Troubleshooting no ringback voice gateway
Hi all,
I'd like to ask about voice gateway that installed on my customer site. below is the topology:
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-parent:"";
mso-padding-alt:0in 5.4pt 0in 5.4pt;
mso-para-margin:0in;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:10.0pt;
font-family:"Times New Roman","serif";}
PSTN – Voice Gateway – C2960-48PST-L – Callmanager and IP Phone
|
|(FO)
Analog phone – PBX.A – Voice Gateway –––– Voice Gateway – PBX.B – Analog phone
the problem is that there's no ring back tone when calling to another site analog phone from ip phone/analog phone. The PBX (NEC) are using E1 connection to the router. Does the ringback provided by PBX or the Voice gateway?
Thanks in advance
Dias
Yes, we did find a solution. It ended up it was on the carrier's end. I believe that they were generating ring back inbound to calls we originated, but not outbound to calls originated from the carriers. They switched that so that they did not send us ring back, but did send carriers ring back for inbound calls and that appeared to solve the issue.
Here's my carrier's explanation in case that wasn't confusing enough: "We made in-band tone available to the SIP trunk via an egress profile change on the SONUS Gateway. Early-media is determined / initiated by
the called or far-end switch."
Similar Messages
-
Troubleshooting SS7 Interconnect for Voice Gateway
Hi
Recently we have implemeted SS7 Interconnect for Voice Gateway Solution in our network.
We are using devices PGW2200, NAS 5400, SLT 2611, 4000 Series S/w and GK.
Customers compalint that they are facing Echo, Clip in Voice and No voice related issues. Can you tell us the commands on PGW/GW/SLT to pinpoint. PGW we are using in Call Control Mode.
Please help.
Thanks in Advance.Hi,
You can check in the /opt/CiscoMGC/etc directory properties.dat file which has the parameter *.EchoCanRequired which should be set to 1. This can binded by an individual trunk-group as well. Also make sure the voice-ports have the echo-cancel coverage and disable VAD to see if it makes it better.
Regards
-Kashif -
I have multiple environments with Avaya Partner phone systems in them. I used touse a multitech VOIP modem for connection back to Headquarters - this works perfectly fine. Situation I am having, when the Multitech units die, I replace them with the Cisco VG202 unit. When placed at the remote locations, the time in the windows of the Avaya phones never corrects it self...it always wrong. It is correct on the Avaya system at the location, but the integration of the Cisco unit seems to make it go awry. All other functionality is there, but the time display is a mess. Any ideas as to where to troubleshoot this? Manually establish the correct time from a console phone at the remote locations, and it goes awry within hours.
Hi Peter,
This forum is for discussions about specific known bugs. You might find better discussion in the IP Telephony forum: https://supportforums.cisco.com/community/netpro/collaboration-voice-video/ip-telephony
Conceptually as an analog voice gateway the VG shouldn't generally provide any time information to an attached phone system. Is it possible the phone system is pointed to the VG as an NTP reference? In that situation the internal platform clock isn't terribly reliable and synch to an more reliable external NTP reference is recommended. The only other possibility that comes to mind is delivery of a timestamp information along with callerid over the analog voice ports. In that situation the router again depends on internal clock. Configuring an NTP reference on the router will help to keep the router clock better synchronized. These are all a bit of a stretch. The most likely, IMHO, is the pone system pointing to VG202 as NTP reference. In that case giving the VG202 a reliable NTP reference so the router can keep more accurate time would be my first approach.
Hope this helps,
Wes -
Cisco 2911 Voice Gateway SIP PSTN Calls Fail
Hello All,
I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway. 2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy. Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below). does anyone have any insight on how to correct this? Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call. Thanks in advance for any help!!
