[Asterisk-Users] sip+oh323 - no voice at sip side

Tue Jul 26 10:23:39 MST 2005

Hello,
I have something like this:
SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN

After calling from SIP to PSTN (and from PSTN to SIP too)
I can't hear anything only in my SIPUSER. At the PSTN side everything is OK.

I have another network with another h323/sip (in the place of asterisk)
and there everything is OK.

In AUDIOCODES logs I see that everything goes fine with asterisk, but SIPUSER
can't hear the PSTN user.

sip.conf:
[general]
disallow=all

[48224789000]
type=friend
username=48224789000
secret=xxxxxx
host=dynamic
nat=yes
qualify=100
disallow=all
allow=g729
context=intern

Here is sip debug:
SIP Debugging Enabled
    -- Inbound H.323 call 'ip$10.0.0.3:61804/23122' detected.
  == Starting OH323/datacom1234, at 10.0.0.3-9de1 at voip-h323,224789000,1 failed so falling back to exten 's'
  == Starting OH323/datacom1234, at 10.0.0.3-9de1 at voip-h323,s,1 still failed so falling back to context 'default'
    -- Executing Dial("OH323/datacom1234, at 10.0.0.3-9de1", "SIP/48224789000") in new stack
    -- Inbound H.323 call 'ip$10.0.0.3:61804/23122', channel 'OH323/datacom1234, at 10.0.0.3-9de1'.
We're at 192.168.0.252 port 15278
Answering/Requesting with root capability 0x100 (g729)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:48224789000 at 172.16.13.169:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc;rport
From: "datacom1234, 224782479 " <sip:224782479 at 192.168.0.252>;tag=as195b9c0f
To: <sip:48224789000 at 172.16.13.169:5060>
Contact: <sip:224782479 at 192.168.0.252>
Call-ID: 4033d3ab470c99f34839c6985d09f351 at 192.168.0.252
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 26 Jul 2005 17:18:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 224

v=0
o=root 15298 15298 IN IP4 192.168.0.252
s=session
c=IN IP4 192.168.0.252
t=0 0
m=audio 15278 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to 194.246.106.22:5060
    -- Called 48224789000

Sip read:
SIP/2.0 100 Trying
To: <sip:48224789000 at 172.16.13.169:5060>
From: "datacom1234, 224782479 " <sip:224782479 at 192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351 at 192.168.0.252
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc
Server: Sipura/SPA1000-2.0.13(g)
Content-Length: 0

8 headers, 0 lines

Sip read:
SIP/2.0 180 Ringing
To: <sip:48224789000 at 172.16.13.169:5060>;tag=b857823bdad08738i0
From: "datacom1234, 224782479 " <sip:224782479 at 192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351 at 192.168.0.252
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc
Server: Sipura/SPA1000-2.0.13(g)
Content-Length: 0

8 headers, 0 lines
    -- SIP/48224789000-8290 is ringing

Sip read:
SIP/2.0 200 OK
To: <sip:48224789000 at 172.16.13.169:5060>;tag=b857823bdad08738i0
From: "datacom1234, 224782479 " <sip:224782479 at 192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351 at 192.168.0.252
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc
Contact: 48224789000 <sip:48224789000 at 172.16.13.169:5060>
Server: Sipura/SPA1000-2.0.13(g)
Content-Length: 236
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 59832 59832 IN IP4 172.16.13.169
s=-
c=IN IP4 172.16.13.169
t=0 0
m=audio 16418 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

12 headers, 12 lines
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 172.16.13.169:16418
Found description format G729a
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:48224789000 at 172.16.13.169:5060>
set_destination: Parsing <sip:48224789000 at 172.16.13.169:5060> for address/port to send to
set_destination: set destination to 172.16.13.169, port 5060
Transmitting:
ACK sip:48224789000 at 172.16.13.169:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK6bacf8a7;rport
From: "datacom1234, 224782479 " <sip:224782479 at 192.168.0.252>;tag=as195b9c0f
To: <sip:48224789000 at 172.16.13.169:5060>;tag=b857823bdad08738i0
Contact: <sip:224782479 at 192.168.0.252>
Call-ID: 4033d3ab470c99f34839c6985d09f351 at 192.168.0.252
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 194.246.106.22:5060
    -- SIP/48224789000-8290 answered OH323/datacom1234, at 10.0.0.3-9de1
       > H.323 call 'ip$10.0.0.3:61804/23122', exception CALL_ESTABLISHED.
Destroying call 'b3759d15-77d7b57e at 172.16.13.169'

Sip read:
BYE sip:224782479 at 192.168.0.252 SIP/2.0
Via: SIP/2.0/UDP 172.16.13.169:5060;branch=z9hG4bK-c77a5b50
From: <sip:48224789000 at 172.16.13.169:5060>;tag=b857823bdad08738i0
To: "datacom1234, 224782479 " <sip:224782479 at 192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351 at 192.168.0.252
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Sipura/SPA1000-2.0.13(g)
Content-Length: 0

9 headers, 0 lines
Sending to 172.16.13.169 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.13.169:5060;branch=z9hG4bK-c77a5b50;received=194.246.106.22;rport=5060
From: <sip:48224789000 at 172.16.13.169:5060>;tag=b857823bdad08738i0
To: "datacom1234, 224782479 " <sip:224782479 at 192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351 at 192.168.0.252
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:224782479 at 192.168.0.252>
Content-Length: 0

 to 194.246.106.22:5060
  == Spawn extension (default, s, 1) exited non-zero on 'OH323/datacom1234, at 10.0.0.3-9de1'
Destroying call '4033d3ab470c99f34839c6985d09f351 at 192.168.0.252'
    -- Hungup 'OH323/datacom1234, at 10.0.0.3-9de1'
sip no debug
SIP Debugging Disabled

IMHO the problem is somewhere in the sip parameters.
But where???

Best regards,
	Bartek.