[asterisk-users] Re: Asterisk With Cisco Voice Router
Timothy Smith
timotsmith at gmail.com
Mon May 18 05:42:02 CDT 2009
Thank David and Neeraj for your input.
Neeraj, I posted the configs in my first post, but i've also attached
some extracts here. they haven't changed much.
David, You're absolutely right and i think the problem could be the
reverse dial-peer or DTMF configuration. I think I have the
corresponding reverse dial-peer and the DTMF conf that you said.
However, I have checked my side and all seems to be ok. I've also
tried changing the dtmfmode to sip-notify on the gateway (and info in
sip.conf) but no luck!
Please look at the attached and give me some pointers.
Thanks,
Tim
>
> Did you ever set up that reverse dial-peer? If not, do that first.
>
> You put a three second timeout on the Read(). By any chance, is the
> call hanging up 3 seconds after you call? That would be expected
> behavior. Well, actually you give it two tries. So it should be
> beep
> three second wait
> beep
> three second wait
> hangup
>
> If you're actually entering numbers on your dialpad and they're not
> getting read, you have a misconfiguration on your DTMF. If you enable
> sip debugging on your asterisk side you can see exactly what's coming
> over the wire from the Cisco side. There are a lot of choices for DTMF
> on the asterisk side and the Cisco side, and they need to agree for
> the button presses to be encoded and passed correctly. You can pass
> them in-line as real audio, or you can convert them to a special dtmf
> sip encoding. You'll notice all those choices when you go to configure
> the Cisco dial-peer.
>
> My personal preference:
> on the Cisco dial-peer side
> dtmf-relay rtp-nte
>
> on the asterisk side
> I left the dtmf config blank, and I don't remember which default you
> end up with, but it worked in the default config for me.
>
-------------- next part --------------
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no cdp enable
!
interface Serial0/0/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn negotiate-bchan resend-setup
no cdp enable
dial-peer voice 1 pots
destination-pattern 0T
port 0/0/1:15
forward-digits all
!
dial-peer voice 3 pots
incoming called-number .
direct-inward-dial
port 0/0/1:15
dial-peer voice 4 pots
incoming called-number .
direct-inward-dial
port 0/0/1:15
!
dial-peer voice 112 voip
destination-pattern 730732888
monitor probe icmp-ping
session protocol sipv2
session target ipv4:172.19.3.150
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
VG2# show dial-peer voice 112
VoiceOverIpPeer112
peer type = voice, system default peer = FALSE, information type = voice,
description = `',
tag = 112, destination-pattern = `730732888',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 112, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = enabled,
modem transport = system,
URI classes:
Incoming (Request) =
Incoming (To) =
Incoming (From) =
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
type = voip, session-target = `ipv4:172.19.3.150',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip signaling DSCP = af31,
ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
ip video rsvp-fail DSCP = af41,
UDP checksum = disabled,
session-protocol = sipv2, session-transport = udp,
req-qos = best-effort, acc-qos = best-effort,
req-qos video = best-effort, acc-qos video = best-effort,
req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,
dtmf-relay = rtp-nte,
RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
A-law=8, GSMAMR-NB=117 iLBC=116
h263+=118, h264=119
G726r16 using static payload
G726r24 using static payload
RTP comfort noise payload type = 19
fax rate = voice, payload size = 20 bytes
fax protocol = system
fax-relay ecm enable
Fax Relay SG3-to-G3 Enabled (by system configuration)
fax NSF = 0xAD0051 (default)
codec = g711ulaw, payload size = 160 bytes,
video codec = None
voice class codec = `'
text relay = disabled
Media Setting = flow-through (global)
Expect factor = 10, Icpif = 20,
Playout Mode is set to adaptive,
Initial 60 ms, Max 250 ms
Playout-delay Minimum mode is set to default, value 40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = cas,
VAD = enabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip rel1xx = system,
tvoice class sip outbound-proxy = system,
voice class sip asserted-id = system,
voice class sip privacy = system,
voice class sip e911 = system,
voice class sip authenticate redirecting-number = system,
redirect ip2ip = disabled
local peer = false
monitor probe method: icmp-ping monitoring session target,
Monitored destination reachable
Secure RTP: system (use the global setting)
voice class perm tag = `'
Time elapsed since last clearing of voice call statistics never
Connect Time = 32060, Charged Units = 0,
Successful Calls = 86, Failed Calls = 3, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "2F ",
Last Disconnect Text is "no resource (47)",
Last Setup Time = 378671190.