From: <sip:[email protected]>:tag=6166CDC4-882
To: <sip:[email protected]>
Shawn C. Smithi have same problem my cucm ip is 192.168.200.53
my Voice Gateway is SIP by ip 192.168.200.86 for internal
and 172.29.7.94
and my SIP Server is 10.208.9.69
if its oky can yuo take a look at my problem please
this is the syslog from debug
May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
Session-Expires: 1800
P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=2217156
----- ccCallInfo IE subfields -----
cisco-ani=2217156
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=90555769123
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
Interface=0x30CF41D4, Call Info(
Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=90555769123(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 1
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=90555769123(TON=Unknown, NPI=Unknown))
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
Event=0x2B82D890
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 90555769123
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
Context=0x2ABC2E44
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=0555769123(TON=Unknown, NPI=Unknown),
Redirect Number=, Display Info=Aysar Mohamed
Account Number=2217156, Final Destination Flag=TRUE,
Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=2217156
----- ccCallInfo IE subfields -----
cisco-ani=2217156
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=0555769123
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 2
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
Context=0x2ABC1984
May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
Outgoing Dial-peer=802
May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
Interface=0x30CF41D4, Progress Indication=NULL(0)
May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1401481174
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Contact: <sip:[email protected]:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 328
Content-Type: application/sdp
v=0
o=- 17192647 17192647 IN IP4 10.208.9.69
s=SBC call
c=IN IP4 10.208.9.69
t=0 0
m=audio 39910 RTP/AVP 8 0 102 102 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=fmtp:116 0-15
a=fmtp:18 annexb=yes
May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
Modem=0x0, Codec Bytes=160, Signal Type=2)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=1000(ms), Fax Nom=300(ms))
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event=170, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event=98, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
Cause Value=0
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
(confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
(confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
delay media to slow start case, codec negotation is not done
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0x16, Destination Call Id=466)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
Call Entry(Conference Id=0x16, Destination Call Id=465)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
Conference Id=0x16, Call Id1=465, Call Id2=466
May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 233
v=0
o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
s=SIP Call
c=IN IP4 192.168.200.86
t=0 0
m=audio 18288 RTP/AVP 8 0 19
c=IN IP4 192.168.200.86
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
Cause Value=41, Interface=0x30CF41D4, Call Id=466
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=466
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
Conference Id=0x16, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: vsacount in free is 1
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=41
Content-Length: 0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: vsacount in free is 0
May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.7.94:5060 SIP/2.0
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>;tag=739BBC-1CE2
Date: Fri, 30 May 2014 20:19:36 GMT
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 446
v=0
o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
s=SIP Call
c=IN IP4 172.29.7.94
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.7.94
m=image 0 udptl t38
c=IN IP4 172.29.7.94
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
My SIP GW internal ip address is 192.168.200.86
and the Public IP is : 172.29.7.94
My CUCM is 192.168.200.53
my GW Config is :
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9\(\)/ /\1/
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
voice translation-profile OUT
translate called 3
voice translation-profile REDIAL
translate calling 5
voice translation-profile SIP-NEW
translate called 4
application
service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
license udi pid CISCO2921/K9 sn FCZ164960G0
hw-module pvdm 0/0
hw-module pvdm 0/1
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.200.86 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 172.29.7.94 255.255.255.252
duplex auto
speed auto
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 192.168.200.1
ip route 10.208.9.0 255.255.255.0 172.29.7.93
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register NAGHI-MTP
dspfarm profile 2 mtp
codec g711alaw
maximum sessions hardware 25
associate application SCCP
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 813 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 814 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 815 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 816 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 817 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 818 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Please i need ur help ASAP -
Where can I learn the structure of Voice gateway ?
I'm making a essay about Structure of Voice gateway: hardware and software construction. I can not find any books or any manual deal with it . So,can someone give me more information about structure of voice gateway or give me some useful resources please?
I'm waiting for your replying.Cisco doesn't make public the software structure of their products; to a certain extent it can be inferred from documentation and familiarity with the product, but probably not enough for an in-deep analysis.
For your task, I suggest you focus on some open-source development of GW, that has no secrets.
Hope this helps, please rate post if it does! -
How to create a Global Contacts in our CM or Voice Gateway
Hi
we have a UCM6.1.2 and a H.323 voice gateway
we get many calls from different vendors and so on
i want to some how assign a contact or a name to the calls that come in often
for example
if we get 100 calls from 973-333-3333 i want that number to show up and also the customer's name for example Customer XYZ
can i do this any where?
is it possible?
any help will be appreciated.
Thanks
RegardsYou can configure Caller ID which is an analog service offered by a central office, which supplies calling party information to subscribers.
