Problem zweiter Anruf beim SmartNode 4120

pa7r1ck · 28 Sep 2012

Hallo zusammen,
vorab: Ich bin noch recht unerfahren bei dem Umgang mit MediaGateways und dem Thema ISDN. Daher hoffe ich hier auf eure Hilfe.

Aktuell betreibe ich in einem sehr kleinem Testszenario einen Asterisk Server und ein Patton SmartNode 4120 Media Gateway mit zwei BRI-Anschlüssen.
Ein Anruf von ISDN zu SIP und umgekehrt funktionieren problemlos.

Problem: Es funktioniert immer nur ein Anruf gleichzeitig.

Das heißt, wenn ich einen Anruf vom ISDN zum SIP-Telefon aufgebaut habe und beispielsweise von einem Softclient auf meinem Rechner einen weiteren Anruf zu einem ISDN Telefon aufbauen will, kommt ein "SIP/2.0 503 Service Unavailable" (Das heißt schonmal der Anruf kommt durch)
Wenn ich den ersten Anruf beende, kommt der zweite ohne Probleme durch.

Die zwei BRI-Anschlüsse sind in ordnung, die habe ich überprüft.
Wie Eingangs erwähnt, bin ich noch nicht so tief in der Materie und daher hoffe ich auf Hilfe oder einen Hinweis von euch.

Ich vermute, dass ich noch etwas an der Routing-Tabelle auf dem Smartnode ändern muss.

Die Konfiguration sieht aktuell folgendermaßen aus:

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4120/2BIS4V                                                  #
# R6.2 2012-07-13 H323 SIP                                       #
# 2012-09-18T13:37:12                                            #
# SN/00A0BA085D02                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
administrator administrator password LNOVrt1b6LZ8vjRhaMx1nQ== encrypted
clock local default-offset +00:00
dns-client server 172.22.1.10
webserver port 80 language en
sntp-client
sntp-client server primary 0.de.pool.ntp.org port 123 version 4

system

  ic voice 0
    low-bitrate-codec g729

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1

profile ppp default

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default
  no autonomous-transitioning

profile aaa default
  method 1 local
  method 2 none

context ip router

  interface IF_IP_LAN
    ipaddress 172.22.149.150 255.255.240.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context cs switch
  national-prefix 0
  international-prefix 00

  routing-table called-e164 RT_SIP_TO_ISDN
    route default dest-service SV_HUNT_PSTN

  routing-table calling-e164 RT_ISDN_TO_SIP
    route T dest-interface IF_SIP

  interface isdn IF_ISDN_0
    route call dest-table RT_ISDN_TO_SIP

  interface isdn IF_ISDN_1
    route call dest-table RT_ISDN_TO_SIP

  interface sip IF_SIP
    bind context sip-gateway GW_SIP
    route call dest-table RT_SIP_TO_ISDN
    remote 172.22.149.149 5060

  service hunt-group SV_HUNT_PSTN
    route call 1 dest-interface IF_ISDN_0
    route call 2 dest-interface IF_ISDN_1

context cs switch
  no shutdown

context sip-gateway GW_SIP

  interface IF_LAN
    bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP
  no shutdown

port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface IF_IP_LAN router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_ISDN_0 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_ISDN_1 switch

port bri 0 1
  no shutdown

SIP Debug auf dem Asterisk (Rot makierte Stelle)

Code:

<--- SIP read from UDP:172.22.152.96:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.22.152.96:5060;branch=z9hG4bK-d8754z-c49766a11d3d2ea3-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "Laptop"<sip:[email protected]>;tag=7b81f95a
Call-ID: YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 244

v=0
o=- 12993305559028883 1 IN IP4 172.22.152.96
s=CounterPath X-Lite 5.0.0
c=IN IP4 172.22.152.96
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 97 8 3 101
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
Sending to 172.22.152.96:5060 (NAT)
Using INVITE request as basis request - YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
Found peer '2000' for '2000' from 172.22.152.96:5060