Last Disconnect Time = 378671508.
-------------- next part --------------
[May 18 09:54:58] DEBUG[28907] channel.c: Avoiding initial deadlock for channel '0x893750'
[May 18 09:54:58] DEBUG[28907] channel.c: Avoiding initial deadlock for channel '0x893750'
[May 18 09:54:58] DEBUG[10017] pbx.c: Launching 'Set'
[May 18 09:54:58] VERBOSE[10017] logger.c: -- Executing [730732888 at default:1] Set("SIP/172.17.3.248-008c4790", "CALLERID(num)=730730199") in new stack
[May 18 09:54:58] DEBUG[10017] pbx.c: Launching 'Wait'
[May 18 09:54:58] VERBOSE[10017] logger.c: -- Executing [730732888 at default:2] Wait("SIP/172.17.3.248-008c4790", "3") in new stack
[May 18 09:55:01] DEBUG[10017] pbx.c: Launching 'Read'
[May 18 09:55:01] VERBOSE[10017] logger.c: -- Executing [730732888 at default:3] Read("SIP/172.17.3.248-008c4790", "NUM,beep,4,2,3") in new stack
[May 18 09:55:01] VERBOSE[10017] logger.c: -- Accepting a maximum of 4 digits.
[May 18 09:55:01] DEBUG[10017] chan_sip.c: SIP answering channel: SIP/172.17.3.248-008c4790
[May 18 09:55:01] DEBUG[10017] chan_sip.c: Setting framing from config on incoming call
[May 18 09:55:01] DEBUG[10017] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
[May 18 09:55:01] DEBUG[10017] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[May 18 09:55:01] VERBOSE[10017] logger.c: Audio is at 172.19.3.150 port 15200
[May 18 09:55:01] VERBOSE[10017] logger.c: Adding codec 0x4 (ulaw) to SDP
[May 18 09:55:01] VERBOSE[10017] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[May 18 09:55:01] DEBUG[28907] channel.c: Avoiding initial deadlock for channel '0x893750'
[May 18 09:55:01] VERBOSE[10017] logger.c:
<--- Reliably Transmitting (NAT) to 172.17.3.248:61939 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK10518C0;received=172.17.3.248
From: <sip:730730199 at 172.17.3.248>;tag=E20FA500-53D
To: <sip:730732888 at 172.19.3.150>;tag=as1a442e78
Call-ID: 7864A6E7-42AF11DE-B211C927-51AF51F3 at 172.17.3.248
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:730732888 at 172.19.3.150>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 877709162 877709162 IN IP4 172.19.3.150
s=Asterisk PBX 1.6.0.9
c=IN IP4 172.19.3.150
t=0 0
m=audio 15200 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
[May 18 09:55:01] DEBUG[10017] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 172.17.3.248:61939
[May 18 09:55:01] VERBOSE[28920] logger.c:
<--- SIP read from UDP://172.17.3.248:61939 --->
ACK sip:730732888 at 172.19.3.150:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK106BE5
From: <sip:730730199 at 172.17.3.248>;tag=E20FA500-53D
To: <sip:730732888 at 172.19.3.150>;tag=as1a442e78
Date: Mon, 18 May 2009 06:53:47 GMT
Call-ID: 7864A6E7-42AF11DE-B211C927-51AF51F3 at 172.17.3.248
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
<------------->
[May 18 09:55:01] VERBOSE[28920] logger.c: --- (10 headers 0 lines) ---
[May 18 09:55:01] DEBUG[28920] chan_sip.c: Stopping retransmission on '7864A6E7-42AF11DE-B211C927-51AF51F3 at 172.17.3.248' of Response 101: Match Found
[May 18 09:55:01] VERBOSE[28920] logger.c:
<--- SIP read from UDP://172.17.3.248:61939 --->
BYE sip:730732888 at 172.19.3.150:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK107F4B
From: <sip:730730199 at 172.17.3.248>;tag=E20FA500-53D
To: <sip:730732888 at 172.19.3.150>;tag=as1a442e78
Date: Mon, 18 May 2009 06:53:47 GMT
Call-ID: 7864A6E7-42AF11DE-B211C927-51AF51F3 at 172.17.3.248
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1242629630
CSeq: 102 BYE
Reason: Q.850;cause=47
Content-Length: 0
<------------->
[May 18 09:55:01] VERBOSE[28920] logger.c: --- (12 headers 0 lines) ---
[May 18 09:55:01] DEBUG[28920] chan_sip.c: Initializing initreq for method BYE - callid 7864A6E7-42AF11DE-B211C927-51AF51F3 at 172.17.3.248
[May 18 09:55:01] VERBOSE[28920] logger.c: Sending to 172.17.3.248 : 61939 (NAT)
[May 18 09:55:01] VERBOSE[28920] logger.c:
<--- Transmitting (NAT) to 172.17.3.248:61939 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK107F4B;received=172.17.3.248
From: <sip:730730199 at 172.17.3.248>;tag=E20FA500-53D
To: <sip:730732888 at 172.19.3.150>;tag=as1a442e78
Call-ID: 7864A6E7-42AF11DE-B211C927-51AF51F3 at 172.17.3.248