Caller ID Name Delivery Issues on Cisco IOS Gateways:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml
Caller ID:
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vclclid.html -
cucm 10.1version - any free training videos and hand guides on understanding voice gateway h323 and SIP and how to configure one? thanks
Learncisco gives a very good introduction to CUCM - I recommend you start there.
-
Setup of 2 B-channel transfer (TBCT) on Cisco Voice Gateway
Hi all,
In my scenario, I have several remote locations, each with their own T1-PRI connection to the PSTN via Cisco voice gateways.
The call flow has the caller calling a PSTN number at the remote location. This call hits a CUCM translation pattern and is immediately hairpinned/tromboned out to a PSTN number located at the UCCE central location.
The customer would like to perform a 2 b-channel transfer back up to the CO switch, to free up B-channels to the remote location.
My question is this: Assuming my switch type is supported and the PSTN provider supports 2 b-channel transfers, do I simply need to add "isdn supp-service tbct" to my serial interface configuration?
I have seen discussions that it is as simple as this, to needing some tcl/xml code, to also requiring aaa to be setup.
As there are over 100 locations being considered for this enhancement, the customer would like to know if it is 15 minutes or 2 hours per location to implement.
Thanks,
JoeThanks Paolo,
That was my expectation.
They have this going on the VG's utilized by CVP, so I will ask them to review the XML in use there for tweaking for the remote locations.
Thanks again!
Joe -
Hi all,
I am configuring a SIP voice gateway. After finsihing the config, the outbound calls are working properly. But I met some issue for inbound calls.
There is no debug log for show ccsip message and show voice dialpeer command. Only q931 has the debug log.
Can any one have any idea for this issue?
below is the debug shows for show isdn q931.
Jun 1 05:27:49.435: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x03A3
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xE1858384
Preferred, Interface 5, Channel 4
Calling Party Number i = 0x2181, '18600586101'
Plan:ISDN, Type:National
Called Party Number i = 0xC1, '82197910'
Plan:ISDN, Type:Subscriber(local)
High Layer Compat i = 0x9181
High Layer Compat i = 0x9181
Jun 1 05:27:49.439: ISDN Se0/0/0:15 Q931: Received SETUP callref = 0x83A3 callID = 0x0004 switch = primary-net5 interface = User
d8w-sr-2811f20a-vpn#
Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting call
Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3
Cause i = 0x80E418 - Invalid information element contents
下午01:28:17: Zhuliang: Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting call
Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3
Cause i = 0x80E418 - Invalid information element contentsChannel ID i = 0xE1858384 Preferred, Interface 5, Channel 4 Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3 Cause i = 0x80E418 - Invalid information element contents下午01:28:17: Zhuliang: Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting callJun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3 Cause i = 0x80E418 - Invalid information element contents
To what are you connecting? Call comes with explicit interface indicator IE, that is not normal for ISDN E1. -
Can I set an ISR 2951 as CUBE to receive SIP trunks and configure the same box as voice gateway to deliver TDM E1 voice channels to an enterprise PBx?
Thanks
Sent from Cisco Technical Support iPhone AppYes You can...
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay" -
Configure E1 card on voice gateway
Hi
I have CUCM 9 and Cisco 2801 voice gateway
There is 2 card on cisco 2801 : VIC2-4FXO and VWIC3-1MFT-T1/E1
I would like to know how I will configure the gateway in the CUCM (H323, MGCP ...)
Should I add H323 or MGCP gateway or something else ?
I also need to know how I will configure E1 card on the Cisco 2801 to handle CUCM PSTN incoming and outgoing call
I will apreciate if you help with some how to document
Thanks in advance
RegardsWhich protocol to use, is up to you, you should know the requirements of your customer to define that.
-
How to Save Global Contacts in our CM or Voice Gateway
Hi
we have a UCM6.1.2 and a H.323 voice gateway
we get many calls from different vendors and so on
i want to some how assign a contact or a name to the calls that come in often
for example
if we get 100 calls from 973-333-3333 i want that number to show up and also the customer's name for example Customer XYZ
can i do this any where?
is it possible?
any help will be appreciated.
Thanks
RegardsYou can configure Caller ID which is an analog service offered by a central office, which supplies calling party information to subscribers.