<--- Reliably Transmitting (NAT) to 172.22.152.96:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.22.152.96:5060;branch=z9hG4bK-d8754z-c49766a11d3d2ea3-1---d8754z-;received=172.22.152.96;rport=5060
From: "Laptop"<sip:[email protected]>;tag=7b81f95a
To: <sip:[email protected]>;tag=as2f95f56f
Call-ID: YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.8.2~dfsg-1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="64fd0a49"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:172.22.152.96:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.22.152.96:5060;branch=z9hG4bK-d8754z-c49766a11d3d2ea3-1---d8754z-;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as2f95f56f
From: "Laptop"<sip:[email protected]>;tag=7b81f95a
Call-ID: YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:172.22.152.96:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.22.152.96:5060;branch=z9hG4bK-d8754z-fae80b270f76097c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "Laptop"<sip:[email protected]>;tag=7b81f95a
Call-ID: YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Authorization: Digest username="2000",realm="asterisk",nonce="64fd0a49",uri="sip:[email protected]",response="23e171288376674ae4fbb0e14d90bb28",algorithm=MD5
Content-Length: 244

v=0
o=- 12993305559028883 1 IN IP4 172.22.152.96
s=CounterPath X-Lite 5.0.0
c=IN IP4 172.22.152.96
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 97 8 3 101
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 172.22.152.96:5060 (NAT)
Using INVITE request as basis request - YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
Found peer '2000' for '2000' from 172.22.152.96:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format SPEEX for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.152.96:5062
Looking for 8901 in default (domain 172.22.149.149)
list_route: hop: <sip:[email protected]:5060>

<--- Transmitting (NAT) to 172.22.152.96:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.152.96:5060;branch=z9hG4bK-d8754z-fae80b270f76097c-1---d8754z-;received=172.22.152.96;rport=5060
From: "Laptop"<sip:[email protected]>;tag=7b81f95a
To: <sip:[email protected]>
Call-ID: YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.2~dfsg-1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0

<------------>
    -- Executing [8901@default:1] Dial("SIP/2000-000000a6", "SIP/8901@gw_patton,60,tT") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.22.149.150:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.149.149:5060;branch=z9hG4bK4ad8942d;rport
Max-Forwards: 70
From: "Phone1" <sip:[email protected]>;tag=as2f406f24
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.8.2~dfsg-1
Date: Fri, 28 Sep 2012 11:32:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 70017428 70017428 IN IP4 172.22.149.149
s=Asterisk PBX 1.8.8.2~dfsg-1
c=IN IP4 172.22.149.149
t=0 0
m=audio 13342 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/8901@gw_patton

<--- SIP read from UDP:172.22.149.150:5060 --->
[B][COLOR="#FF0000"]SIP/2.0 503 Service Unavailable[/COLOR][/B]
Via: SIP/2.0/UDP 172.22.149.149:5060;branch=z9hG4bK4ad8942d;rport=5060;received=172.22.149.149
From: "Phone1" <sip:[email protected]>;tag=as2f406f24
To: <sip:[email protected]:5060>;tag=1594093892
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Patton SN4120 2BIS4V 00A0BA085D02 R6.2 2012-07-13 H323 SIP M5T SIP Stack/4.0.30.30
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 503 "Service Unavailable" back from 172.22.149.150:5060
Transmitting (NAT) to 172.22.149.150:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.149.149:5060;branch=z9hG4bK4ad8942d;rport
Max-Forwards: 70
From: "Phone1" <sip:[email protected]>;tag=as2f406f24
To: <sip:[email protected]:5060>;tag=1594093892
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.2~dfsg-1
Content-Length: 0

---
    -- SIP/gw_patton-000000a7 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [8901@default:2] Hangup("SIP/2000-000000a6", "") in new stack
  == Spawn extension (default, 8901, 2) exited non-zero on 'SIP/2000-000000a6'
Scheduling destruction of SIP dialog 'YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 172.22.152.96:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.152.96:5060;branch=z9hG4bK-d8754z-fae80b270f76097c-1---d8754z-;received=172.22.152.96;rport=5060
From: "Laptop"<sip:[email protected]>;tag=7b81f95a
To: <sip:[email protected]>;tag=as44fae08b
Call-ID: YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.2~dfsg-1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<--- SIP read from UDP:172.22.152.96:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.22.152.96:5060;branch=z9hG4bK-d8754z-fae80b270f76097c-1---d8754z-;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as44fae08b
From: "Laptop"<sip:[email protected]>;tag=7b81f95a
Call-ID: YTZiMmQwNmJkM2UwMDUxOWZlYzJiZWI2YzgwZTVkZTM.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: INVITE

pa7r1ck · 1 Okt 2012

So, noch ein kleines Update. Also das parallele Anrufen vom ISDN zu SIP funktioniert soweit ohne Probleme.
Scheint also wahrscheinlich was mit der Routing-Table zu tun zu haben.

