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- About mohammed asif suleman
08-17-2018
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Member since
12-18-2012
333
Posts
20
Helpful
2
Solutions
08-01-2016
11:14 PM
Hi,
i am trying to configure MAC Filtering for few PCs which should have access to internet and remaining all should be blocked.
we have a remote branch connected over dmvpn tunnel and internet traffic going out from remote gateway.
i have to block all users from accessing internet except few but allow all to access few servers over dmvpn tunnel.
interface FastEthernet4 ip address dhcp ip nat outside ip virtual-reassembly in duplex auto speed auto ! interface Vlan1 description "To LAN" ip address 192.168.15.161 255.255.255.224 ip nat inside ip virtual-reassembly in ip tcp adjust-mss 1452
! ip nat inside source list 2000 interface FastEthernet4 overload ip route 48.96.27.138 255.255.255.255 192.168.1.1 ! ! access-list 2000 deny ip any 192.168.20.4 255.255.255.255 access-list 2000 deny ip any 192.168.20.30 255.255.255.255 access-list 2000 deny ip any 192.168.20.54 255.255.255.255 access-list 2000 permit ip any any
traffic for 3 servers going on dmvpn tunnel and the remaining traffic going out to internet.
i want to block all PC's access to internet except few and all users must access these 3 servers.
... View more
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08-01-2016
12:32 AM
Hi,
i am trying to configure MAC Filtering for few PCs which should have access to internet and remaining all should be blocked.
we have a remote branch connected over dmvpn tunnel and internet traffic going out from remote gateway.
i have to block all users from accessing internet except few but allow all to access few servers over dmvpn tunnel.
interface FastEthernet4 ip address dhcp ip nat outside ip virtual-reassembly in duplex auto speed auto ! interface Vlan1 description "To LAN" ip address 192.168.15.161 255.255.255.224 ip nat inside ip virtual-reassembly in ip tcp adjust-mss 1452
! ip nat inside source list 2000 interface FastEthernet4 overload ip route 48.96.27.138 255.255.255.255 192.168.1.1 ! ! access-list 2000 deny ip any 192.168.20.4 255.255.255.255 access-list 2000 deny ip any 192.168.20.30 255.255.255.255 access-list 2000 deny ip any 192.168.20.54 255.255.255.255 access-list 2000 permit ip any any
traffic for 3 servers going on dmvpn tunnel and the remaining traffic going out to internet.
i want to block all PC's access to internet except few and all users must access these 3 servers.
... View more
Labels:
Created by
mohammed asif suleman
in IP Telephony and Phones
in IP Telephony and Phones
01-06-2016
11:50 PM
01-06-2016
11:50 PM
you both are correct.
the translation was not happening because when configuring i missed 0 in the translation pattern, thats why the extensions ending with 0 are giving the problems.
rule 2 /^30\(.[12345679]\)/ /8062301/
rule 2 /^30\(.[ 0 12345679]\)/ /8062301/
now every thing working fine
thank you all
... View more
01-06-2016
11:41 PM
HI,
thank you for your replies,
its taking the wrong dial-peer,it should take dial-peer 20.
attached is the working call from another extension..
... View more
Created by
mohammed asif suleman
in IP Telephony and Phones
in IP Telephony and Phones
01-06-2016
11:09 PM
01-06-2016
11:09 PM
Hi,
we have cucm 8.6 and cisco gateway 2811 with sip trunk to istp. these are working fine except one extension having outbound calls issue.
i am getting 484 address incomplete msg in ccsip messages.
call flow is like
IP Phone-----CUCM----(Sip Trunk)---Gateway--(SIP Trunk)-----ITSP.