Caller ID Name Delivery Issues on Cisco IOS Gateways:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml
Caller ID:
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vclclid.html -
How to install permenant UC License on 2951 voice gateway
Hi All,
I have C2951 Voice gateway and I want to install Permenant UC license and I don't know how to get it and how to install on it and how to activate it,so I would like to request all of you that if you have any idea please explain me in details with step by step, i will be appreciate all of you and hop I will receive response form you very soon.
Thanks
RizwanHi Rizwan,
a. u need to send the mail to [email protected] along with PAK code and serial number of 2951 router.this u can get by running command show version or show license udi
b. after u recive the license file[uck9_license.lic], upload this file in router flash using command copy tftp flash:
c. u can check the same by running command show flash:
d. now u need to install the license file . run the command license install flash:uck9_license.lic
e. now run the command license boot module c2900 technology-package uck9 and accept :yes
f. reboot the router
g. after the router reboots, run command show license feature /sh license all.
regds,
aman -
How to verify voice channels on voice gateway
Hello,
I am in the testing phase of deploying a 2921 voice gateway with 2 PRIs. After connecting the T1s, how do I verify if all the channels are up and ready for use? Also, how do I verify if clocking on the T1 interfaces are configured correctly?
This 2921 is a replacement for a 2811 currently being used. When I login to the 2811 via SSH, I see this:
560240: Aug 13 20:37:24.368: %ISDN-6-CONNECT: Interface Serial0/2/0:3 is now connected to 3854143418 N/A
560241: Aug 13 20:37:24.368: %ISDN-6-CONNECT: Interface Serial0/2/0:3 is now connected to 3854143418 N/A
560242: Aug 13 20:37:24.544: %ISDN-6-CONNECT: Interface Serial0/2/0:5 is now connected to 7753428028 N/A
560243: Aug 13 20:37:24.544: %ISDN-6-CONNECT: Interface Serial0/2/0:5 is now connected to 7753428028 N/A
560244: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 7753296300 N/A
560245: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 7753296300 N/A
560246: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:22 is now connected to 3296300 N/A
560247: Aug 13 20:37:27.128: %ISDN-6-CONNECT: Interface Serial0/2/0:22 is now connected to 3296300 N/A
560248: Aug 13 20:37:27.652: %ISDN-6-CONNECT: Interface Serial0/2/0:20 is now connected to 3525101 N/A
560249: Aug 13 20:37:27.652: %ISDN-6-CONNECT: Interface Serial0/2/0:20 is now connected to 3525101 N/A
560250: Aug 13 20:37:30.320: %ISDN-6-CONNECT: Interface Serial0/2/1:22 is now connected to 3525101 N/A
560251: Aug 13 20:37:30.320: %ISDN-6-CONNECT: Interface Serial0/2/1:22 is now connected to 3525101 N/A
How do I do that with the 2921 in an SSH session? I've enabled "logging monitor" and "term mon" but I still can't see messages like that.
I thank you very much for your time.Hello,
To check clocking you can use the "show network-clock" command. For example:
SiteA-RTR#show network-clock
Network Clock Configuration
Priority Clock Source Clock State Clock Type
1 T1 0/0/0 GOOD T1
10 Backplane GOOD PLL
Current Primary Clock Source
Priority Clock Source Clock State Clock Type
1 T1 0/0/0 GOOD T1
The above is an example of a "good" configuration, assuming that T1 0/0/0 is the controller terminating the line from your carrier. In your case, it would be T1 0/2/0 or 0/2/1. You can also look at "show controller t1 [interface]" (e.g. show controller t1 0/2/0) and pay particular attention to "Line Code Violations", "Slip Secs", and "Errored Secs" counters. Note that the "Clock Source Line" status in the "show controller t1" command is not enough to verify you have proper clocking.
Example of a "bad" clock scenario:
SiteA-RTR(config-voiceport)#do sh controller t1 0/0/0
T1 0/0/0 is up.
Applique type is Channelized T1
Cablelength is long 0db
No alarms detected.
alarm-trigger is not set
Soaking time: 3, Clearance time: 10
AIS State:Clear LOS State:Clear LOF State:Clear
Version info Firmware: 20080918, FPGA: 13, spm_count = 0
Framing is ESF, Line Code is B8ZS, Clock Source is Line.
