Mailing List Archive: AS5300 voice call answering problem..

AS5300 voice call answering problem..

May 25, 2006, 4:10 PM

Post #1 of 3 (8006 views)

Hi All..

Hopefully this is just a quick one.. Im trying to get our AS5300 to forward
incoming PSTN calls to our asterisk server (192.168.1.2)

AS5300, IOS (tm) 5300 Software (C5300-IK9S-M), Version 12.3(15), RELEASE
SOFTWARE (fc3)
4 Channelized E1/PRI port(s)
30 DSP(s), 60 Voice resource(s)

What is happening is.. a voice call comes into the chassis, the voice card
answers it, but...

May 26 08:51:22.373 AEST: %CALLTRKR-6-CALL_RECORD: ct_hndl=6624,
service=None, origin=Answer, category=IsdnSync, DS0
slot/port/ds1/chan=0/0/0/6, called=85550050, calling=(n/a), resource
slot/port=(n/a)/(n/a), userid=(n/a), ip=0.0.0.0, account id=12869,
setup=05/26/2006 08:51:10, conn=0.00, phys=0.00, service=0.00, authen=0.00,
init-rx/tx b-rate=0/0, rx/tx chars=0/0, time=0.06, disc subsys=ISDN, disc
code=0x1, disc text=Unallocated/unassigned number

Yet in the 5300 ive got :

voice-port 0:D
input gain -6
output attenuation 14
echo-cancel coverage 32
echo-cancel suppressor
cptone AU
description E1
bearer-cap Speech
!
dial-peer voice 30050 voip
application session
destination-pattern 85550050
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!
dial-peer voice 20050 pots
incoming called-number 85550050
direct-inward-dial
forward-digits extra
!
sip-ua
retry invite 4
retry response 3
retry bye 2
retry cancel 2
timers trying 1000
sip-server ipv4:192.168.1.2:5060
!

asterisk is set for :

[AS5300]
host=192.168.1.1
context=as5300-pstn-inbound
type=peer
dtmf=rfc2833
nat=no
canreinvite=yes
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ulaw

What am I missing ??

Steve

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Re: AS5300 voice call answering problem.. [ In reply to ]

dsladden at cisco

May 25, 2006, 5:21 PM

Post #2 of 3 (7810 views)

Permalink

Do you have "isdn incoming-voice modem"
under your serial interface ?

you need a "debug isdn q931" and "debug ccsip"
to be sure.

Regards,
Darryl Sladden
Product Manager AS5000
Cisco Systems - ABU
dsladden@cisco.com
408-525-8970

> -----Original Message-----
> From: cisco-nas-bounces@puck.nether.net
> [mailto:cisco-nas-bounces@puck.nether.net] On Behalf Of Steve
> Monkhouse
> Sent: Thursday, May 25, 2006 4:11 PM
> To: cisco-nas@puck.nether.net
> Subject: [cisco-nas] AS5300 voice call answering problem..
>
>
> Hi All..
>
> Hopefully this is just a quick one.. Im trying to get our
> AS5300 to forward incoming PSTN calls to our asterisk server
> (192.168.1.2)
>
> AS5300, IOS (tm) 5300 Software (C5300-IK9S-M), Version
> 12.3(15), RELEASE SOFTWARE (fc3)
> 4 Channelized E1/PRI port(s)
> 30 DSP(s), 60 Voice resource(s)
>
> What is happening is.. a voice call comes into the chassis,
> the voice card answers it, but...
>
> May 26 08:51:22.373 AEST: %CALLTRKR-6-CALL_RECORD:
> ct_hndl=6624, service=None, origin=Answer, category=IsdnSync,
> DS0 slot/port/ds1/chan=0/0/0/6, called=85550050,
> calling=(n/a), resource slot/port=(n/a)/(n/a), userid=(n/a),
> ip=0.0.0.0, account id=12869,
> setup=05/26/2006 08:51:10, conn=0.00, phys=0.00,
> service=0.00, authen=0.00, init-rx/tx b-rate=0/0, rx/tx
> chars=0/0, time=0.06, disc subsys=ISDN, disc code=0x1, disc
> text=Unallocated/unassigned number
>
> Yet in the 5300 ive got :
>
> voice-port 0:D
> input gain -6
> output attenuation 14
> echo-cancel coverage 32
> echo-cancel suppressor
> cptone AU
> description E1
> bearer-cap Speech
> !
> dial-peer voice 30050 voip
> application session
> destination-pattern 85550050
> session protocol sipv2
> session target sip-server
> dtmf-relay rtp-nte
> no vad
> !
> dial-peer voice 20050 pots
> incoming called-number 85550050
> direct-inward-dial
> forward-digits extra
> !
> sip-ua
> retry invite 4
> retry response 3
> retry bye 2
> retry cancel 2
> timers trying 1000
> sip-server ipv4:192.168.1.2:5060
> !
>
> asterisk is set for :
>
> [AS5300]
> host=192.168.1.1
> context=as5300-pstn-inbound
> type=peer
> dtmf=rfc2833
> nat=no
> canreinvite=yes
> dtmfmode=rfc2833
> disallow=all
> allow=g729
> allow=ulaw
>
>
> What am I missing ??
>
> Steve
>
>
> _______________________________________________
> cisco-nas mailing list
> cisco-nas@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-nas
>

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Re: AS5300 voice call answering problem.. [ In reply to ]

dsladden at cisco

May 25, 2006, 6:10 PM

Post #3 of 3 (7784 views)

Permalink

Steve,

It appears that the AS5300 is sending the invite out and
ASTERIK is rejecting it.

