@Trezeguet17 wrote:
Hi people,
im havig a real nasty issue , got an extension , a sip extension in another city , where i have the router configurated, all open ports sip and rtp to allow voice trafic, it is registered, but when i call no audio is transmitted, look what i found when i did the call
<--- SIP read from UDP:190.248.187.222:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.250.5:5060;branch=z9hG4bK003e9d60;rport=5060;received=116.48.147.188
From: "Grupo Over-17435737" ;tag=as775bf58f
To: ;tag=819468181
Call-ID: 6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060
CSeq: 102 INVITE
Contact:
Supported: replaces, path, timer
User-Agent: Grandstream GXP1405 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 264v=0
o=1034 8000 8000 IN IP4 192.168.1.1
s=SIP Call
c=IN IP4 192.168.1.1
t=0 0
m=audio 5004 RTP/AVP 0 8 111 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 111
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw|alaw|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.1:5004
list_route: hop:
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.1.1:5060
Transmitting (NAT) to 190.248.187.222:5060:
ACK sip:1034@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.5:5060;branch=z9hG4bK0f2ee1d1;rport
Max-Forwards: 70
From: "Grupo Over-17435737" ;tag=as775bf58f
To: ;tag=819468181
Contact:
Call-ID: 6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.76.4(11.21.2)
Content-Length: 0
<--- SIP read from UDP:190.249.187.222:5060 --->
<------------->
Scheduling destruction of SIP dialog '6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060' in 72960 ms (Method: INVITE)
set_destination: Parsing for address/port to send to
set_destination: set destination to 192.168.1.1:5060
Reliably Transmitting (NAT) to 190.249.187.222:5060:
BYE sip:1034@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.5:5060;branch=z9hG4bK64b13e09;rport
Max-Forwards: 70
From: "Grupo -17435737" ;tag=as775bf58f
To: ;tag=819468181
Call-ID: 6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060
CSeq: 103 BYE
User-Agent: FPBX-12.0.76.4(11.21.2)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<--- SIP read from UDP:190.249.187.222:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.250.5:5060;branch=z9hG4bK64b13e09;rport=5060;received=181.48.147.188
From: "Grupo Over-17435737" ;tag=as775bf58f
To: ;tag=819468181
Call-ID: 6e9cda4c5629634b3c60bb7f66afde8d@192.168.250.5:5060
CSeq: 103 BYE
Contact:
Supported: replaces, path, timer
User-Agent: Grandstream GXP1405 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Posts: 1
Participants: 1