CRC Threshold is 320. Reported from firmware is 320.
Data in current interval (483 seconds elapsed):
3 Line Code Violations, 4 Path Code Violations
207 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
206 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 109 Unavail Secs
To make sure the clock is set correctly, look at the "network-clock-participate" and the "network-clock-select" commands.
For the ISDN status information, you can use "show isdn status". For example:
SiteA-RTR(config)#do sh isdn stat
Global ISDN Switchtype = primary-ni
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply
ISDN Serial0/0/0:23 interface
dsl 0, interface ISDN Switchtype = primary-ni
L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x800000FF
Number of L2 Discards = 0, L2 Session ID = 15
*Note: The above is output from a gateway provisioned to use MGCP.
For all gateways, just make sure that Layer 1 is "ACTIVE" and that Layer 2 has "MULTIPLE_FRAME_ESTABLISHED".
You can also use the "show isdn service" command to look at status for individual B-channels.
SiteA-Rtr#sh isdn service
PRI Channel Statistics:
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not appl
ISDN Se0/0/0:23, Channel [1-24]
Configured Isdn Interface (dsl) 0
Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3
Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2
Use the "Channel State" and "Service State" to verify things are what you expect.
HTH.
-Bill (@ucguerrilla)
http://ucguerrilla.com -
How to make CUCM as a TFTP server , then copy files to Voice Gateway ?
how to make CUCM as a TFTP server , then copy files to Voice Gateway ? anyone knows?
Hi,
Please check the following link
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/6_1_1/ccmsys/cucm-accm-611/a02tftp.html#wp1023004
Understanding How Devices Access the TFTP Server
You can enable the IP phones and gateways to discover the TFTP server IP address in one or more of the following ways, depending on the device type:
•Gateways and phones can use DHCP custom option 150.
Cisco recommends this method. With this method, you configure the TFTP server IP address as the option value.
•Gateways and phones can use DHCP option 066.
You may configure either the host name or IP address of the TFTP server as the option value.
•Gateways and phones can query CiscoCM1.
Ensure the Domain Name System (DNS) can resolve this name to the IP address of the TFTP server. Cisco does not recommend this option because it does not scale.
•You can configure phones with the IP address of the TFTP server. If DHCP is enabled on the phone, you can still configure an alternate TFTP server IP address locally on the phone that will override the TFTP address that was obtained through DHCP.
•Gateways and phones also accept the DHCP Optional Server Name (sname) parameter.
•The phone or gateway can use the value of Next-Server in the boot processes (siaddr).
Devices save the TFTP server address in nonvolatile memory. If one of the preceding methods was available at least once, but is not currently available, the device uses the address that is saved in memory.
You can configure the TFTP service on the first node or a subsequent node, but usually you should configure it on the first node. For small systems, the TFTP server can coexist with a Cisco Unified Communications Manager on the same server.
HTH
Manish
Maybe you are looking for
-
Dump for Shopper On Behalf of Field
Hi Srm Gurus, My requirement was to restrict the search help for Shopper on Behalf in SRM. I Implemented the BADI 'BBP_F4_READ_ON_ENTRY' for method 'GET_BOBUSER'. In testing i got the expected result in the Search Help. When i try to select the entry
-
Can I redeem gift card purchase at Australia on China Apple store?
i was wondering if I could use the gift card I buy from Australia by redeem it on China Apple Store ?
-
Acrobat looses windows focus after showing a wxWidgets modal dialog
Here's a code sample: AVWindow dlgWindow = AVWindowNewFromPlatformThing(AVWLmodal, NULL, NULL, gExtensionID, dlg->GetHWND()); AVAppBeginModal(dlgWindow); dlg->ShowModal(); // dlg is a wxDialog* AVWin
-
Third Party Tool-Open Hub Destination
dear all, can any one guide me as to how data(master data in particular) from bw can be transferred to oracle data base using open hub destiantion.is rfc connection between bw and third part tool enough,do we have to maintain the same stru
-
Remove data of specific sales organization
Hi guys, Anyone know how can I remove/delete all data specific to a sales organization/company? Thanks