Cisco资深专家在线解决全世界用户voip问题集锦－连载六
28.............
debug infor about FXO port
shan-cao - CHENGDU, GOLDTECH GROUP
Jul 18, 2003, 2:22am PST
I have a debug information showed by command"debug vpm all"about FXO port.I can't distinguish the error in the information.
who can tell me how to find out the error?
Thanks.
debug infor:
3d18h: ccIFShowState (vdbPtr=0x62F47D14, summary)
3d18h: ccIFShowState (vdbPtr=0x62F49994, summary)
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x0 timestamp=48919 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_rig
3d18h: [3/0/0] htsp_start_caller_id_rx
3d18h: [3/0/0] htsp_set_caller_id_rx:BELLCORE
3d18h: htsp_timer - 125 msec
3d18h: htsp_process_event: [3/0/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxolr
3d18h: htsp_timer - 10000 msec
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x4 timestamp=49519 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
3d18h: fxols_ringing_not
3d18h: htsp_timer_stop
3d18h: htsp_timer - 10000 msec
3d18h: [3/0/0] htsp_stop_caller_id_rx
3d18h: hdsprm_close_cleanup
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x0 timestamp=51819 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_RINGING, E_DSP_SIG_0000]
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x4 timestamp=52819 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
3d18h: fxols_ringing_not
3d18h: htsp_timer_stop htsp_setup_ind
3d18h: [3/0/0] get_fxo_caller_id:Caller ID received. Message type=4 length=18 c3
3d18h: [3/0/0] Caller ID String 04 0F 30 37 31 38 31 35 35 31 35 32 36 33 39 33
3d18h: [3/0/0] get_fxo_caller_id calling num=5263932 calling name= calling time
3d18h: cc_api_call_setup_ind (vdbPtr=0x62F47D14, callInfo={called=8059,called_o)
3d18h: cc_api_call_setup_ind type 2 , prot 0
3d18h: cc_insert_call_entry: Increment call volume counter
3d18h: cc_insert_call_entry: current call volume: 1
3d18h: cc_insert_call_entry: entry's incoming TRUE. is_incoming is TRUE
3d18h: cc_incr_if_call_volume: not the VoIP or MMoIP
3d18h: htsp_process_event: [3/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
3d18h: fxols_wait_setup_ack:
3d18h: [3/0/0] set signal state = 0xC timestamp = 0
3d18h: dsp_set_sig_state: [3/0/0] packet_len=12 channel_id=128 packet_id=39 sta0
3d18h: dsp_soutput: [3/0/0]fxols_check_auto_call
3d18h: cc_process_call_setup_ind (event=0x62ED923C)
3d18h: CCAPI handed cid 25 with tag 200 to app "DEFAULT"
3d18h: sess_appl: ev(24=CC_EV_CALL_SETUP_IND), cid(25), disp(0)
3d18h: sess_appl: ev(SSA_EV_CALL_SETUP_IND), cid(25), disp(0)
3d18h: ssaCallSetupInd
3d18h: ccCallSetContext (callID=0x19, context=0x631FB148)
3d18h: ssaCallSetupInd cid(25), st(SSA_CS_MAPPING),oldst(0), ev(24)ev-e.evCall1
3d18h: ssaCallSetupInd finalDest cllng(5263932), clled(8059)
3d18h: ssaCallSetupInd cid(25), st(SSA_CS_CALL_SETTING),oldst(0), ev(24)dpMatch0
3d18h: ssaSetupPeer cid(25) peer list: tag(100) called number (8059)
3d18h: ssaSetupPeer cid(25), destPat(8059), matched(4), prefix(), peer(62A8DF54)
3d18h: ccCallProceeding (callID=0x19, prog_ind=0x0)
3d18h: ccCallSetupRequest (Inbound call = 0x19, outbound peer =100, dest=, para1
3d18h: ccCallSetupRequest numbering_type 0x81
3d18h: ccCallSetupRequest encapType 2 clid_restrict_disable 1 null_orig_clg 0 c2
3d18h: dest pattern 8059, called 8059, digit_strip 0
3d18h: callingNumber=5263932, calledNumber=8059, redirectNumber= display_info= 0
3d18h: accountNumber=, finalDestFlag=1,
guid=7ea5.51a9.17e5.11cc.8034.e670.5153.4d65
3d18h: peer_tag=100
3d18h: ccIFCallSetupRequestPrivate: (vdbPtr=0x62CDA89C, dest=, callParams={call1
3d18h: ccIFCallSetupRequestPrivate: (vdbPtr=0x62CDA89C, dest=, callParams={call)
3d18h: cc_insert_call_entry: not incoming entry
3d18h: cc_insert_call_entry: entry's incoming FALSE. is_incoming is FALSE
3d18h: ccSaveDialpeerTag (callID=0x19, dialpeer_tag=0x64)
3d18h: ccCallSetContext (callID=0x1A, context=0x631FB6BC)
3d18h: ccCallReportDigits (callID=0x19, enable=0x0)
3d18h: cc_api_call_report_digits_done (vdbPtr=0x62F47D14, callID=0x19, disp=0)
3d18h: sess_appl: ev(52=CC_EV_CALL_REPORT_DIGITS_DONE), cid(25), disp(0)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(1)
3d18h: -cid2(26)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_MAPPING)
3d18h: ssaReportDigitsDone cid(25) peer list: (empty)
3d18h: ssaReportDigitsDone callid=25 Reporting disabled.