You can use

debug ccsip messages

(I think that is right)

To actually see the outgoing SIP INVITE.

Regards,
Darryl Sladden
Product Manager AS5000
Cisco Systems - ABU
dsladden@cisco.com
408-525-8970

> -----Original Message-----
> From: Steve Monkhouse [mailto:steve.monkhouse@ethertech.com.au]
> Sent: Thursday, May 25, 2006 5:55 PM
> To: Darryl Sladden (dsladden); cisco-nas@puck.nether.net
> Subject: RE: [cisco-nas] AS5300 voice call answering problem..
>
> Thanks for your quick response Darryl..
>
> Yes we do have 'isdn incoming-voice modem' as below..
>
> interface Serial0:15
> no ip address
> encapsulation ppp
> dialer idle-timeout 4000
> dialer load-threshold 5 either
> dialer-group 1
> isdn switch-type primary-net5
> isdn incoming-voice modem
> no keepalive
> no fair-queue
> no cdp enable
> ppp authentication chap ms-chap-v2 pap
>
> after enabling the extra debugs..
>
> melgw1#
> May 26 10:49:16.914 AEST: ISDN Se0:15 Q931: RX <- SETUP pd =
> 8 callref =
> 0x0015
> Sending Complete
> Bearer Capability i = 0x8090A3
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA9838D
> Exclusive, Channel 13
> Calling Party Number i = 0x00A3, N/A
> Plan:Unknown, Type:Unknown
> Called Party Number i = 0xC1, '85550050'
> Plan:ISDN, Type:Subscriber(local) May 26
> 10:49:16.918 AEST: AAA/BIND(00001D2C): Bind i/f Serial0:12
> May 26 10:49:16.918 AEST: AAA/ACCT/DS0: channel=12, ds1=0,
> t3=0, slot=0,
> ds0=12
> May 26 10:49:16.918 AEST: AAA/ACCT/DS0: channel=12, ds1=0,
> t3=0, slot=0,
> ds0=12
> May 26 10:49:16.934 AEST: adding call id 66 to table
>
> May 26 10:49:16.934 AEST: Queued event from SIP SPI :
> SIPSPI_EV_CC_CALL_SETUP
> May 26 10:49:16.934 AEST: CCSIP-SPI-CONTROL:
> act_idle_call_setup May 26 10:49:16.938 AEST:
> act_idle_call_setup:Not using Voice Class Codec
>
> May 26 10:49:16.938 AEST: act_idle_call_setup:
> preferred_codec set[0] type
> :g729r8 bytes: 20
> May 26 10:49:16.938 AEST: sipSPICopyPeerDataToCCB: From CLI:
> Modem NSE payload = 100, Passthrough = 0,Modem relay = 0,
> Gw-Xid = 1 SPRT latency 200, SPRT Retries = 12, Dict Size =
> 1024 String Len = 32, Compress dir = 3 May 26 10:49:16.938
> AEST: sipSPIValidateGtd: Signal Forward disabled May 26
> 10:49:16.938 AEST: sipSPICanSetFallbackFlag - Local Fallback
> is not active May 26 10:49:16.938 AEST: Queued event from SIP SPI :
> SIPSPI_EV_CREATE_CONNECTION
> May 26 10:49:16.938 AEST: ****Adding to UAC table. ccb=0x630EC2E0
> key=4B2101BA-EB8811DA-80CADA1C-DBBEEDD0@192.168.1.1
> May 26 10:49:16.938 AEST: sipSPIUsetBillingProfile: sipCallId
> for billing records = 4B2101BA-EB8811DA-80CADA1C-DBBEEDD0@192.168.1.1
> May 26 10:49:16.946 AEST: CCSIP-SPI-CONTROL:
> act_idle_connection_created May 26 10:49:16.946 AEST:
> CCSIP-SPI-CONTROL: act_idle_connection_created:
> Connid(1) created to 192.168.1.2:5060, local_port 52760 May
> 26 10:49:16.946 AEST: CCSIP-SPI-CONTROL:
> sipSPIOutgoingCallSDP May 26 10:49:16.946 AEST: Preferred
> method of dtmf relay is: 6, with payload : 101
>
> May 26 10:49:16.946 AEST: convert_codec_bytes_to_ptime:
> Values :Codec:
> g729r8 codecbytes :20, ptime: 20
>
> May 26 10:49:16.946 AEST: sip_generate_sdp_xcaps_list: Modem
> Relay and T38 disabled. X-cap not needed May 26 10:49:16.946
> AEST: CCSIP-SPI-CONTROL: Converting TimeZone AEST to SIP
> default timezone = GMT May 26 10:49:16.946 AEST: Received
> Octet3A=0xA3 -> Setting ;screen=yes ;privacy=full May 26
