HT-488 FXS can't dial out over FXO
Submitted by Atcom Alberta on Thu, 08/07/2008 - 9:25pm.
Here's another HT-488 issue:
I have both the FXO & the FXS side configured
Have flashed the unit with the latest firmware
Using one-stage dialing on HT-488
PBX is trixbox 2.6.1.2
I can make calls over the POTS trunk from other SIP phones - no problems
I can call in on the POTS (FXO) line & answer the call at any SIP phone - including the HT-488 FXS port
I cannot make an outgoing call from the FXS port over the POTS line (FXO port). The call hangs up immediately after being setup.
Here is some of the sip debug:
<--- SIP read from 192.168.101.102:59631 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.101.104:5060;branch=z9hG4bK59c57746;rport
From: "ATA"
To:
Call-ID: 4b87687f61c087797a69de061511833d@192.168.101.104
CSeq: 102 INVITE
User-Agent: Grandstream HT488 1.0.3.96 FXO
Contact:
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 220
v=0
o=HT-488 8000 8000 IN IP4 192.168.101.102
s=SIP Call
c=IN IP4 192.168.101.102
t=0 0
m=audio 47306 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.101.102:47306
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.101.102:47306
list_route: hop:
set_destination: Parsing
set_destination: set destination to 192.168.101.102, port 59631
Transmitting (no NAT) to 192.168.101.102:59631:
ACK sip:HT-488@192.168.101.102:59631 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.104:5060;branch=z9hG4bK023631a6;rport
From: "ATA"
To:
Contact:
Call-ID: 4b87687f61c087797a69de061511833d@192.168.101.104
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/HT-488-08bf2928 answered SIP/226-08b92560
Audio is at 192.168.101.104 port 13426
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
rixbox*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.101.102:11101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.101.102:11101;branch=z9hG4bK46dc38e43edc937c;received=192.168.101.102
From: "ATA"
To:
Call-ID: ac3f7c3f5e3f66df@192.168.101.102
CSeq: 37670 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 2575 2575 IN IP4 192.168.101.104
s=session
c=IN IP4 192.168.101.104
t=0 0
m=audio 13426 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/226-08b92560 and SIP/HT-488-08bf2928
rixbox*CLI>
<--- SIP read from 192.168.101.102:11101 --->
ACK sip:94035310027@192.168.101.104 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.102:11101;branch=z9hG4bK763c335483d030e8
From: "ATA"
To:
Contact:
Proxy-Authorization: Digest username="226", realm="asterisk", algorithm=MD5, uri="sip:94035310027@192.168.101.104", nonce="11f15d5e", response="a3a659fb1ae44c106c3a1cd52d4d576f"
Call-ID: ac3f7c3f5e3f66df@192.168.101.102
CSeq: 37670 ACK
User-Agent: Grandstream HT488 1.0.3.96 FXS
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
rixbox*CLI>
<--- SIP read from 192.168.101.102:59631 --->
BYE sip:4032011968@192.168.101.104 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.102:59631;branch=z9hG4bKe31dc397f64d977d
From:
To: "ATA"
Call-ID: 4b87687f61c087797a69de061511833d@192.168.101.104
CSeq: 51412 BYE
User-Agent: Grandstream HT488 1.0.3.96 FXO
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.101.102 : 59631 (no NAT)
<--- Transmitting (no NAT) to 192.168.101.102:59631 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.101.102:59631;branch=z9hG4bKe31dc397f64d977d;received=192.168.101.102
From:
To: "ATA"
Call-ID: 4b87687f61c087797a69de061511833d@192.168.101.104
CSeq: 51412 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/226-08b92560' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/226-08b92560'
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/226-08b92560", "hangupcall|") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/226-08b92560", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/226-08b92560", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/226-08b92560", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/226-08b92560", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/226-08b92560", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/226-08b92560", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/226-08b92560' in macro 'hangupcall'
Really destroying SIP dialog '4b87687f61c087797a69de061511833d@192.168.101.104' Method: BYE
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/226-08b92560'
Scheduling destruction of SIP dialog 'ac3f7c3f5e3f66df@192.168.101.102' in 32000 ms (Method: ACK)
set_destination: Parsing
set_destination: set destination to 192.168.101.102, port 11101
Reliably Transmitting (no NAT) to 192.168.101.102:11101:
BYE sip:226@192.168.101.102:11101 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.104:5060;branch=z9hG4bK363a29ca;rport
From:
To: "ATA"
Call-ID: ac3f7c3f5e3f66df@192.168.101.102
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
rixbox*CLI>
<--- SIP read from 192.168.101.102:11101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.101.104:5060;branch=z9hG4bK363a29ca;rport
From:
To: "ATA"
Call-ID: ac3f7c3f5e3f66df@192.168.101.102
CSeq: 102 BYE
User-Agent: Grandstream HT488 1.0.3.96 FXS
Contact:
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'ac3f7c3f5e3f66df@192.168.101.102' Method: ACK
rixbox*CLI>
<--- SIP read from 192.168.101.101:29356 --->
Member Since:
2008-07-14