Patton SN4638/5BIS - Problem mit ausgehenden Anrufen

zelos · December 31, 2010, 5:19am

Hallo Leute,
Bei der Inbetriebnahme des Patton Gateway treten folgende Probleme auf.

Verzögerung des Anrufs, wenn ich eine Rufnummer wähle dann kann es bis zu 7 Sek. dauern dass ich endlich mal ein Wähl ton höre.

Ich weis leider nicht wie lange das sonst dauert aber wenn ich das mit einem SIP Account vergleiche dann sind das ca. 6 Sek. zu viel

Ca. von 10 Ausgehenden anrufen schlagen 30% mit der Meldung “Service Unavailible” fehl.
Hat jemand eine Idee?
Anbei noch das Debugging…

v=0
o=root 35237456 35237456 IN IP4 192.168.60.17
s=call
c=IN IP4 192.168.60.17
t=0 0
m=audio 63428 RTP/AVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:e9xixfmtZ8OpdfXe3ZBp7SwOPYXcvoTuvR+dz/ey
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

— (20 headers 19 lines) —
Using INVITE request as basis request - 3c278ffc7468-hewyujzauo5g
Sending to 192.168.60.17 : 5060 (NAT)
Found user ‘213’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 99
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 192.168.60.17:63428
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 05572000000 in internal (domain 192.168.60.253)
list_route: hop: <sip:213@192.168.60.17:5060>
Transmitting (no NAT) to 192.168.60.17:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.60.17:5060;branch=z9hG4bK-qirlra95qc81;received=192.168.60.17;rport=5060
From: “Alexander Klien” <sip:213@192.168.60.253>;tag=k32egfzbbt
To: <sip:05572000000@192.168.60.253>
Call-ID: 3c278ffc7468-hewyujzauo5g
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:05572000000@192.168.60.253>
Content-Length: 0

-- Executing Set("SIP/213-081fe908", "MDC_CHANNEL_PROTOCOL=SIP") in new stack
-- Executing Set("SIP/213-081fe908", "MDC_CHANNEL_ID=213-081fe908") in new stack
-- Executing Set("SIP/213-081fe908", "MDC_CHANNEL_TMP=213-081fe908") in new stack
-- Executing Set("SIP/213-081fe908", "MDC_CHANNEL_NAME=213") in new stack
-- Executing GosubIf("SIP/213-081fe908", "1?mdc_initcall-int|05572000000|1") in new stack
-- Executing NoOp("SIP/213-081fe908", "initial call") in new stack
-- Executing Set("SIP/213-081fe908", "__MDC_DIALDESCENT=int") in new stack
-- Executing Set("SIP/213-081fe908", "__MDC_DIALCALLEENUM=05572000000") in new stack
-- Executing Set("SIP/213-081fe908", "__MDC_DIALCALLERNUM=213") in new stack
-- Executing GotoIf("SIP/213-081fe908", "1?nozap") in new stack
-- Goto (mdc_initcall-int,05572000000,8)
-- Executing Set("SIP/213-081fe908", "__MDC_DIALCALLERNAME=Alexander Klien") in new stack
-- Executing Return("SIP/213-081fe908", "") in new stack
-- Executing Gosub("SIP/213-081fe908", "mdc_defcall|05572000000|1") in new stack
-- Executing GotoIf("SIP/213-081fe908", "1?nozap") in new stack
-- Goto (mdc_defcall,05572000000,4)
-- Executing Set("SIP/213-081fe908", "__MDC_DIALCHANNELNAME=213") in new stack
-- Executing Return("SIP/213-081fe908", "") in new stack
-- Executing SIPAddHeader("SIP/213-081fe908", ""Alert-Info:&lt;http://www.notused.de&gt;;info=alert-internal;x-line-id=0"") in new stack
-- Executing Macro("SIP/213-081fe908", "pre-main") in new stack