Calling Number 3030
Called number 0558902195
-----------------------------------------------------------------------------------------------
ASICO-DAM#debug ccsip messages 2734609: *Jan 6 12:39:16.305: %ENVMON-4-FAN_LOW_RPM: Fan 1 service recommended 2734610: *Jan 6 12:39:16.305: %ENVMON-4-FAN_LOW_RPM: Fan 2 service recommended SIP Call messages tracing is enabled ASICO-DAM# 2734611: *Jan 6 15:39:31.177: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: OPTIONS sip:172.29.25.246:5060 SIP/2.0 Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKsbcthLNQZtAxpHZnAHBxnAnBQt4nL4n xBT10033 Call-ID: sbcthLNQpBZmLp4mVtxBcBVncqcQc4LntcHAtH4m@52p1ULIII From: <sip:172.29.25.246:5060>;tag=sbc0806sbcthLNQV4HmHaxx To: <sip:172.29.25.246> CSeq: 1 OPTIONS Max-Forwards: 70 Content-Length: 0
2734612: *Jan 6 15:39:31.189: //3028716/5DF46AF98D4C/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKsbcthLNQZtAxpHZnAHBxnAnBQt4nL4n xBT10033 From: <sip:172.29.25.246:5060>;tag=sbc0806sbcthLNQV4HmHaxx To: <sip:172.29.25.246>;tag=1720EDC4-B8F Date: Wed, 06 Jan 2016 12:39:31 GMT Call-ID: sbcthLNQpBZmLp4mVtxBcBVncqcQc4LntcHAtH4m@52p1ULIII Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 OPTIONS Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER Allow-Events: telephone-event Accept: application/sdp Supported: timer,resource-priority,replaces,sdp-anat Content-Type: application/sdp Content-Length: 170
v=0 o=CiscoSystemsSIP-GW-UserAgent 9786 1860 IN IP4 172.29.25.246 s=SIP Call c=IN IP4 192.168.33.5 t=0 0 m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3 c=IN IP4 192.168.33.5
2734613: *Jan 6 15:39:31.725: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:0558902195@192.168.33.5:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK43ad5b394463 From: "Irfan" <sip:3030@192.168.12.190>;tag=83944~70e9433b-1d79-44ae-9a16-09a52b e377c5-19528919 To: <sip:0558902195@192.168.33.5> Date: Wed, 06 Jan 2016 12:32:09 GMT Call-ID: 7f96b000-68d10949-2775-be0ca8c0@192.168.12.190 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 Supported: X-cisco-srtp-fallback Supported: Geolocation Cisco-Guid: 2140581888-0000065536-0000002438-3188500672 Session-Expires: 1800 P-Asserted-Identity: "Irfan" <sip:3030@192.168.12.190> Remote-Party-ID: "Irfan" <sip:3030@192.168.12.190>;party=calling;screen=yes;priv acy=off Contact: <sip:3030@192.168.12.190:5060;transport=tcp> Max-Forwards: 70 Content-Type: application/sdp Content-Length: 217
v=0 o=CiscoSystemsCCM-SIP 83944 1 IN IP4 192.168.12.190 s=SIP Call c=IN IP4 192.168.12.190 t=0 0 m=audio 31512 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2734614: *Jan 6 15:39:31.769: //3028718/7F96B0000000/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:0558902195@10.205.20.50:5060 SIP/2.0 Via: SIP/2.0/UDP 172.29.25.246:5060;branch=z9hG4bK1E77E162C Remote-Party-ID: "Irfan" <sip:3030@172.29.25.246>;party=calling;screen=yes;priva cy=off From: "Irfan" <sip:3030@172.29.25.246>;tag=1720F008-209F To: <sip:0558902195@10.205.20.50> Date: Wed, 06 Jan 2016 12:39:31 GMT Call-ID: 5E4CEA1F-B3A911E5-8D52B1FE-1BC3CC1A@172.29.25.246 Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2140581888-0000065536-0000002438-3188500672 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1452083971 Contact: <sip:3030@172.29.25.246:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 69 Session-Expires: 1800 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 250
v=0 o=CiscoSystemsSIP-GW-UserAgent 6066 4519 IN IP4 172.29.25.246 s=SIP Call c=IN IP4 172.29.25.246 t=0 0 m=audio 16492 RTP/AVP 8 101 c=IN IP4 172.29.25.246 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
