bartek@datacomsa.pl
2005-Jul-26 10:23 UTC
[Asterisk-Users] sip+oh323 - no voice at sip side
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,@10.0.0.3-9de1 at voip-h323,224789000,1 failed
so falling back to exten 's'
== Starting OH323/datacom1234,@10.0.0.3-9de1 at voip-h323,s,1 still failed so
falling back to context 'default'
-- Executing Dial("OH323/datacom1234,@10.0.0.3-9de1",
"SIP/48224789000") in new stack
-- Inbound H.323 call 'ip$10.0.0.3:61804/23122', channel
'OH323/datacom1234,@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@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@192.168.0.252>;tag=as195b9c0f
To: <sip:48224789000@172.16.13.169:5060>
Contact: <sip:224782479@192.168.0.252>
Call-ID: 4033d3ab470c99f34839c6985d09f351@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@172.16.13.169:5060>
From: "datacom1234, 224782479 "
<sip:224782479@192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351@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@172.16.13.169:5060>;tag=b857823bdad08738i0
From: "datacom1234, 224782479 "
<sip:224782479@192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351@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@172.16.13.169:5060>;tag=b857823bdad08738i0
From: "datacom1234, 224782479 "
<sip:224782479@192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351@192.168.0.252
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.252:5060;branch=z9hG4bK411177bc
Contact: 48224789000 <sip:48224789000@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@172.16.13.169:5060>
set_destination: Parsing <sip:48224789000@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@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@192.168.0.252>;tag=as195b9c0f
To: <sip:48224789000@172.16.13.169:5060>;tag=b857823bdad08738i0
Contact: <sip:224782479@192.168.0.252>
Call-ID: 4033d3ab470c99f34839c6985d09f351@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,@10.0.0.3-9de1
> H.323 call 'ip$10.0.0.3:61804/23122', exception
CALL_ESTABLISHED.
Destroying call 'b3759d15-77d7b57e@172.16.13.169'
Sip read:
BYE sip:224782479@192.168.0.252 SIP/2.0
Via: SIP/2.0/UDP 172.16.13.169:5060;branch=z9hG4bK-c77a5b50
From: <sip:48224789000@172.16.13.169:5060>;tag=b857823bdad08738i0
To: "datacom1234, 224782479 "
<sip:224782479@192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351@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@172.16.13.169:5060>;tag=b857823bdad08738i0
To: "datacom1234, 224782479 "
<sip:224782479@192.168.0.252>;tag=as195b9c0f
Call-ID: 4033d3ab470c99f34839c6985d09f351@192.168.0.252
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:224782479@192.168.0.252>
Content-Length: 0
to 194.246.106.22:5060
== Spawn extension (default, s, 1) exited non-zero on
'OH323/datacom1234,@10.0.0.3-9de1'
Destroying call '4033d3ab470c99f34839c6985d09f351@192.168.0.252'
-- Hungup 'OH323/datacom1234,@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.
bartek@datacomsa.pl
2005-Jul-27 03:52 UTC
[Asterisk-Users] Re: sip+oh323 - no voice at sip side
On 26-07-2005 at 07:23:39PM +0200, bartek@datacomsa.pl wrote:> Hello, > I have something like this: > SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN >If I call from SIP to PSTN, at the beginning of the call (1 second) after getting phone at the PSTN side I hear voice at the SIP side. After this 1 second I don't hear anything in SIP phone (at the PSTN phone everything is OK). Nobody has had any problems like me? Bartek.
bartek@datacomsa.pl
2005-Jul-28 01:20 UTC
[Asterisk-Users] sip+oh323 - no voice at sip side
On 26-07-2005 at 07:23:39PM +0200, bartek@datacomsa.pl wrote:> 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. >The problem was in oh323.conf: fastStart=no (was yes) Now everything goes fine. Bartek.