DSP hardware in Cisco 2911?

I have a Cisco 2911 router that needs to do T1 PRI with 32 voice DSPs.
I think I need UC license and VWIC3-1MFT-T1/E1=. I don't see any onboard DSP in "show inv" or "show diag" so is it correct I need to purchase a PVDM2-32? Should this be installed onboard or does this require a NM-HDV2 to house it?

Hi,
You need one PVDM3-32. You don't need NM.
Regards,
- Adrian.

Similar Messages

Hardware Needed for Cisco 2911 as a console server

Hi,
We need to setup Cisco 2911 router as a console server for OOB (Out of band) connectivity to console of each DC device (upto 20 devices) Could someone please respond to the following questions we had:
1. What interface module can we install on the 2911 ISR for this purpose?
2. What cable (part number please) will go to the that 2911-ISR interface slot and then we can connect the consoel ports of out network devices to that cable.
It needs to be something similar to the cisco octal cable which I know we used for Cisco 2500 series for console purposes to other devices. But not sure about Cisco 2911.
I would highly appreciate your information and help.
Thanks
Lovleen

You would need an interface providing asynchronous serial ports. Something like the HWIC-8A or the HWIC-16A. Needing 20 ports you will most likely use one of each type (having then 24 ports).
The cable to connect to the HWIC is the CAB-HD8-ASYNC. If you attach routers and switches console ports directly to the RJ45 plugs everything is fine. If you have other types of serial ports to serve (DB-9 or DB-25) then you need the according adapters. I don't know if they have product numbers...
Or to have it all on one single PDF follow this link
BR
Björn

Cisco 2911/K9 Block P2P / Torrent

Dear Experts,
We currently have a Cisco 2911/K9 router and I feel there's a chunk of people who are eating up our bandwidth.
I am no expert in this area, but I have searched endlessly and haven't been able to find much information on how to resolve this issue on this router or if it is even possible.
Could someone shed some light onto this? Is it possible with this current router and no extra hardware (maybe extra software) to somehow monitor/block/de-prioritize the P2P/Torrent users?
Thank you!
Kind regards,
PIU

Without any additional license or equipment, that's very hard to achieve.
The best way to solve that problem is to use an ASA-NGFW instead of the IOS-router.
If you have to stick with the router you could use Cisco Web Security (CWS) formaly known as Scansafe. But that needs also an additional license.
With only the router you could try some dirty hacks. For example you can deny all unwanted IPs (that of Facebook, Youtube ...) in an ACL. But that is very hard to manage. Or you could control the DNS-communication in a way that your DNS-server return an internal IP of your own webserver for all the unwanted domains.
But all in all, you are using the wrong tool for that problem.

Cisco 2911 Side mounting hole dimension schematic

Hi,
My mechanical engineer needs detailed measurements for the hole patterns on the sides of the Cisco 2911.
Any ideas where I could find this for him?
I've searched and read the installation docs, which shows pictures of the units. Although helpful, they does not show exact measurements, angles, and hole diameters.
thank you,
Mark

hi Mark,
Here is a 2911 hardware related document ...
http://www.cisco.com/en/US/docs/routers/access/2900/hardware/installation/guide/Install_Connect.html#wp1089950
Hope this helps!
Thanks,
Karthik

Cisco 2911 security slot

Does Cisco 2911 K9 router have a security slot to put a anti-theft cable into?
Thanks.

No they do not appear to have a security slot
http://www.cisco.com/c/en/us/td/docs/routers/access/2900/hardware/installation/guide/Hardware_Installation_Guide/Overview.html#wp1082742
I cant check a couple 2911's in person and make sure for you tomorrow.