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<------------>
[May 18 09:55:01] DEBUG[28920] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 172.17.3.248:61939
[May 18 09:55:01] DEBUG[10017] pbx.c: Extension 730732888, priority 3 returned normally even though call was hung up
[May 18 09:55:01] DEBUG[10017] channel.c: Soft-Hanging up channel 'SIP/172.17.3.248-008c4790'
[May 18 09:55:01] DEBUG[10017] channel.c: Hanging up channel 'SIP/172.17.3.248-008c4790'
[May 18 09:55:01] DEBUG[10017] chan_sip.c: Hangup call SIP/172.17.3.248-008c4790, SIP callid 7864A6E7-42AF11DE-B211C927-51AF51F3 at 172.17.3.248
[May 18 09:55:01] VERBOSE[28920] logger.c: Really destroying SIP dialog '7864A6E7-42AF11DE-B211C927-51AF51F3 at 172.17.3.248' Method: BYE
[May 18 09:55:14] VERBOSE[9223] logger.c: -- Remote UNIX connection disconnected
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Initializing initreq for method OPTIONS - callid 6979bc010cad89f67f32e3b508075a5d at 172.19.3.150
[May 18 09:55:21] VERBOSE[28920] logger.c: Reliably Transmitting (NAT) to 172.17.10.150:5060:
OPTIONS sip:172.17.10.150 SIP/2.0
Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK71cec630
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 172.19.3.150>;tag=as55864c67
To: <sip:172.17.10.150>
Contact: <sip:asterisk at 172.19.3.150>
Call-ID: 6979bc010cad89f67f32e3b508075a5d at 172.19.3.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.9
Date: Mon, 18 May 2009 06:55:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
---
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for 172.17.10.150:5060
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Initializing initreq for method OPTIONS - callid 6a5999803ab2c8364e9d6c2a33726e6a at 172.19.3.150
[May 18 09:55:21] VERBOSE[28920] logger.c: Reliably Transmitting (NAT) to 172.17.3.249:5060:
OPTIONS sip:172.17.3.249 SIP/2.0
Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK597b7843;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 172.19.3.150>;tag=as09b5f074
To: <sip:172.17.3.249>
Contact: <sip:asterisk at 172.19.3.150>
Call-ID: 6a5999803ab2c8364e9d6c2a33726e6a at 172.19.3.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.9
Date: Mon, 18 May 2009 06:55:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
---
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for 172.17.3.249:5060
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Initializing initreq for method OPTIONS - callid 512cece90ef7874550fd7bc45676ce4a at 172.19.3.150
[May 18 09:55:21] VERBOSE[28920] logger.c: Reliably Transmitting (NAT) to 172.17.3.248:5060:
OPTIONS sip:172.17.3.248 SIP/2.0
Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK2b7ca253
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 172.19.3.150>;tag=as2cf79799
To: <sip:172.17.3.248>
Contact: <sip:asterisk at 172.19.3.150>
Call-ID: 512cece90ef7874550fd7bc45676ce4a at 172.19.3.150
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.9
Date: Mon, 18 May 2009 06:55:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
---
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for 172.17.3.248:5060
[May 18 09:55:21] VERBOSE[28920] logger.c:
<--- SIP read from UDP://172.17.10.150:5060 --->
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK71cec630
From: "asterisk" <sip:asterisk at 172.19.3.150>;tag=as55864c67
To: <sip:172.17.10.150>
Call-ID: 6979bc010cad89f67f32e3b508075a5d at 172.19.3.150
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
[May 18 09:55:21] VERBOSE[28920] logger.c: --- (7 headers 0 lines) ---
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Stopping retransmission on '6979bc010cad89f67f32e3b508075a5d at 172.19.3.150' of Request 102: Match Found
[May 18 09:55:21] VERBOSE[28920] logger.c: Really destroying SIP dialog '6979bc010cad89f67f32e3b508075a5d at 172.19.3.150' Method: OPTIONS
[May 18 09:55:21] VERBOSE[28920] logger.c:
<--- SIP read from UDP://172.17.3.249:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK597b7843;rport
From: "asterisk" <sip:asterisk at 172.19.3.150>;tag=as09b5f074
To: <sip:172.17.3.249>;tag=6114EB94-1696
Date: Mon, 18 May 2009 06:53:53 GMT
Call-ID: 6a5999803ab2c8364e9d6c2a33726e6a at 172.19.3.150
