Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Audiocodes MP124-FXS replying BUSY when lin


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
atis at iq-labs.net
Guest





PostPosted: Mon Mar 10, 2008 1:48 pm    Post subject: [asterisk-users] Audiocodes MP124-FXS replying BUSY when lin Reply with quote

Hello,

Has anybody seen that Audiocodes gateway is replying with "486 Busy
here" when it's actually not (last call ended ~15 seconds ago).

I see this quite often. From other logs i see that previous call ends
at 11:13:01, then app_queue tries to dial at 11:13:14 and fails
numerous times, before succeeding at 11:14:02

I have attached sample SIP debug log:

Any ideas what i could try to change in config to avoid this? It's
config seems huge, maybe anybody has some experience with those
gateways?

Regards,
Atis
-------------------- start of log --------------------

[Mar 10 11:13:14] VERBOSE[30165] logger.c: -- Executing
[28254 at local_dial:70] Dial("Local/28254 at default_queue-e8c3,2",
"SIP/90166|15|gtM(queue_call_answer^28254)") in new stack
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Everything is fine.
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Insert SQL: INSERT INTO channels SET uniqueid = '1205172794.6453',
started = '1205172794', channel = 'SIP/90166-45079a0', state = 'Down',
callerid_num = '', callerid_name = '', accountcode = '', context =
'default-sip', exten = 's', priority = '1', application = '', data =
''
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime: row
inserted on table: channels, id: 0
[Mar 10 11:13:14] DEBUG[30165] chan_sip.c: Call to peer '90166' is 1 out of 8
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Audio is at aa.bb.cc.dd port 47732
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x4 (ulaw) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x2 (gsm) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x8 (alaw) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x10 (g726aal2) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x20 (adpcm) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x40 (slin) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x80 (lpc10) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x400 (ilbc) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding codec 0x800 (g726) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Adding non-codec 0x1
(telephone-event) to SDP
[Mar 10 11:13:14] VERBOSE[30165] logger.c: Reliably Transmitting (NAT)
to ee.ff.gg.hh:5060:
INVITE sip:90166 at ee.ff.gg.hh SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport
From: "28901-----2067217913" <sip:2067217913 at aa.bb.cc.dd>;tag=as18481a04
To: <sip:90166 at ee.ff.gg.hh>
Contact: <sip:2067217913 at aa.bb.cc.dd>
Call-ID: 398ef46a157b71af443414704f2fabde at aa.bb.cc.dd
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Mar 2008 18:13:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 31887 31887 IN IP4 aa.bb.cc.dd
s=session
c=IN IP4 aa.bb.cc.dd
t=0 0
m=audio 47732 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

[Mar 10 11:13:14] VERBOSE[30165] logger.c: -- Called 90166
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Everything is fine.
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Update SQL: UPDATE channels SET callerid_num = '28254', callerid_name
= '', accountcode = '1205172743.6428', con
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Updated 1 rows on table: channels
<--- SIP read from ee.ff.gg.hh:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport
From: "28901-----2067217913" <sip:2067217913 at aa.bb.cc.dd>;tag=as18481a04
To: <sip:90166 at ee.ff.gg.hh>;tag=1c1673732975
Call-ID: 398ef46a157b71af443414704f2fabde at aa.bb.cc.dd
CSeq: 102 INVITE
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004
Content-Length: 0


<------------->
[Mar 10 11:13:14] VERBOSE[31897] logger.c: --- (10 headers 0 lines) ---
[Mar 10 11:13:14] VERBOSE[31897] logger.c:
<--- SIP read from ee.ff.gg.hh:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport
From: "28901-----2067217913" <sip:2067217913 at aa.bb.cc.dd>;tag=as18481a04
To: <sip:90166 at ee.ff.gg.hh>;tag=1c1673732975
Call-ID: 398ef46a157b71af443414704f2fabde at aa.bb.cc.dd
CSeq: 102 INVITE
Contact: <sip:90166 at ee.ff.gg.hh>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004
Reason: Q.850 ;cause=17
Content-Length: 0


<------------->
[Mar 10 11:13:14] VERBOSE[31897] logger.c: --- (12 headers 0 lines) ---
[Mar 10 11:13:14] VERBOSE[31897] logger.c: -- Got SIP response 486
"Busy Here" back from ee.ff.gg.hh
[Mar 10 11:13:14] VERBOSE[31897] logger.c: Transmitting (NAT) to
ee.ff.gg.hh:5060:
ACK sip:90166 at ee.ff.gg.hh SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport
From: "28901-----2067217913" <sip:2067217913 at aa.bb.cc.dd>;tag=as18481a04
To: <sip:90166 at ee.ff.gg.hh>;tag=1c1673732975
Contact: <sip:2067217913 at aa.bb.cc.dd>
Call-ID: 398ef46a157b71af443414704f2fabde at aa.bb.cc.dd
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Mar 10 11:13:14] VERBOSE[30165] logger.c: -- SIP/90166-c45079a0 is busy
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Everything is fine.
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Delete SQL: DELETE FROM channels WHERE uniqueid = '1205172794.6453'
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Deleted 1 rows on table: channels
[Mar 10 11:13:14] DEBUG[30165] chan_sip.c: Call to peer '90166'
removed from call limit 8
[Mar 10 11:13:14] VERBOSE[30165] logger.c: == Everyone is
busy/congested at this time (1:1/0/0)
----------------------------------------- end of log -------------------

--
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services