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Whenever I dial, I can see only "request timeout #536

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago 
What steps will reproduce the problem?
1. Whenever I dial, I can see only "request timeout"
2. Nimbuzz working
3.

What is the expected output? What do you see instead?
I hope working well

What version of the product are you using? On what operating system?
IOS 4.3.3, Siphon 3.9.17

Please provide any additional information below.

Whenever I dial, I can see available, but when dialing, I can see only request
time out without dialing signal.

Below is my log file

--end msg--
 13:43:04.986 AKSIPUserAgent  Call 0 state changed to CALLING
 13:43:05.029 os_core_unix.c  Info: possibly re-registering existing thread
 13:43:05.486   pjsua_core.c  TX 981 bytes Request msg INVITE/cseq=5531 (tdta0x48e9200) to UDP 220.90.201.200:5060:
INVITE sip:0222431010@proxy.sipasp.kr SIP/2.0
Via: SIP/2.0/UDP 
79.170.5.242:54713;rport;branch=z9hG4bKPjPSSEGvjmV4logYFZmkAoOzr2L6LoP-Ee
Max-Forwards: 70
From: "090XXXXXXXX" 
<sip:090XXXXXXXX@proxy.sipasp.kr>;tag=tCvMuAnCn7hbto7c2l57aygdHbGUXVlb
To: <sip:0222431010@proxy.sipasp.kr>
Contact: <sip:090XXXXXXXX@79.170.5.242:54713;transport=UDP;ob>
Call-ID: Rf.qzB5dJoN0usatnGSPJYewpjTa8gEz
CSeq: 5531 INVITE
Route: <sip:proxy.sipasp.kr;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, 
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Siphon 3.9.17
Content-Type: application/sdp
Content-Length:   277

v=0
o=- 3520924984 3520924984 IN IP4 192.168.10.118
s=pjmedia
c=IN IP4 192.168.10.118
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 18 96
a=rtcp:4001 IN IP4 192.168.10.118
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15

Original issue reported on code.google.com by wonok.P...@gmail.com on 29 Jul 2011 at 11:03

GoogleCodeExporter commented 8 years ago 
Did you have a more complete log file ?
Do you receive an answer from the server?
Can you try to enable more codec?

Original comment by samuelv0...@gmail.com on 29 Jul 2011 at 11:20

GoogleCodeExporter commented 8 years ago 
1. I upload complete log file
2. My account is only for sending, not receving
3. I tried other codec like G711. My sip provided support G729.

Thanks.

Original comment by wonok.P...@gmail.com on 29 Jul 2011 at 11:40

Attachments:

GoogleCodeExporter commented 8 years ago 
So, the incoming calls work fine, but not the outgoing calls.
It's very strange because Siphon sends the INVITE message but it doesn't 
receive anything.
Can you try to enable/disable the NAT option? You find this option in advanced 
settings of account?

Original comment by samuelv0...@gmail.com on 29 Jul 2011 at 11:51

GoogleCodeExporter commented 8 years ago 
Thank you for your response.
I on/off NAT option, but still same.

Original comment by wonok.P...@gmail.com on 30 Jul 2011 at 4:15