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. Smith

i 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

CME B-ACD on Cisco 2911 with IOS 15.2(4)M5 not working

Hi Folks,
I am currently setting up CME version 9.1 with B-ACD (app-b-acd-aa-3.0.0.2.tcl & app-b-acd-3.0.0.2.tcl), running on
Cisco 2911 with IOS ver 15.2(4)M5, this is for lab purposes.
Below is my CME & B-ACD configuration :
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
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
h323
h225 listen-port 1820
no call service stop
sip
bind control source-interface Vlan400
bind media source-interface Vlan400
registrar server expires max 600 min 60
voice register global
mode cme
source-address 172.25.202.1 port 5060
max-dn 2
max-pool 2
load 9971 sip9971.9-2-2SR1-9
authenticate register
timezone 28
time-format 24
date-format D/M/Y
tftp-path flash:
create profile sync 0004714411607756
voice register dn 1
number 3005
name br2phn2
voice register dn 2
number 3006
name br2phn4
voice register template 1
dialplan 1
voice register dialplan 1
type 7940-7960-others
pattern 1 3...
pattern 2 999
voice register pool 1
id mac 1C1D.86C4.0D6D
type 9971
number 1 dn 1
template 1
dtmf-relay rtp-nte
username 3005 password cisco
description 3214-3005
codec g711ulaw
voice register pool 2
id mac 1C1D.86C4.A574
type 9971
number 1 dn 2
template 1
dtmf-relay rtp-nte
username 3006 password cisco
description 3214-3006
codec g711ulaw
voice hunt-group 1 parallel
list 3002,3006
pilot 3210
application
service aa flash:/app-b-acd-aa-3.0.0.2.tcl
paramspace english index 1
param number-of-hunt-grps 2
param handoff-string aa
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 3500
paramspace english location flash://
param second-greeting-time 60
param welcome-prompt _bacd_welcome.au
param call-retry-timer 15
param voice-mail 3001
param max-time-call-retry 90
param service-name queue
service aa-drop flash:/app-b-acd-aa-3.0.0.2.tcl
paramspace english index 1
param service-name queue
param drop-through-option 2
param second-greeting-time 60
paramspace english language en
param max-time-vm-retry 2
param max-time-call-retry 90
param voice-mail 3001
paramspace english location flash://
param aa-pilot 3501
param number-of-hunt-grps 1
param handoff-string aa-drop
param call-retry-timer 15
service queue flash:/app-b-acd-3.0.0.2.tcl
param queue-len 15
param aa-hunt10 3006
param queue-manager-debugs 1
param number-of-hunt-grps 2
param aa-hunt2 3210
interface Loopback0
ip address 172.25.110.3 255.255.255.255
ip ospf network point-to-point
h323-gateway voip interface
h323-gateway voip id Spain ipaddr 172.25.110.1 1719
h323-gateway voip h323-id BR2-RTR
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 172.25.110.3
interface Vlan400
ip address 172.25.202.1 255.255.255.0
ip pim dense-mode
dial-peer voice 3500 voip
service aa
destination-pattern 3500
session target ipv4:172.25.110.3
incoming called-number 3500
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 3501 voip
service aa-drop
destination-pattern 3501
session target ipv4:172.25.110.3
incoming called-number 3501
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
telephony-service
no auto-reg-ephone
max-ephones 2
max-dn 2 no-reg both
ip source-address 172.25.110.3 port 2000
cnf-file location flash:
load 7965 term65.default.loads
time-zone 28
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
moh "music-on-hold.au"
web admin system name admin password cisco
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 14 2014 05:54:44
ephone-template 1
softkeys connected Endcall Hold Park Trnsfer Acct Flash
ephone-dn 1 octo-line
number 3001 no-reg both
description 3214-3001
name br2phn1
ephone-dn 2 octo-line
number 3002 no-reg both
description 3214-3002
name br2phn3
ephone 1
device-security-mode none
mac-address 189C.5DB6.D303
ephone-template 1
max-calls-per-button 5
busy-trigger-per-button 3
type 7965
button 1:1
ephone 2
device-security-mode none
description 3214-3002
mac-address 984B.E194.FDDD
ephone-template 1
max-calls-per-button 5
busy-trigger-per-button 3
type 7960
button 1:2
Problem :
1. When I test call from CME Phone both SIP and SCCP Phone by dial 3500 or 3501, I get the busy tone.
2. Debug voip dial-peer, match with dial-peer voice 3500 for (aa service) & 3501 for (aa-drop service).
3. Debug voice application script, show nothing.
Is there something wrong with my configuration ?
Rgds
Novri

