Hi,
I have this simple configuration:
# cat sip.conf
[general]
context=default
allowguest=no
nat=yes
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=alaw
allow=ulaw
t38udptlsupport=yes
dtmfmode=inband
canreinvite=no
[authentication]
[stc-in]
type=friend
host=10.0.0.1
context=stc-in
allow=alaw
allow=ulaw
nat=no
canreinvite=no
dtmfmode=info
insecure=very
# cat extensions.conf
[stc-in]
exten => 2687993,1,Answer()
exten => 2687993,2,Playback(hello-world)
exten => 2687993,3,Hangup()
My local side is last CallWeaver trunk, remote side is HuaweiSoftX3000.
On incoming call from HuaweiSoftX3000 to 2687993 the following happens:
HuaweiSoftX3000 -> CallWeaver : INVITE
CallWeaver -> HuaweiSoftX3000 : Trying
CallWeaver -> HuaweiSoftX3000 : OK
HuaweiSoftX3000 -> CallWeaver : ACK
How connection is established but HuaweiSoftX3000 immediately breaks it
with BYE, so caller can't hear 'hello-world':
HuaweiSoftX3000 -> CallWeaver : BYE
CallWeaver -> HuaweiSoftX3000 : OK
More detailed console output is attached.
This problem looks like HuaweiSoftX3000 problem, because there are no
such problems with many other softswitches. But this problem is
reproducable with Asterisk 1.4.4 running without ztdummy and not
reproducable while ztdummy is loaded. CallWeaver can't use ztdummy, so
it can't work with HuaweiSoftX3000.
How to resolve this problem?
--
Thanks,
Eugene Prokopiev
<-- SIP read from 10.0.0.1:5070:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5070;branch=z9hG4bK6d6b3032b
Call-ID: [EMAIL PROTECTED]
From: <sip:[EMAIL PROTECTED];user=phone>;tag=6d6b3032
To: <sip:[EMAIL PROTECTED];user=phone>
CSeq: 1 INVITE
Contact: <sip:[EMAIL PROTECTED]:5070;user=phone>
Supported: 100rel
Max-Forwards: 70
Allow:
INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER
Content-Length: 254
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 196600 196600 IN IP4 10.0.0.1
s=Sip Call
c=IN IP4 80.254.111.211
t=0 0
m=audio 27638 RTP/AVP 18 4 8 97
a=rtpmap:18 G729/8000a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
--- (12 headers 11 lines) ---
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:4268 sip_alloc: Allocating new
SIP dialog for [EMAIL PROTECTED] - INVITE (With RTP)
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:1150 parse_sip_options: * SIP
extension value: 2 for call [EMAIL PROTECTED]
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 10.0.0.1 : 5070 (NAT)
Found peer 'stc-in'
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:9505 check_user_full: Setting NAT
on RTP to 1
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:9515 check_user_full: Setting NAT
on UDPTL to 1
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 97
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:4789 process_sdp: Activating RTP
on response [EMAIL PROTECTED] (1)
Peer audio RTP is at port 80.254.111.211:27638
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:4860 process_sdp: Peer audio RTP
is at port 80.254.111.211:27638
Found description format G729
Found description format G723
Found description format PCMA
Found description format telephone-event
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:5067 process_sdp: T38 state
changed to 0 on channel <none>
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x109
(g723|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:13283 handle_request_invite:
Checking SIP call limits for device
Looking for 2687993 in stc-in (domain 10.0.0.2;user=phone)
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:8234 build_route: build_route:
Contact hop: <sip:[EMAIL PROTECTED]:5070;user=phone>
list_route: hop: <sip:[EMAIL PROTECTED]:5070;user=phone>
Transmitting (NAT) to 10.0.0.1:5070:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5070;branch=z9hG4bK6d6b3032b;received=10.0.0.1
From: <sip:[EMAIL PROTECTED];user=phone>;tag=6d6b3032
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
May 23 20:00:55 DEBUG[1074288960]: channel.c:784 channel_find_locked: Avoiding