> 10:49:16.946 AEST: sipSPIAddLocalContact May 26 10:49:16.950
> AEST: Queued event from SIP SPI :
> SIPSPI_EV_SEND_MESSAGE
> May 26 10:49:16.950 AEST: sip_stats_method May 26
> 10:49:16.950 AEST: act_idle_connection_created: Transaction active.
> Facilities will be queued.
> May 26 10:49:16.950 AEST: HandleUdpSocketWrites - Using new
> buffer for sip message May 26 10:49:16.954 AEST: ISDN Se0:15
> Q931: TX -> CALL_PROC pd = 8 callref = 0x8015
> Channel ID i = 0xA9838D
> Exclusive, Channel 13
> May 26 10:49:17.130 AEST: HandleUdpSocketReads :Msg enqueued
> for SPI with
> IPaddr: 203.56.92.50:5060
> May 26 10:49:17.130 AEST: CCSIP-SPI-CONTROL:
> act_sentinvite_new_message May 26 10:49:17.130 AEST:
> CCSIP-SPI-CONTROL: sipSPICheckResponse May 26 10:49:17.130
> AEST: sip_stats_status_code May 26 10:49:17.130 AEST:
> Roundtrip delay 184 milliseconds for method INVITE
>
> May 26 10:49:17.130 AEST: Queued event from SIP SPI :
> SIPSPI_EV_SEND_MESSAGE
> May 26 10:49:17.130 AEST: sip_stats_method May 26
> 10:49:17.130 AEST: ccsip_set_release_source_for_peer:ownCallId[102],
> src[4]
>
> May 26 10:49:17.130 AEST: CCSIP-SPI-CONTROL:
> sipSPIInitiateCallDisconnect :
> Initiate call disconnect(1) for outgoing call May 26
> 10:49:17.134 AEST: Queued event from SIP SPI :
> SIPSPI_EV_CC_CALL_DISCONNECT
> May 26 10:49:17.134 AEST: CCSIP-SPI-CONTROL:
> act_disconnecting_disconnect May 26 10:49:17.134 AEST:
> CCSIP-SPI-CONTROL: sipSPICallCleanup May 26 10:49:17.134
> AEST: sipSPIIcpifUpdate :CallState: 2 Playout: 0
> DiscTime:-1918533452 ConnTime 0
>
> May 26 10:49:17.134 AEST: The Call Setup Information is :
> Call Control Block (CCB) : 0x630EC2E0
> State of The Call : STATE_DEAD
> TCP Sockets Used : NO
> Calling Number :
> Called Number : 85550050
> Number of Media Streams : 1
>
> May 26 10:49:17.134 AEST: Media Stream 1
> Negotiated Codec : No Codec
> Negotiated Codec Bytes : 0
> Negotiated Dtmf-relay : 0
> Dtmf-relay Payload : 0
> Source IP Address (Media): 192.168.1.1
> Source IP Port (Media): 18440
> Destn IP Address (Media): 0.0.0.0
> Destn IP Port (Media): 0
>
> May 26 10:49:17.138 AEST: Orig Destn IP Address:Port (Media):
> 0.0.0.0:0
>
> May 26 10:49:17.138 AEST:
> Source IP Address (Sig ): 192.168.1.1
> Destn SIP Req Addr:Port : 192.168.1.2:5060 Destn SIP Resp
> Addr:Port : 0.0.0.0:0
> Destination Name : 192.168.1.2
>
> May 26 10:49:17.138 AEST:
> Disconnect Cause (CC) : 1
> Disconnect Cause (SIP) : 404
>
> melgw1#
> May 26 10:49:17.150 AEST: ISDN Se0:15 Q931: TX -> DISCONNECT
> pd = 8 callref = 0x8015
> Cause i = 0x8081 - Unallocated/unassigned number May
> 26 10:49:17.326 AEST: ISDN Se0:15 Q931: RX <- RELEASE pd = 8
> callref =
> 0x0015
> May 26 10:49:17.330 AEST: AAA/ACCT/DS0: channel=12, ds1=0,
> t3=0, slot=0,
> ds0=12
> May 26 10:49:17.330 AEST: ISDN Se0:15 Q931: TX ->
> RELEASE_COMP pd = 8 callref = 0x8015 melgw1# May 26
> 10:49:22.370 AEST: %CALLTRKR-6-CALL_RECORD: ct_hndl=6629,
> service=None, origin=Answer, category=IsdnSync, DS0
> slot/port/ds1/chan=0/0/0/12, called=85550050, calling=(n/a),
> resource slot/port=(n/a)/(n/a), userid=(n/a), ip=0.0.0.0,
> account id=12879,
> setup=05/26/2006 10:49:13, conn=0.00, phys=0.00,
> service=0.00, authen=0.00, init-rx/tx b-rate=0/0, rx/tx
> chars=0/0, time=0.24, disc subsys=ISDN, disc code=0x1, disc
> text=Unallocated/unassigned number melgw1#
>
> thoughts ?
>
> regards,
> Steve
>

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