Hi Novriadi,
In your configuration
service aa flash:/app-b-acd-aa-3.0.0.2.tcl
service queue flash:/app-b-acd-3.0.0.2.tcl
paramspace english location flash://
Remove "/" and "//" from the configuration
Then use the call application voice load command in privileged EXEC mode to reload the scripts.
Router# call application voice load aa
Router# call application voice load queue
Router# call application voice load aa-drop
You can refer to following document as well for more info
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/bacd/configuration/guide/cme40tcl/40bacd.html#wp1018270
Please find the sample configuration that is required to configure b-acd in CME for reference.
telephony-service
moh music-on-hold.au
multicast moh 239.1.1.1 port 2000
application
service queue flash:app-b-acd-2.1.0.0.tcl
param number-of-hunt-grps 2
param aa-hunt2 1111
param aa-hunt3 1222
param queue-len 15
param queue-manager-debugs 1
service aa flash:app-b-acd-aa-2.1.0.0.tcl
paramspace english index 1
paramspace english language en
paramspace english location flash:
param service-name queue
param handoff-string aa
param aa-pilot 8005550123
param welcome-prompt _bacd_welcome.au
param number-of-hunt-grps 2
param dial-by-extension-option 1
param second-greeting-time 60
param call-retry-timer 15
param max-time-call-retry 700
param max-time-vm-retry 2
param voice-mail 5003
dial-peer voice 222 voip
service aa
destination-pattern 8005550123
session target ipv4:192.168.1.1
incoming called-number 8005550123
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
Thanks & Regards,
Mudit Mathur

Can't establish a Voice gateway (cisco 2911) using SIP with CUCM 9.1

I have configured a Cisco 2911 as a Voice Gateway using SIP (the configuration is attached), but unfortunately can't establish a test call to a phone (CUPC 8.6 SCCP) using csim start. I have done logging the ccsip debug and ccapi debug and attached them. Could anyone help me to solve this problem?

I just did some research on my end and csim is not supported for SIP. The Invite will never be created and sent to the CUCM to initate the call. It disconnects in the router itself with normal cause.
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPIOutgoingCallSDP:
Could not create source SDP for Outgoing Call
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPICreateOutboundSDP:
Error in creating an SDP for the outbound call - Check for supported codecs
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/preprocessSetup:
Error during outbound SDP creation
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for outgoing call
Please use an actual call to test your dial-peer and integration with call manager. csim will not work.
Hantale
Sree

IP SLA Monitor on Cisco 2911

Dear all,
I have a cisco 2911 router that is located in my head office LAN and I use this router to connect to my branch networks. I want to configure IP SLA Monitor on this router to track my WAN Links but it does not support the command IP SLA Monitor. My IOS VERSION is c2900-universalk9-mz.SPA.151-2.T1.bin. Please help tell me how I can configure IP SLA on my router.
Any assistance will be highly appreciated.

The Data Technology Package License part number SL-29-DATA-K9 was changed to the AppX Technology Package License that includes DATA and WAAS features with part number SL-29-APP-K9.
SL-29-APP-K9 (AppX License for Cisco 2900 Series) - USD 1,000.00
Please check the Change in Product Part Number Announcement for the Cisco 2900 Series Integrated Services Routers Data Technology Package Licenses link below for your reference(s):
http://www.cisco.com/c/en/us/products/collateral/routers/2900-series-integrated-services-routers-isr/eos-eol-notice-c51-730946.html

