Registered(UDP)(unknown) EXP(2014-10-10 10:32:04)
> EXPSECS(2796)
> Host:       freeswitch
> IP:         192.168.41.198
> Port:       5060
> Auth-User:   106
> Auth-Realm: freeswitch.bashton.eu
> MWI-Account: 106 at freeswitch.bashton.eu
>
> Total items returned: 1
>
> =================================================================================================
>
> freeswitch at internal> sofia status profile external
>
> =================================================================================================
> Name             external
> Domain Name       N/A
> Auto-NAT         false
> DBName           sofia_reg_external
> Pres Hosts
> Dialplan         XML
> Context           public
> Challenge Realm   auto_to
> RTP-IP           192.168.196.101
> SIP-IP           192.168.196.101
> URL               sip:mod_sofia at 192.168.196.101:5060
> BIND-URL         sip:mod_sofia at 192.168.196.101:5060;transport=udp,tcp
> HOLD-MUSIC       N/A
> OUTBOUND-PROXY   N/A
> CODECS IN         PCMU,PCMA
> CODECS OUT
> TEL-EVENT         101
> DTMF-MODE         rfc2833
> CNG               13
> SESSION-TO       0
> MAX-DIALOG       0
> NOMEDIA           false
> LATE-NEG         false
> PROXY-MEDIA       false
> ZRTP-PASSTHRU     false
> AGGRESSIVENAT     false
> CALLS-IN         2
> FAILED-CALLS-IN   1
> CALLS-OUT         0
> FAILED-CALLS-OUT 0
> REGISTRATIONS     0
>
>
>
> freeswitch at internal> sofia status profile internal
>
> =================================================================================================
> Name             internal
> Domain Name       N/A
> Auto-NAT         false
> DBName           sofia_reg_internal
> Pres Hosts
> Dialplan         XML
> Context           default
> Challenge Realm   auto_to
> RTP-IP           192.168.196.101
> Ext-RTP-IP       192.168.196.101
> SIP-IP           192.168.196.101
> Ext-SIP-IP       192.168.196.101
> URL               sip:mod_sofia at 192.168.196.101:5080
> BIND-URL         sip:mod_sofia at 192.168.196.101:5080
> ;maddr=192.168.196.101;transport=udp
> HOLD-MUSIC       N/A
> OUTBOUND-PROXY   N/A
> CODECS IN         PCMU,PCMA
> CODECS OUT       PCMU,PCMA
> TEL-EVENT         101
> DTMF-MODE         rfc2833
> CNG               13
> SESSION-TO       0
> MAX-DIALOG       0
> NOMEDIA           false
> LATE-NEG         false
> PROXY-MEDIA       false
> ZRTP-PASSTHRU     false
> AGGRESSIVENAT     false
> CALLS-IN         0
> FAILED-CALLS-IN   0
> CALLS-OUT         2
> FAILED-CALLS-OUT 2
> REGISTRATIONS     1
>
> freeswitch at internal> sofia status
>                      Name   Type                                      Data
> State
>
> =================================================================================================
>                  external profile
> sip:mod_sofia at 192.168.196.101:5060 RUNNING (0)
>       external::magrathea gateway
> sip:bashton at sipgw.magrathea.net REGED
>                  internal profile
> sip:mod_sofia at 192.168.196.101:5080 RUNNING (0)
>
> =================================================================================================
> 2 profiles 0 aliases
>
>
> The SIP client is definitely listening:
>
> gdh at gdh-e7440:~$ netstat -anp | grep 5060
> tcp        0      0 192.168.41.198:5060     0.0.0.0:*
> LISTEN      17566/qutecom
> udp        0      0 192.168.41.198:5060     0.0.0.0:*
>       17566/qutecom
>
> And I can see that when the call comes in, there is traffic received by
> the SIP client (tcpdump + wireshark):
>
> INVITE sip:106 at 192.168.41.198:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.196.101:5080;rport;branch=z9hG4bKX71j6K414rQNa
> Max-Forwards: 67
> From: "None
> " <sip:07951357XXX at freeswitch.bashton.eu>;tag=rBeB1Nv1Bampm
> To: <sip:106 at 192.168.41.198:5060>
> Call-ID: f850b9a6-cb05-1232-7693-02b891c191f6
> CSeq: 66131306 INVITE
> Contact: <sip:mod_sofia at 192.168.196.101:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.4.9~64bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 253
> X-CALLINFO: cdr=5437AC53=94F6B14;
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "None
> " <sip:07951357XXX at freeswitch.bashton.eu
> >;party=calling;screen=yes;privacy=off
>
>
> The incoming call is handled in the public context with:
>
>        <action application="transfer" data="106 XML default"/>
>
> and in extension 106 in the default context, there's this:
>
>         <action application="bridge" data="user/$1"/>
>
> I have tried both qutecom and linphone with the same behaviour. Any ideas?
>
> Cheers,
> Gavin.
>
>
>
> _________________________________________________________________________
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> 
> 
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