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 166
v=0
o=CiscoSystemsSIP-GW-UserAgent 3560 8246 IN IP4 172.19.2.41
s=SIP Call
c=IN IP4 172.17.3.249
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 172.17.3.249
<------------->
[May 18 09:55:21] VERBOSE[28920] logger.c: --- (14 headers 7 lines) ---
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Stopping retransmission on '6a5999803ab2c8364e9d6c2a33726e6a at 172.19.3.150' of Request 102: Match Found
[May 18 09:55:21] VERBOSE[28920] logger.c: Really destroying SIP dialog '6a5999803ab2c8364e9d6c2a33726e6a at 172.19.3.150' Method: OPTIONS
[May 18 09:55:21] VERBOSE[28920] logger.c:
<--- SIP read from UDP://172.17.3.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK2b7ca253
From: "asterisk" <sip:asterisk at 172.19.3.150>;tag=as2cf79799
To: <sip:172.17.3.248>;tag=E2100140-13C1
Date: Mon, 18 May 2009 06:54:11 GMT
Call-ID: 512cece90ef7874550fd7bc45676ce4a at 172.19.3.150
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,resource-priority,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 166
v=0
o=CiscoSystemsSIP-GW-UserAgent 1387 515 IN IP4 172.17.3.248
s=SIP Call
c=IN IP4 172.17.3.248
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 172.17.3.248
<------------->
[May 18 09:55:21] VERBOSE[28920] logger.c: --- (14 headers 7 lines) ---
[May 18 09:55:21] DEBUG[28920] chan_sip.c: Stopping retransmission on '512cece90ef7874550fd7bc45676ce4a at 172.19.3.150' of Request 102: Match Found
[May 18 09:55:21] VERBOSE[28920] logger.c: Really destroying SIP dialog '512cece90ef7874550fd7bc45676ce4a at 172.19.3.150' Method: OPTIONS
-------------- next part --------------
May 18 06:57:22.762: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x0E82
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98385
Exclusive, Channel 5
Calling Party Number i = 0x2183, '730730199'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '730732888'
Plan:ISDN, Type:National
May 18 06:57:22.770: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8E82
Channel ID i = 0xA98385
Exclusive, Channel 5
OULKLAVG2#
May 18 06:57:25.778: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8 callref = 0x8E82
May 18 06:57:25.790: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E82
May 18 06:57:25.790: %ISDN-6-CONNECT: Interface Serial0/0/1:4 is now connected to 730730199 N/A
May 18 06:57:25.930: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E82
Cause i = 0x82AF - Resource unavailable, unspecified
May 18 06:57:25.930: %ISDN-6-CONNECT: Interface Serial0/0/1:4 is now connected to 730730199 N/A
May 18 06:57:25.930: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8E82
May 18 06:57:25.942: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E82
Cause i = 0x80D1 - Invalid call reference value
May 18 06:57:25.954: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E82
May 18 06:57:25.954: ISDN Se0/0/1:15 **ERROR**: Ux_BadMsg: Invalid Message for call state 19, call id 0x3611, call ref 0x8E82, event 0xF
May 18 06:57:25.954: ISDN Se0/0/1:15 Q931: TX -> STATUS pd = 8 callref = 0x8E82
Cause i = 0x80E20F - Message not compatible with call state or not implemented
Call State i = 0x13
May 18 06:57:25.962: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E82
Cause i = 0x82AF - Resource unavailable, unspecified
May 18 06:57:25.966: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000
Channel ID i = 0xA98385
Exclusive, Channel 5
Restart Indicator i = 0x80
May 18 06:57:25.966: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000
Channel ID i = 0xA98385
Exclusive, Channel 5
Restart Indicator i = 0x80
May 18 06:57:25.978: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x0E83
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98395
Exclusive, Channel 21
Calling Party Number i = 0x2183, '730730199'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '730732888'
Plan:ISDN, Type:National
May 18 06:57:25.978: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E82
Cause i = 0x80E5 - Message not compatible with call state
May 18 06:57:25.982: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8E82 callid 0x0
OULKLAVG2#