Cisco 2911 stops responding after a period of time

I have a Cisco 2911 router with 4 T1 connections. Two are set as a multilink and the other two are for two other locations. The router will run fine, but after a month I cannot ping the gigabit ethernet 0/0 interface. I would have to manually reboot the router to get it to respond again. Before I noticed a lot of interface discards which would shutdown the 2911 and a manual reboot would be needed, but for this time it isn't the case. Where would I start with this the memory and cpu usage are fine.
Here is the config:
Current configuration : 2905 bytes
version 15.0
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
hostname 2911
boot-start-marker
boot-end-marker
card type t1 0 0
card type t1 0 1
no aaa new-model
clock timezone gmt -5
clock summer-time cdt recurring
no network-clock-participate wic 0
no network-clock-participate wic 1
no ipv6 cef
ip source-route
ip cef
multilink bundle-name authenticated
license udi pid CISCO2911/K9 sn FTX1513ALLS
controller T1 0/0/0 -- Multilink
cablelength long 0db
channel-group 0 timeslots 1-24
controller T1 0/0/1 -- Multilink
clock source internal
cablelength long 0db
channel-group 0 timeslots 1-24
controller T1 0/1/0
clock source internal
cablelength long 0db
channel-group 3 timeslots 1-24
controller T1 0/1/1
clock source internal
cablelength long 0db
channel-group 2 timeslots 1-24
buffers middle permanent 200
buffers middle max-free 230
buffers middle min-free 50
buffers big permanent 75
buffers big max-free 200
buffers big min-free 15
buffers verybig permanent 20
buffers verybig max-free 20
buffers tune automatic
interface Multilink1
ip address 192.168.200.1 255.255.255.252
ip flow ingress
ip flow egress
load-interval 30
ppp multilink
ppp multilink group 1
ppp multilink fragment disable
no cdp enable
hold-queue 4000 out
interface GigabitEthernet0/0
ip address 10.10.99.1 255.255.255.0
ip flow ingress
ip flow egress
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
interface Serial0/0/0:0
no ip address
encapsulation ppp
ppp multilink
ppp multilink group 1
no fair-queue
hold-queue 40 out
interface Serial0/0/1:0
no ip address
encapsulation ppp
ppp multilink
ppp multilink group 1
no fair-queue
hold-queue 40 out
interface Serial0/1/0:3
ip address 192.168.1.2 255.255.255.0
ip flow ingress
ip flow egress
load-interval 60
no fair-queue
hold-queue 4000 out
interface Serial0/1/1:2
ip address 192.168.8.2 255.255.255.0
ip flow ingress
ip flow egress
load-interval 30
no fair-queue
hold-queue 4000 out
ip forward-protocol nd
no ip http server
no ip http secure-server
ip flow-cache timeout active 1
ip flow-export source GigabitEthernet0/0
ip flow-export version 5
ip flow-export destination 10.10.14.49 2055
ip route 0.0.0.0 0.0.0.0 10.10.99.10
ip route 10.10.17.0 255.255.255.0 192.168.1.1
ip route 10.10.25.0 255.255.255.0 192.168.8.1
ip route 10.10.94.0 255.255.254.0 192.168.200.2
snmp-server community ipBalance RO
snmp-server community SolarWinds RO
control-plane
line con 0
logging synchronous
line aux 0
line vty 0 4
session-timeout 60
privilege level 15
password 7
logging synchronous
login
transport input telnet
scheduler allocate 20000 1000
end

Kishore,
I just hard coded the gigabit 0/0 to 1000 full duplex. The interface errors were occuring on the serial interfaces due to someone doing videoconferencing and trying to use more than 1.5Mbps over the T1. Once they throttled down the video conferencing equipment, the errors seemed to go away.
IOS is
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M4, RELEASE SOFTWARE (fc1)
For syslogs I enabled:
logging trap notifications
logging IP of syslog server
service timestamps debug datetime msec
service timestamps log datetime msec localtime show-timezone year

Cisco 2911 ISR Firewall

Hi everyone,
I would like to inquire on how to deploy Cisco 2911 ISR routers to act as Firewall to protect segments of my network. We have more than 10 units of the said router on our branch and i would like to ask on how i can make it a Firewall, it is running on IOS with sec/k9 license.
Hope that anyone can help me with my problem.
Thank you very much in advance
Best Regards,
Jayson Cruz