Dec 31 14:09:42 WARNING[21720]: app_macro.c:208 macro_exec: No such context ‘macro-pre-main’ for macro ‘pre-main’
– Executing GosubIf(“SIP/213-081fe908”, “1?mdc_initloop|s|1”) in new stack
– Executing NoOp(“SIP/213-081fe908”, “initial loop”) in new stack
– Executing Set(“SIP/213-081fe908”, “MDC_ALIAS_HOP=0”) in new stack
– Executing Return(“SIP/213-081fe908”, “”) in new stack
– Executing Goto(“SIP/213-081fe908”, “main|05572000000|1”) in new stack
– Goto (main,05572000000,1)
– Executing Set(“SIP/213-081fe908”, “TMP_INPREFIX=”) in new stack
– Executing Goto(“SIP/213-081fe908”, “mdc_outgoing-trunk-4|05572000000|1”) in new stack
– Goto (mdc_outgoing-trunk-4,05572000000,1)
– Executing Set(“SIP/213-081fe908”, “TMP_USERFIELD=”) in new stack
– Executing Monitor(“SIP/213-081fe908”, “gsm|monitor-1293800982-213-05572000000-20101231-140942-out|m”) in new stack
– Executing Set(“SIP/213-081fe908”, “CDR(userfield)=<dst>05572000000</dst><name>213</name><inprefix></inprefix>”) in new stack
– Executing Set(“SIP/213-081fe908”, “TOUCH_MONITOR=213-05572000000-20101231-140942-out”) in new stack
– Executing Macro(“SIP/213-081fe908”, “int2nat|05572000000|0043|43|+43|5576”) in new stack
– Executing NoOp(“SIP/213-081fe908”, “macro-int2nat:: EXTEN: 05572000000 - intVAZ: 0043 - lkz: 43 - natVAZ: +43 - onKz: 5576”) in new stack
– Executing GotoIf(“SIP/213-081fe908”, “0?s-internat|1”) in new stack
– Executing Set(“SIP/213-081fe908”, “MDC_EXTEN=05572000000”) in new stack
– Executing Set(“SIP/213-081fe908”, “CALLERID(num)=5576000000”) in new stack
– Executing Dial(“SIP/213-081fe908”, “SIP/administrator/05572000000||Wt”) in new stack
We’re at 192.168.60.253 port 17078
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.60.252:5061:
INVITE sip:05572000000@192.168.60.252:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.60.253:5060;branch=z9hG4bK458e5fbf;rport
From: “Alexander Klien” <sip:5576000000@administrator>;tag=as68dde4b0
To: <sip:05572000000@192.168.60.252:5061>
Contact: <sip:5576000000@192.168.60.253>
Call-ID: 608fd3a90f120aa13900c86631852de5@administrator
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 31 Dec 2010 13:09:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Alert-Info: <http://www.notused.de>;info=alert-internal;x-line-id=0
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 4170 4170 IN IP4 192.168.60.253
s=session
c=IN IP4 192.168.60.253
t=0 0
m=audio 17078 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-- Called administrator/05572000000

Retransmitting #1 (no NAT) to 192.168.60.252:5061:
INVITE sip:05572000000@192.168.60.252:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.60.253:5060;branch=z9hG4bK458e5fbf;rport
From: “Alexander Klien” <sip:5576000000@administrator>;tag=as68dde4b0
To: <sip:05572000000@192.168.60.252:5061>
Contact: <sip:5576000000@192.168.60.253>
Call-ID: 608fd3a90f120aa13900c86631852de5@administrator
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 31 Dec 2010 13:09:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Alert-Info: <http://www.notused.de>;info=alert-internal;x-line-id=0
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 4170 4170 IN IP4 192.168.60.253
s=session
c=IN IP4 192.168.60.253
t=0 0
m=audio 17078 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

voip*CLI>
<-- SIP read from 192.168.60.252:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.60.253:5060;branch=z9hG4bK458e5fbf;rport=5060;received=192.168.60.253
From: “Alexander Klien” <sip:5576000000@administrator>;tag=as68dde4b0
To: <sip:05572000000@192.168.60.252:5061>
Call-ID: 608fd3a90f120aa13900c86631852de5@administrator
CSeq: 102 INVITE
Server: Patton SN4638 5BIS 00A0BA05D3CC R5.5 2010-07-09 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0