Ich fische weiter im trüben, Petri Heil

silvan · 9 Okt 2012

Ich kann dir leider nicht absolut weiterhelfen. Jedoch sieht mir deine Hunt-Group etwas komisch aus. In Beispielen welche ich gesehen habe, siehts jeweils eher so aus:

Code:

  service hunt-group SER_HUNT_PSTN
    cyclic
    timeout 6
    drop-cause normal-unspecified
    drop-cause no-circuit-channel-available
    drop-cause network-out-of-order
    drop-cause temporary-failure
    drop-cause switching-equipment-congestion
    drop-cause access-info-discarded
    drop-cause circuit-channel-not-available
    drop-cause resources-unavailable
    route call 1 dest-interface IF_TE_00
    route call 2 dest-interface IF_TE_01

Ich selber konnte ein 4120 mit folgender Konfiguration grundsätzlich zum laufen bringen, verwende jedoch nur einen BRI Port und keine Routing Tables / Hunt Group. Vielleicht hilft es dir trotzdem weiter:

Code:

#----------------------------------------------------------------#
#                                                                #
# SN4120/2BIS4V                                                  #
# R6.2 2012-07-13 H323 SIP                                       #
# 2012-10-07T23:14:10                                            #
# SN/00A0BA06ED3F                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
clock local default-offset +00:00
timer PROVISIONING now + 1 minute "provisioning execute PF_PROVISIONING_CONFIG"
dns-client server 10.0.0.1
webserver port 80 language en
sntp-client
sntp-client server primary pool.ntp.org port 123 version 4
system hostname sn4120

system

  ic voice 0
    low-bitrate-codec g729

system
  clock-source 1 bri 0 0
  clock-source 2 bri 0 1

profile ppp default

profile tone-set default

profile voip default
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default
  no autonomous-transitioning

profile aaa default
  method 1 local
  method 2 none

profile provisioning PF_PROVISIONING_CONFIG
  destination configuration
  location 1 http://redirect.patton.com/$(system.mac);mac=$(system.mac);serial=$(system.serial);hwMajor=$(system.hw.major);hwMinor=$(system.hw.minor);swMajor=$(system.sw.major);swMinor=$(system.sw.minor);swDate=$(system.sw.date);productName=$(system.product.name);cliMajor=$(cli.major);cliMinor=$(cli.minor);osName=$(cli.major>=4|Trinity|SmartWare);subDirTrinity=$(cli.major>=4|/Trinity);subDirSmartWare=$(cli.major<4|/SmartWare);dhcp66=$(dhcp.66);dhcp67=$(dhcp.67)
  location 2 $(dhcp.66)
  location 3 $(dhcp.66)/$(system.mac).cfg
  location 4 http://$(dhcp.66)/$(dhcp.67)
  location 5 http://$(dhcp.66)/$(system.mac).cfg
  location 6 tftp://$(dhcp.66)/$(dhcp.67)
  location 7 tftp://$(dhcp.66)/$(system.mac).cfg
  activation reload immediate

context ip router

  interface eth0
    ipaddress 10.0.0.3 255.255.255.0
    tcp adjust-mss rx mtu
    tcp adjust-mss tx mtu

context ip router
  route 0.0.0.0 0.0.0.0 10.0.0.1 0

context cs switch

  interface isdn IF_TE_00
    route call dest-interface IF_SIP_ASTERISK

  interface sip IF_SIP_ASTERISK
    bind context sip-gateway GW_ASTERISK
    route call dest-interface IF_TE_00
    remote 10.0.0.2

context cs switch
  no shutdown

context sip-gateway GW_ASTERISK

  interface eth0
    bind interface eth0 context router port 5060

context sip-gateway GW_ASTERISK
  no shutdown

port ethernet 0 0
  medium auto
  encapsulation ip
  bind interface eth0 router
  no shutdown

port bri 0 0
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending
      encapsulation cc-isdn
      bind interface IF_TE_00 switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side user
      bchan-number-order ascending

port bri 0 1
  shutdown

pa7r1ck · 9 Okt 2012

Hey silvan,
danke für deine Antwort, ich hab die "drop-causes" rausgelassen, um es übersichtlicher zu halten.
Ich probiere die Lösung der HUNT-GROUP dann nochmal. Vielleicht helfen die cyclic und timeout 6 bereits, ich les mal nach was das macht.

Viele Grüße

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