3d18h: htsp_process_event: [3/0/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_ofc
3d18h: htsp_timer - 120000 msec
3d18h: cc_api_supported_data data_mode=0x10002
3d18h: cc_incr_if_call_volume: remote IP is x.x.x.x
3d18h: cc_incr_if_call_volume: hwidb is Serial1/0:0
3d18h: cc_incr_if_call_volume: create entry in list: 1
3d18h: ccTDUtilGetInstanceCount: For tagID[1] of callID[26]
3d18h: ccTDPvtProfileTableObjectAccessManager: No profileTable set for callID[2]
3d18h: ccTDUtilGetInstanceCount: For tagID[2] of callID[26]
3d18h: ccTDPvtProfileTableObjectAccessManager: No profileTable set for callID[2]
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0xC timestamp=53222 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_PROCEEDING, E_DSP_SIG_1100]fxols_offhoc
3d18h: htsp_timer2 - 350 msec
3d18h: cc_api_call_proceeding(vdbPtr=0x62CDA89C, callID=0x1A,
prog_ind=0x0, rawmsgPtr=0x0)
3d18h: sess_appl: ev(21=CC_EV_CALL_PROCEEDING), cid(26), disp(0)
3d18h: cid(26)st(SSA_CS_CALL_SETTING)ev(SSA_EV_CALL_PROCEEDING)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(0)fDest(0)
3d18h: -cid2(25)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_CALL_SETTING)
3d18h: ssaCallProc
3d18h: ccGetDialpeerTag (callID=0x19)
3d18h: ssaIgnore cid(26), st(SSA_CS_CALL_SETTING),oldst(1), ev(21)
3d18h: cc_api_call_alert(vdbPtr=0x62CDA89C, callID=0x1A, prog_ind=0x0, sig_ind=)
3d18h: sess_appl: ev(7=CC_EV_CALL_ALERT), cid(26), disp(0)
3d18h: cid(26)st(SSA_CS_CALL_SETTING)ev(SSA_EV_CALL_ALERT)
oldst(SSA_CS_CALL_SETTING)cfid(-1)csize(0)in(0)fDest(0)
3d18h: -cid2(25)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_CALL_SETTING)
3d18h: ssaAlert
3d18h: ccGetDialpeerTag (callID=0x19)
3d18h: ccCallAlert (callID=0x19, prog_ind=0x0, sig_ind=0x1)htsp_alert_notify
3d18h: htsp_process_event: [3/0/0, FXOLS_PROCEEDING, E_HTSP_EVENT_TIMER2]fxols_m
3d18h: htsp_timer_stop
3d18h: htsp_timer_stop2
3d18h: cc_api_call_disconnected(vdbPtr=0x62F47D14, callID=0x19, cause=0x10)
3d18h: sess_appl: ev(11=CC_EV_CALL_DISCONNECTED), cid(25), disp(0)
3d18h: cid(25)st(SSA_CS_ALERT_RCVD)ev(SSA_EV_CALL_DISCONNECTED)
oldst(SSA_CS_CALL_SETTING)cfid(-1)csize(0)in(1)fDest(1)
3d18h: -cid2(26)st2(SSA_CS_ALERT_RCVD)oldst2(SSA_CS_CALL_SETTING)
3d18h: ssaDiscSetting
3d18h: ssaFlushPeerTagQueue cid(25) peer list: (empty)
3d18h: ssa: Disconnected cid(25) state(17) cause(0x10)
3d18h: ccCallDisconnect (callID=0x19, cause=0x10 tag=0x0)
3d18h: cc_api_get_transfer_info: (callID=0x19)
3d18h: ccCallDisconnect (callID=0x1A, cause=0x10 tag=0x0)
bazhong#sh voice call su
PORT CODEC VAD VTSP STATE VPM STATE
============ ======== === ==================== ======================
3/0/0 None y S_SETUP_IND_PEND FXOLS_RINGING
3/0/1 - - - FXOLS_ONHOOK
Sincerely
thusain - CISCO SYSTEMS, CCIE
Jul 20, 2003, 3:27am PST
there does not seem to be an error. it seems that the call was in alerting state, the phone was possibly ringing and the
originating party dropped the call.