2734615: *Jan 6 15:39:31.769: //3028717/7F96B0000000/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.12.190:5060;branch=z9hG4bK43ad5b394463 From: "Irfan" <sip:3030@192.168.12.190>;tag=83944~70e9433b-1d79-44ae-9a16-09a52b e377c5-19528919 To: <sip:0558902195@192.168.33.5> Date: Wed, 06 Jan 2016 12:39:31 GMT Call-ID: 7f96b000-68d10949-2775-be0ca8c0@192.168.12.190 CSeq: 101 INVITE Allow-Events: kpml, telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0
2734616: *Jan 6 15:39:31.781: //3028718/7F96B0000000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.29.25.246:5060;branch=z9hG4bK1E77E162C Call-ID: 5E4CEA1F-B3A911E5-8D52B1FE-1BC3CC1A@172.29.25.246 From: "Irfan"<sip:3030@172.29.25.246>;tag=1720F008-209F To: <sip:0558902195@10.205.20.50> CSeq: 101 INVITE Content-Length: 0
2734617: *Jan 6 15:39:31.841: //3028718/7F96B0000000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 172.29.25.246:5060;branch=z9hG4bK1E77E162C Record-Route: <sip:10.205.20.50:5060;transport=udp;lr> Call-ID: 5E4CEA1F-B3A911E5-8D52B1FE-1BC3CC1A@172.29.25.246 From: "Irfan"<sip:3030@172.29.25.246>;tag=1720F008-209F To: <sip:0558902195@10.205.20.50>;tag=sbc0802sbcthLNQ4BAmnLLt CSeq: 101 INVITE Reason: Q.850;cause=28;text="address incomplete" Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR" Content-Length: 0
2734618: *Jan 6 15:39:31.865: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:0558902195@10.205.20.50:5060 SIP/2.0 Via: SIP/2.0/UDP 172.29.25.246:5060;branch=z9hG4bK1E77E162C From: "Irfan" <sip:3030@172.29.25.246>;tag=1720F008-209F To: <sip:0558902195@10.205.20.50>;tag=sbc0802sbcthLNQ4BAmnLLt Date: Wed, 06 Jan 2016 12:39:31 GMT Call-ID: 5E4CEA1F-B3A911E5-8D52B1FE-1BC3CC1A@172.29.25.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0
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Labels:
Created by
mohammed asif suleman
in IP Telephony and Phones
in IP Telephony and Phones
11-26-2015
02:38 AM
11-26-2015
02:38 AM
its not always but most of the time, regardless which number called..
will upload the debugs in few mins
... View more
11-26-2015
02:27 AM
no,
the message getting played by my cucm, not the called number voice mail.
actually the call should be disconnected with disconnect tone if not answered but if the outbound call not answered the call is getting disconnecs with the message being played by my system as if he dialed wrong number which doesn't match any route pattern...
... View more
11-26-2015
01:27 AM
Hi Rajan,
The Outbound calls made from IP phone to PSTN...
sorry the regarding cofusion over call flow,the call flow is in reverse, i just wrote it to get the understanding of setup
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Created by
mohammed asif suleman
in IP Telephony and Phones
in IP Telephony and Phones
11-26-2015
01:17 AM
11-26-2015
01:17 AM
Hi,
we have cucm 8.6 with the below call flow
PSTN-------sip-------------Gateway--------sip---------CUCM-----sip-----IP PHONE.
when user dials any outbound call , if the call is not answered the user getting the msg like " your call cannot be completed as dialed pleasae consult your directory.........."
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Created by
mohammed asif suleman
in IP Telephony and Phones
in IP Telephony and Phones
11-04-2015
12:19 AM
11-04-2015
12:19 AM
Hi,
we have fax machines configured on ATA with CUCM,starting this week the Inbound faxes are failing.
i am getting error 484 address incomplete in csip messages.
nothing was changed on gateway or cucm.
attaching logs....