initial deadlock for 'SIP/10.0.0.1-0063ef30'
May 23 20:00:55 DEBUG[1092405568]: pbx.c:2042 pbx_extension_helper: Launching
'Answer'
-- Executing Answer("SIP/10.0.0.1-0063ef30", "") in new stack
May 23 20:00:55 DEBUG[1092405568]: chan_sip.c:3379 sip_answer:
sip_answer(SIP/10.0.0.1-0063ef30)
We're at 10.0.0.2 port 10780
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.0.1:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5070;branch=z9hG4bK6d6b3032b;received=10.0.0.1
From: <sip:[EMAIL PROTECTED];user=phone>;tag=6d6b3032
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as630f8d47
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 27135 27135 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 10780 RTP/AVP 8 0 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
---
May 23 20:00:55 DEBUG[1092405568]: pbx.c:2042 pbx_extension_helper: Launching
'Playback'
-- Executing Playback("SIP/10.0.0.1-0063ef30", "hello-world") in new stack
May 23 20:00:55 DEBUG[1092405568]: generator.c:125 opbx_generator_deactivate:
Trying to deactivate generator in SIP/10.0.0.1-0063ef30
May 23 20:00:55 DEBUG[1092405568]: generator.c:156 opbx_generator_deactivate:
Generator on SIP/10.0.0.1-0063ef30 stopped after 0 iterations
May 23 20:00:55 DEBUG[1092405568]: rtp.c:1767 opbx_rtp_write: Ooh, format
changed from unknown to alaw
-- Playing 'hello-world' (language 'en')
<-- SIP read from 10.0.0.1:5070:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5070;branch=z9hG4bK93ef9c2eb
Call-ID: [EMAIL PROTECTED]
From: <sip:[EMAIL PROTECTED];user=phone>;tag=6d6b3032
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as630f8d47
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
--- (8 headers 0 lines) ---
May 23 20:00:55 DEBUG[1091873088]: chan_sip.c:1603 __sip_ack: Stopping
retransmission on '[EMAIL PROTECTED]' of Response 1: Match Found
<-- SIP read from 10.0.0.1:5070:
BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5070;branch=z9hG4bKb524cc98f
Call-ID: [EMAIL PROTECTED]
From: <sip:[EMAIL PROTECTED];user=phone>;tag=6d6b3032
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as630f8d47
CSeq: 2 BYE
Reason: Q.850;cause=31;text="normal unspecified"
Content-Length: 0
--- (8 headers 0 lines) ---
Sending to 10.0.0.1 : 5070 (NAT)
Transmitting (NAT) to 10.0.0.1:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5070;branch=z9hG4bKb524cc98f;received=10.0.0.1
From: <sip:[EMAIL PROTECTED];user=phone>;tag=6d6b3032
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as630f8d47
Call-ID: [EMAIL PROTECTED]
CSeq: 2 BYE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
May 23 20:00:55 DEBUG[1092405568]: generator.c:125 opbx_generator_deactivate:
Trying to deactivate generator in SIP/10.0.0.1-0063ef30
May 23 20:00:55 DEBUG[1092405568]: generator.c:156 opbx_generator_deactivate:
Generator on SIP/10.0.0.1-0063ef30 stopped after 0 iterations
May 23 20:00:55 DEBUG[1092405568]: pbx.c:2725 __opbx_pbx_run: Spawn extension
(stc-in,2687993,2) exited non-zero on 'SIP/10.0.0.1-0063ef30'
== Spawn extension (stc-in, 2687993, 2) exited non-zero on
'SIP/10.0.0.1-0063ef30'
May 23 20:00:55 DEBUG[1092405568]: cdr.c:972 opbx_cdr_detach: Dropping CDR !
May 23 20:00:55 DEBUG[1092405568]: channel.c:1123 opbx_hangup: Hanging up
channel 'SIP/10.0.0.1-0063ef30'
May 23 20:00:55 DEBUG[1092405568]: chan_sip.c:3227 sip_hangup: Hangup call
SIP/10.0.0.1-0063ef30, SIP callid [EMAIL PROTECTED])
May 23 20:00:55 DEBUG[1092405568]: chan_sip.c:3236 sip_hangup:
update_call_counter() - decrement call limit counter
May 23 20:00:55 DEBUG[1092405568]: channel.c:1137 opbx_hangup: Generator :
deactivate after channel unlock (hangup function)
May 23 20:00:55 DEBUG[1092405568]: generator.c:125 opbx_generator_deactivate:
Trying to deactivate generator in SIP/10.0.0.1-0063ef30
May 23 20:00:55 DEBUG[1092405568]: generator.c:156 opbx_generator_deactivate:
Generator on SIP/10.0.0.1-0063ef30 stopped after 0 iterations
Destroying call '[EMAIL PROTECTED]'
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