It is also helpful to turn on
service timestamps debug date msec
for milisecond debugging.
shan-cao - CHENGDU, GOLDTECH GROUP
Jul 20, 2003, 7:18pm PST
I just received the response from customer engineer about the issue that the problem is the IVR cannot make the correct
response to the call from the branch Office. the originated caller only hear the normal ring, but can't hear the IVR replied
sound just like" welcome to xxx company". Could u tell me how to make the right troubleshooting to solve the problem? I have
a question that whether the call can arrive the network gateway correctly, but can't be transferred to the call manager
gateway correctly. Wait for your help. Thanks
Sincerely
29.............
Rookies ask about dial-peer
weltje@sunvone.com
Jul 16, 2003, 10:17pm PST
Dear All,
I'm a beginner in cisco and I need help. My boss ask me to configure our as5300 gateway (we call it GW B). GW B will gather
all calls from gateway A(as5300) before to send it to other as5300 gateway (GW C).
How do I configure dial-peer at GW B, because as I know, if I receive calls from GW A (send to B with session target ipv4), I
can not forward the call to GW C with session target ipv4.
I'm really need help. Thank you.
Best regards,
Weltje
thusain - CISCO SYSTEMS, CCIE
Jul 20, 2003, 3:25am PST
Your GW B needs to act as an ip ip gateway. Please refer to the ip ip gateway documents and the feature release in 12.2.13T
30...........
As5350
dynacom_eng@dynanet.lk - Sales Engineer, DYNACOM ELECTRONICS PTE LTD
Jul 14, 2003, 11:30am PST
Connected calls are remain as it is although the link is
down. And also the other side of the E1 cannot see this this down. But once it happen the 5350 does not process any new
calls.
This is very critical at this stage since i can see its even not 50%
utilize the box once this is happening. I have also bring back the IOS from 12.3.1 to 12.2.15 but its still same.
Regards,
Noufel
sbilgi
Jul 18, 2003, 11:33am PST
I guess you running into the following bug:CSCdx47512.
31...........
VXML on the AS5350
pmont - TNTLogistics
Jul 9, 2003, 7:22am PST
I am implementing an IVR system using the AS5350 and I am sure that I will have a bunch of questions about the details of the
implementation of the VXML 2.0 spec within it. Is there an active forum other than this one that is set up to address this?
Are the conformance scripts available anywhere online? I have seen the conformance table:
http://www.cisco.com/univercd/cc ... vxmlprg/refgap2.htm
But I was interested in the scripts that were used to prove conformance. Having those available online would help to answer a
lot of questions people like me might have.
Thanks.
rterhune - DNS INC
Jul 16, 2003, 8:16am PST
If you do a search on CCO for "VoiceXML" you will find some sample apps. The page I started with was:
http://www.cisco.com/univercd/cc ... vxmlprg/refgde3.htm
pmont - TNTLogistics
Jul 16, 2003, 10:24am PST
Thanks, That is a good document and one I have used.
The set of documents provided by Cisco are good, I was just trying to forestall any questions that will probably arise out of
using features that are not "Exampled" in the docs you referenced.
To be VXML 2.0 "compliant" I would assume that they have to have a set of scripts that exercise the browser. There is one
sUCh set at:
http://www.voicexml.org/conformance/samples/
but it is a generic set and not of much use beyond the simple stuff. I would think that the publishing of the scripts that
Cisco uses would be usefull to all of us. They might however feel that they are proprietary.
Thanks
Paul.
32...........
"Fancy" call routing on Cisco GK?
m.laporta
Jun 24, 2003, 9:21am PST
Hi EXPerts.
My Customer is (will be) a Wholesale VoIP Service Provider, taking traffic from and giving traffic to VoIP Service Providers.
I've advised to build a Cisco-based H.323 network with Cisco MCM GK and IP-IP-GW and now would like to know whether this
network (esp. the GK) is able to perform an intelligent call routing: as downstream VoIP Service Providers eventually lead to
the same destination, is it possible to associate a composed metric to them and to choose the best downstream SP based on
priority, committed "circuits" or something like that?
Or, if not, are there any fancy call routing mechanisms in a Cisco Gatekeeper?
Thank you!
michele
sbilgi
Jun 30, 2003, 8:38am PST
I have heard that fancy routing is not permitted in the gatekeepers.
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rterhune - DNS INC
Jul 16, 2003, 9:03am PST
Michele this may help you.
