Sorry my English
I had such problem described above, but after firmware, the problem while has faded.
Software Version: Program - 1.0.1.25 Loader - 1.1.3.4 Boot - 1.1.3.2.
As has noted that FreePBX somehow not correctly sampled the outbound router. That is, in rules of a set one has been pointed router, and he sampled another. I have swapped entering PSTN line on gateway and swapped accounts on it. But the following problem is watched now.
I show the my log:
<— SIP read from 192.168.XXX.23:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.XXX.23:5060;branch=z9hG4bK5373a9f0e84341a7
From: “unknown”;tag=52553eb1d8b41cc4
To:
Contact:
Supported: replaces, timer, path
Call-ID: [email protected]
CSeq: 15259 INVITE
User-Agent: Grandstream GXW4108 (HW 1.0, Ch:6) 1.0.1.25
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 313
v=0
o=41380 8006 8000 IN IP4 192.168.XXX.23
s=SIP Call
c=IN IP4 192.168.XXX.23
t=0 0
m=audio 5028 RTP/AVP 0 8 4 18 3 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
— (13 headers 15 lines) —
Sending to 192.168.XXX.23 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer 'GXW4108p2-41370’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.XXX.23:5028
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xfae (gsm|ulaw|alaw|g726|adpcm|lpc10|g729|speex|ilbc), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - (ulaw|alaw|g729|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.XXX.23:5028
Looking for 7777 in from-trunk-custom (domain 192.168.XXX.1)
list_route: hop:
<— Transmitting (no NAT) to 192.168.XXX.23:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.XXX.23:5060;branch=z9hG4bK5373a9f0e84341a7;received=192.168.XXX.23
From: “unknown”;tag=52553eb1d8b41cc4
To:
Call-ID: [email protected]
CSeq: 15259 INVITE
User-Agent: Roks PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
From my log it is visible that account (v=o; o=41380) one and peer another (Found peer ’ GXW4108p2-41370 ')
Why so?
Please help.
Feb