[Asterisk-Users] rxfax(spandsp-0.0.2pre18) and HT488

[Zen Kato]

Hi,

spandsp-0.0.2pre18 works fine for txfax with HT488(version-1.0.1.2),
but rxfax doesn't work. After some FAX sounds, it hangup!
Could someone tell me how to debug? 

The following is the * CLI> log

 to 192.168.0.161:43222
    -- Executing NoOp("SIP/4881-bde9", "") in new stack
    -- Executing RxFAX("SIP/4881-bde9", "/home/zenkato/voip/asterisk/fax/tif/send22.tif") in new stack


Sip read:
ACK sip:4883 at 192.168.0.3:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.161:43222;branch=z9hG4bK3d3700001a29ffff
From: <sip:4881 at 192.168.0.3:5070>;tag=bdf000008f360000
To: <sip:4883 at 192.168.0.3:5070>;tag=as4090e42f
Contact: <sip:4881 at 192.168.0.161:43222>
Proxy-Authorization: DIGEST username="4881", realm="asterisk", algorithm=MD5, uri="sip:4883 at 192.168.0.3:5070", nonce="05ea8e51", response="04269c40dad3c4c71a9d53e56ef6c790"
Call-ID: 52b500004eb40000 at 192.168.0.161
CSeq: 42398 ACK
User-Agent: Grandstream HT488 1.0.1.2
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


12 headers, 0 lines


Sip read:
BYE sip:4883 at 192.168.0.3:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.161:43222;branch=z9hG4bK746cffff2b140000
From: <sip:4881 at 192.168.0.3:5070>;tag=bdf000008f360000
To: <sip:4883 at 192.168.0.3:5070>;tag=as4090e42f
Proxy-Authorization: DIGEST username="4881", realm="asterisk", algorithm=MD5, uri="sip:4883 at 192.168.0.3:5070", nonce="05ea8e51", response="f18d922cb703582e1f9de0f0e2fd040b"
Call-ID: 52b500004eb40000 at 192.168.0.161
CSeq: 42399 BYE
User-Agent: Grandstream HT488 1.0.1.2
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


11 headers, 0 lines
Sending to 192.168.0.161 : 43222 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.161:43222;branch=z9hG4bK746cffff2b140000
From: <sip:4881 at 192.168.0.3:5070>;tag=bdf000008f360000
To: <sip:4883 at 192.168.0.3:5070>;tag=as4090e42f
Call-ID: 52b500004eb40000 at 192.168.0.161
CSeq: 42399 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4883 at 192.168.0.3:5070>
Content-Length: 0


 to 192.168.0.161:43222
    -- Executing Hangup("SIP/4881-bde9", "") in new stack
  == Spawn extension (sip, h, 1) exited non-zero on 'SIP/4881-bde9'
Destroying call '52b500004eb40000 at 192.168.0.161'

Regards,

Zen