...Snip from CCO
http://www.cisco.com/en/US/partn ... w/ps4371/index.Html
The Cisco Carrier Sensitive Route (CSR) Server is a software application that provides advanced, rules-based routing logic
within a Cisco H.323 VoIP network. Targeted at the highly competitive wholesale voice marketplace, the Cisco CSR Server
allows service providers to optimize the routing decisions within their network to fit their business needs. The routing
decision within the Cisco CSR Server can be made based on:
Least cost
Best quality (QoS)
Time of day
Percent allocation to different carriers
In addition, the routing rules can combine several of these parameters in different orders to form flexible and powerful
routing logic.
The Cisco CSR Server runs on the Solaris 2.8 operating system on top of reliable, carrier-grade Sun hardware platforms and
works with the proven family of Cisco H.323 gatekeepers using the Gatekeeper Transaction Management Protocol (GKTMP). The
Cisco CSR Server offers a Graphical User Interface (GUI) for day-to-day configuration and management tasks plus a Structured
Query Language (SQL) interface and an open database schema for bulk provisioning of routing data and for integration with
existing back office systems.
The Cisco CSR Server is an integral part of the Cisco Voice Infrastructure and Applications (VIA) Solution, giving service
providers the power to customize their routing logic and to maximize the profitability of their networks.
33..........
gateway to gateway security best practices
anilbatra007 - SR. ENGINEER, CISCO CONSULTANT PROGRAM
Jul 10, 2003, 12:55am PST
We have a client who is voip carrier , he is going to terminate voip traffic from peering gateways dierectly without using
gatkeeper.How can he make sure that he is going to accept voip traffic from authorized carrier's gatwway only and not from
anyone else gateway.I know one solution could be using accesslist and barring the traffic from unknown IP address, but the
issue here is that the partners can always add or cahnge the gateways and their IP address , so is there any other way that
he accespts traffic frorm authorized parteners only.Also is there any best pratcises document available for gateway to
gateway security.
Thanks and regards,
Anil
ebreniz
Jul 16, 2003, 8:01am PST
How about reviewing and changing the passWords on a regular basis?
34.............
AS5350 with AS5300
anazarenko
Jul 12, 2003, 10:50pm PST
Our partner is using AS5350 and we have problem with g729r8 codec between these gateways.
We use codec = g729r8, payload size = 20 bytes on both ends, but when AS5300 terminating calls I see calls with pre-ietf
codec
22E6 : 30114375hs.147859 +338 +357 pid:3 Answer xxxx
dur 00:00:00 tx:0/0 rx:0/0 2C (no requested circuit.)
IP x.x.x.x:18770 rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf
About 30% of all calls come with right g729r8 codec (non pre-ietf) and AS5300 can successfully terminate them. But pre-eitf
calls get "dsp error" on further dial-peers.
1353 : 30163075hs.147862 +-1 +347 pid:9821 Originate xxxx
dur 00:00:00 tx:0/0 rx:10/1600 AC (dsp error)
IP x.x.x.x:18892 rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms clear-channel
(Call passes PSTN Loopback before reaching this dial-peer)
I've read that AS5350 doesn't support real g729r8 and g729br8 high-complex codecs, but it support only medium-complex g729
ones. Can this be the cause of problem?
thusain - CISCO SYSTEMS, CCIE
Jul 14, 2003, 1:41am PST
Are the pre-ietf codecs coming from a cisco gateway. If they are then you can upgrade that router, to something higher than
12.1 and it will not send pre ietf anymore. ALternatively you can configure your 5350 to negotiate the pre ietf codec, by
configuring g729 pre-ietf under the dial peer or voice class.
Taimoor
35........
Call record question on AS5350
v.chow - ENGINEER, MACROVIEW TELECOM LIMITED
Jun 26, 2003, 9:13am PST
Dear,
I have implemented several AS5350 as voip gateway. I want to retrieve the call details from each gateway into a Linux server
which already run the radius . The call details should not only include radius attributes but also include the packet loss,
delay, jitter, echo detail that already show by "sh call act voice brief".
Is there any method to collect the above information by a server after each end of calls? This information will be useful for
quality measurement (not just accounting)!
Vincent
thusain - CISCO SYSTEMS, CCIE
Jul 8, 2003, 10:31am PST
You can use AAA accounting with raidus and get all the vsa's that cisco gateway sends for h323. Cisco gateways send a voice
quality attribute as well, that indicates the quality of the voice call. However they do not send details of the packet loss
and jitter etc. If you need to get that information, you would have to poll the gateway using an snmp mib or something, i am
not aware of a mib that polls that information.
You should look into QoS tools such as QPM or QDM which give you more real time information about QoS on voice calls. Please
contact your local cisco team for that.
Hope this helps.