Hi Julio,
A good day its me again. My apologies to bother you again. May i ask for your advice regarding the set-up of my IOS Zone-Based Firewall via 2911 routers.
I have 2 2911 beanch routers with bgp peering on a WAN links to reach the branch. On the LAN interface of the said Branch Routers are the LAN segments configured via subinterface command and running HSRP with the other branch router.
How would i implement Zone-Based Firewall with HA without having drops because of asymetric routing. Im sorry since the configuration guide that you have sent me as so many options and configurations that i tend to be confusing on which one is another option and which one is prt of the previous procedure. I hope you could help me with this one as i need to implement it within this week.
Thanks you very much and I'm sorry for bothering you.
Thank you very much!
Jayson
Sent from Cisco Technical Support Android App

Cisco 2911 vesio 12,4 , i have some noise when i make call

hi
1: i have cisco 2911 with 2 card 4FXO i can make the call buth i heath some pertubation in the conversation
2 : i can heat th ring when i make a external call
this my configuration
voice call carrier capacity active
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol cisco
h323
no call service stop
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
voice class h323 1
h225 timeout tcp establish 3
voice-port 0/0/0
supervisory disconnect anytone
cptone FR
connection plar opx 0
caller-id enable
voice-port 0/0/1
supervisory disconnect anytone
no battery-reversal
input gain -3
output attenuation 4
echo-cancel coverage 24
no comfort-noise
cptone FR
timeouts interdigit 6
timeouts call-disconnect 5
timing hookflash-out 500
connection plar opx 0
impedance complex2
caller-id enable
best regard!

Try this:
1 - Use the default configuration for the port. Fw:
voice-port 0/0/0
supervisory disconnect anytone
cptone FR
connection plar opx 0
caller-id enable
voice-port 0/0/1
supervisory disconnect anytone
cptone FR
connection plar opx 0
caller-id enable
2 - Do you hear noise in both ports?
3 - Usually in the case of noise, the problem is not in the router of the beholder. Often the problem is the user of the remote end. The Other End. (Ask for verification on the other side too)
4 - Check the qos is never enough.
I hope I have helped.
Luciane de Medeiros

Cisco 2911 VRRP

Hello
Does Cisco 2911 support VRRP?
I can’t find in datasheet anything about it.
Thanks you in advance.
Br.,
Andrei

Yes, VRRP is supported. You find these infos in the feature-navigator: www.cisco.com/go/fn
Don't stop after you've improved your network! Improve the world by lending money to the working poor:
http://www.kiva.org/invitedby/karsteni

SSH v3 to cisco 2911

Hi,
I've a cisco 2911, IOS software, c2900 software (c2900-Universalk9-M), Version 15.1(4)M4,RELEASE Software (fc1)
How to update ssh v2 on ssh v3?

Hello @nzhanaev01,
SSH 3.0 is just a client version, till now the official protocols are SSHv1 and SSHv2. Refer to the below link.
- http://www.vandyke.com/products/vshell/faq/015.html
Also for further information about SSH you can consult into the RFC4253
- https://tools.ietf.org/html/rfc4253

Cisco 2911 SFP Question

Hello,
Have what I hope is a simple question for you. How best to connect my 250 Mbit/s internet circuit to a newly arrived Cisco 2911? We're changing over our internet circuit from being managed by AT&T (includes router) to an unmanaged one (we provide router). The existing demarc extension terminates in a mult-mode LC connector and connects to a SFP module in the managed AT&T router.
Was sent a 2911 with no additional modules or cards. I have some GLC-SX-MM= SFP modules, but did not see any SFP slots in the 2911. Do I have to go with a higher router, like the 2921? Or are there SFP carrier cards or other modules we can use from Cisco? Thanks.
Scott

For the ISR G2, like the 2900, you should be looking for the EHWIC-1GE-SFP-CU.
How best to connect my 250 Mbit/s internet circuit to a newly arrived Cisco 2911?
However, the main question is this: Can a 2911 handle 250 Mbps bandwidth? The answer is no. The 2911 can handle 180.73 Mbps of traffic. This value is express in HALF duplex and without any encryption.
I believe a 3925E can support 250 Mbps of traffic (full dupex, full encryption).

DSP hardware in Cisco 2911?

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