May 18 06:57:25.982: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8E83
Channel ID i = 0xA98395
Exclusive, Channel 21
OULKLAVG2#
May 18 06:57:29.054: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8 callref = 0x8E83
May 18 06:57:29.066: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E83
May 18 06:57:29.066: %ISDN-6-CONNECT: Interface Serial0/0/1:20 is now connected to 730730199 N/A
May 18 06:57:29.206: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E83
Cause i = 0x82AF - Resource unavailable, unspecified
May 18 06:57:29.206: %ISDN-6-CONNECT: Interface Serial0/0/1:20 is now connected to 730730199 N/A
OULKLAVG2#
May 18 06:57:29.206: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8E83
May 18 06:57:29.210: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000
Channel ID i = 0xA98395
Exclusive, Channel 21
Restart Indicator i = 0x80
May 18 06:57:29.210: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000
Channel ID i = 0xA98395
Exclusive, Channel 21
Restart Indicator i = 0x80
May 18 06:57:29.218: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E83
Cause i = 0x80D1 - Invalid call reference value
May 18 06:57:29.218: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8E83 callid 0x0
Cause i = 0x82AF - Resource unavailable, unspecified
May 18 06:57:25.930: %ISDN-6-CONNECT: Interface Serial0/0/1:4 is now connected to 730730199 N/A
May 18 06:57:25.930: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8E82
May 18 06:57:25.942: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E82
Cause i = 0x80D1 - Invalid call reference value
May 18 06:57:25.954: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E82
May 18 06:57:25.954: ISDN Se0/0/1:15 **ERROR**: Ux_BadMsg: Invalid Message for call state 19, call id 0x3611, call ref 0x8E82, event 0xF
May 18 06:57:25.954: ISDN Se0/0/1:15 Q931: TX -> STATUS pd = 8 callref = 0x8E82
Cause i = 0x80E20F - Message not compatible with call state or not implemented
Call State i = 0x13
May 18 06:57:25.962: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E82
Cause i = 0x82AF - Resource unavailable, unspecified
May 18 06:57:25.966: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000
Channel ID i = 0xA98385
Exclusive, Channel 5
Restart Indicator i = 0x80
May 18 06:57:25.966: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000
Channel ID i = 0xA98385
Exclusive, Channel 5
Restart Indicator i = 0x80
May 18 06:57:25.978: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x0E83
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98395
Exclusive, Channel 21
Calling Party Number i = 0x2183, '730730199'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '730732888'
Plan:ISDN, Type:National
May 18 06:57:25.978: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E82
Cause i = 0x80E5 - Message not compatible with call state
May 18 06:57:25.982: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8E82 callid 0x0
OULKLAVG2#
May 18 06:57:25.982: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8E83
Channel ID i = 0xA98395
Exclusive, Channel 21
OULKLAVG2#
May 18 06:57:29.054: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8 callref = 0x8E83
May 18 06:57:29.066: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E83
May 18 06:57:29.066: %ISDN-6-CONNECT: Interface Serial0/0/1:20 is now connected to 730730199 N/A
May 18 06:57:29.206: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E83
Cause i = 0x82AF - Resource unavailable, unspecified
May 18 06:57:29.206: %ISDN-6-CONNECT: Interface Serial0/0/1:20 is now connected to 730730199 N/A
OULKLAVG2#
May 18 06:57:29.206: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8E83
May 18 06:57:29.210: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000
Channel ID i = 0xA98395
Exclusive, Channel 21
Restart Indicator i = 0x80
May 18 06:57:29.210: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000
Channel ID i = 0xA98395
Exclusive, Channel 21
Restart Indicator i = 0x80
May 18 06:57:29.218: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E83
Cause i = 0x80D1 - Invalid call reference value
May 18 06:57:29.218: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8E83 callid 0x0
More information about the asterisk-users
mailing list