Calling number : 0582754624
Called Number : 2946392
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Labels:
Created by
mohammed asif suleman
in IP Telephony and Phones
in IP Telephony and Phones
04-16-2015
01:21 AM
04-16-2015
01:21 AM
Thank you for the replies, the issue got resolved with the help of TAC. actually we have h.323 and sip gateway configured on CUCM and the outbound calls were using H.323 as it was pointed to cube in RL,RG. now we changed it to sip trunk and the calls with sdp are successful.
... View more
Created by
mohammed asif suleman
in IP Telephony and Phones
in IP Telephony and Phones
04-14-2015
11:06 PM
04-14-2015
11:06 PM
Thank you for the replies, i was not able to test as my working hours were over, attaching the debugs.. i cannot see the SDP in the invite ,i don't know why? i have early-media disabled 180 in sip-ua on gateway, if i remove that there will be no ring back tone for all calls.
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Created by
mohammed asif suleman
in IP Telephony and Phones
in IP Telephony and Phones
04-14-2015
05:55 AM
04-14-2015
05:55 AM
hi, we have cucm8.6 and gateway 2811 with ios12.4T everything was working fine till last week then users started complaining about call drops to certain numbers. when i checked on gateway i got q.850 cause 127 error in debug ccsip messages. i called ITSP and those guys told me there is no early offer in SDP. i configured the early offer support for voice and video calls(insert MTP if needed) but still i am not able to call those numbers. and ITSP saying they are not receiving early offer. i tried to force that on gateway but the command is not available in voice service voip or dial-peer. please can anyone help me
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Labels:
Created by
mohammed asif suleman
in Unified Communications Infrastructure
07-25-2017
07-25-2017
yes TAC is the last hope, but our UCS and cucm contract expired and
company in no mood to renew as w...
Created by
mohammed asif suleman
in Unified Communications Infrastructure
07-25-2017
07-25-2017
Today i restarted the UCS server several times, now it is stuck in
Hardware testing only, but i can ...
Created by
mohammed asif suleman
in Unified Communications Infrastructure
07-25-2017
07-25-2017
i did network management test from local KVM consol not from VMware
host. it just tested the default...
Created by
mohammed asif suleman
in Unified Communications Infrastructure
07-25-2017
07-25-2017
Yes the test works, UCS, Host VM and CUCM all are on same subnet, i can
reach UCS ip only. host vm w...
Created by
mohammed asif suleman
in Unified Communications Infrastructure
07-25-2017
07-25-2017
Yes i cannot access VMware Management Interface, i cannot access the
host from vsphere client. It is...
07-24-2017
Hi Brandon, Thank you very much for the reply, i tested the management
network and its working fine ...
Created by
mohammed asif suleman
in Unified Communications Infrastructure
07-24-2017
07-24-2017
Hi, we have cucm 8.6 cluster hosted on two UCS C-200 M2 servers, until
last month it was working fin...
08-01-2016
Hi, i am trying to configure MAC Filtering for few PCs which should have
access to internet and rema...
08-01-2016
Hi, i am trying to configure MAC Filtering for few PCs which should have
access to internet and rema...
01-06-2016
you both are correct. the translation was not happening because when
configuring i missed 0 in the t...
01-06-2016
HI, thank you for your replies, its taking the wrong dial-peer,it should
take dial-peer 20. attached...
Created by
mohammed asif suleman
in IP Telephony and Phones
01-06-2016
01-06-2016
Hi, we have cucm 8.6 and cisco gateway 2811 with sip trunk to istp.
these are working fine except on...
Public Statistics
| Date Registered | 12-18-2012 01:13 AM |
| Date Last Visited | 08-17-2018 12:04 AM |
| Total Messages Posted | 333 |
| Total Helpful Votes Received | 20 |
Helpful Votes Given To
| User | Helpful Count |
|---|---|
| 5 | |
| 5 | |
| 209 | |
| 10 | |
| 3 |
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