— (8 headers 0 lines) —
voip*CLI>
<-- SIP read from 192.168.60.252:5061:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.60.253:5060;branch=z9hG4bK458e5fbf;rport=5060;received=192.168.60.253
From: “Alexander Klien” <sip:5576000000@administrator>;tag=as68dde4b0
To: <sip:05572000000@192.168.60.252:5061>;tag=2811984651
Call-ID: 608fd3a90f120aa13900c86631852de5@administrator
CSeq: 102 INVITE
Server: Patton SN4638 5BIS 00A0BA05D3CC R5.5 2010-07-09 H323 SIP BRI M5T SIP Stack/4.0.28.28
Content-Length: 0

— (8 headers 0 lines) —
– Got SIP response 503 “Service Unavailable” back from 192.168.60.252
Transmitting (no NAT) to 192.168.60.252:5061:
ACK sip:05572000000@192.168.60.252:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.60.253:5060;branch=z9hG4bK458e5fbf;rport
From: “Alexander Klien” <sip:5576000000@administrator>;tag=as68dde4b0
To: <sip:05572000000@192.168.60.252:5061>;tag=2811984651
Contact: <sip:5576000000@192.168.60.253>
Call-ID: 608fd3a90f120aa13900c86631852de5@administrator
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

-- SIP/administrator-082c74f8 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)
– Executing StopMonitor(“SIP/213-081fe908”, “”) in new stack
Dec 31 14:09:43 NOTICE[21720]: res_monitor.c:316 ast_monitor_stop: monitor executing ( nice -n 19 soxmix “/var/spool/asterisk/monitor/monitor-1293800982-213-05572000000-20101231-140942-out-in.gsm” “/var/spool/asterisk/monitor/monitor-1293800982-213-05572000000-20101231-140942-out-out.gsm” “/var/spool/asterisk/monitor/monitor-1293800982-213-05572000000-20101231-140942-out.gsm” && rm -f “/var/spool/asterisk/monitor/monitor-1293800982-213-05572000000-20101231-140942-out-”* ) &
– Executing Hangup(“SIP/213-081fe908”, “34”) in new stack
== Spawn extension (mdc_outgoing-trunk-4, 05572000000, 9) exited non-zero on ‘SIP/213-081fe908’
Scheduling destruction of call ‘3c278ffc7468-hewyujzauo5g’ in 32000 ms
Reliably Transmitting (no NAT) to 192.168.60.17:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.60.17:5060;branch=z9hG4bK-qirlra95qc81;received=192.168.60.17;rport=5060
From: “Alexander Klien” <sip:213@192.168.60.253>;tag=k32egfzbbt
To: <sip:05572000000@192.168.60.253>;tag=as0b728402
Call-ID: 3c278ffc7468-hewyujzauo5g
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:05572000000@192.168.60.253>
Content-Length: 0

Destroying call ‘608fd3a90f120aa13900c86631852de5@administrator’
voip*CLI>
<-- SIP read from 192.168.60.17:5060:
ACK sip:05572000000@192.168.60.253 SIP/2.0
Via: SIP/2.0/UDP 192.168.60.17:5060;branch=z9hG4bK-qirlra95qc81;rport
From: “Alexander Klien” <sip:213@192.168.60.253>;tag=k32egfzbbt
To: <sip:05572000000@192.168.60.253>;tag=as0b728402
Call-ID: 3c278ffc7468-hewyujzauo5g
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:213@192.168.60.17:5060>;reg-id=1
Content-Length: 0

maik · January 3, 2011, 3:16am

Hallo zelos,

hast Du das Patton GW manuell eingerichtet oder mit dem Assistenten im Commander ?

Grüße

Maik

Mathias · January 3, 2011, 12:48pm

Hallo,

schaut so aus als ob das Patton eine Huntgroup mit nicht aktiven Kanälen hat. Wie Maik schon sagt ist es wichtig, dass Du das Amt mit der Schnellanlage als Pattonamt angelegt hast und das Du das Patton dann automatisch konfiguriert hast. Außerdem ist es wichtig, dass Du keinen ISDN Port konfigurierst und dann kein ISDN ansteckst.

LG
Mathias