ISR as CUBE and Voice Gateway
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
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Yes You can...
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Can I have Cube and Vxml Gateway configured on same Router box ?
Hi
I have one router Model #3900 Series Can I have Cube and Vxml Gateway configured on same router ? If yes, Can any one help me with sample configuration for same.
Thanks and Regards
Anil Kumar.Here is an example of CVP hardware/software compatibility for CVP 10.0, similar doc exists for each version of CVP.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/customer_voice_portal/cvp10_0/reference/guide/CCVP_BK_1ECCDBC7_00_1001-bom/CCVP_BK_1ECCDBC7_00_1001-bom_chapter_00.html -
I have a H.323 gateway. I want to know if I can configure this existing h.323 gateway as a Cube to transport SIP.
If so do I need to bind a seperate interface to SIP and H.323 or the same interface can be used for both
Can the Gateway be registered as h.323 to CCM1 and Can I have a SIP trunk between the same Gateway to CCM2?
Thanks in advanceHi,
"I have a H.323 gateway. I want to know if I can configure this existing h.323 gateway as a Cube to transport SIP".
A CUBE gateway by definition is a voice gateway that connects ip-2-ip calls and plus some extra fancy features. The gateway can support both h.323 and sip at the same time.
"If so do I need to bind a separate interface to SIP and H.323 or the same interface can be used for both"
This is tricky question, the answer is I don't know but why limit yourself when you can bind them into different loopback interfaces.
"Can the Gateway be registered as h.323 to CCM1 and Can I have a SIP trunk between the same Gateway to CCM2?"
Absolutely. -
Site to Site Connectivity Between BE6K and Voice Gateway 2901
Greetings,
Is an Ethernet handoff required for site to site connectivity between BE6K and a voice gateway 2901. My vendor is suggesting that it's required in order for both sites to see the BE6K as one phone system. However, here in lies the problem. I have a point-to-point T-1 between the sites that does not have an Ethernet handoff, just the smartjack to the T-1.
What would I need to get this to work? Have a router at each site? If so, which model? Or is there a component I could add to the BE6K or voice gateway?
Any help would be greatly appreciated.
Thanks in advance.Ethernet handoff just means that the provider will deliver the circuit using Ethernet. What circuit is your provider delivering for you? Do they manage your WAN? Like I said what you need is ip connectivity between your sites and BE6k. If your T1 connection provides WAN connectivity and you have ip connectivity between the sites, then I don't know what you need any handoff for. The question is do you have ip connectivity between the sites via your T1 connection
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ASA 5540 8.4 and Voice Gateway (SIP)
have ASA FW with 8.4(2) : inside is connected to lan , DMZ is having SIP Gateway router , i have configured the Firewall will required ports and enabled sip inspection under policy-map but no VOICE traffic is passing through the firewall , iam using static nat
there are ports opened from both side ASA (dmz-inside and other way)
object-group service TCP tcp
port-object eq sip
port-object range 2000 2443
port-object eq h323
port-object eq rtsp
port-object eq 5061
port-object eq 50693
port-object eq 16341
object-group service UDP udp
port-object eq sip
port-object range 16384 32767
port-object eq tftp
port-object eq ntp
port-object eq 1718
port-object eq 1719
port-object eq 5061
port-object eq 57280
Following is inspection
policy-map global_policy
class inspection_default
inspect ftp
inspect h323 h225
inspect h323 ras
inspect rsh
inspect rtsp
inspect skinny
inspect sunrpc
inspect xdmcp
inspect sip
inspect netbios
inspect tftp
inspect ip-options
inspect dns
service-policy global_policy global
phones are not getting registered .....how about the NAT it should be static between DMZ to inside or No NAT statement
any sugesstions would be helpfullAlthough it is a document about ASA on another version it explains how SIP inspection works on the ASA. For you to understand this document you need to understand the device that you are configuring behind the ASA and how SIP works in general.
http://www.cisco.com/en/US/products/ps6120/products_configuration_example09186a008081042c.shtml
Send a copy of the complete configuration and show service-policy -
How to two DID different bloks work in callmgr and voice gateway
I have trouble to install two different did blocks old did block(6828 7100 to 6828 7699 have been working for two years. Add new did block 6829 3000 to 6829 3199 recently. The PTT gateway 3745 router one ip address in Call mgr gateway has two isdn pri lines. PTT indicated now ISDN pri line 1 is for old did block and pri line 2 is for new did block. However, the gateway router selects the two lines i.e. s1/0:15 & s2/0:25 randomly. Thus, the caller shows the prime number if the call throws to the wrong line. How to fix the call throws to the right ISDN pri line.
Comments from PTT
Can the CallManager throw the calls to the respective lines? For example
for 6828 7100 -7699 it should throw to tp1.01 jgp1348 uam1-01 (ET 1081)
and for 6829 3000 3199 to tp1.02 jgp1348 uam1-01. I am sure this can be
done right? Otherwise the other range (6823 7000 - 7099) would not work
also. If you can ask your vendor on thisPls ignore my problem, i will post in IP telephony.
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Fax and Voice use same DID - SIP/CUBE
Is there any good way to have inbound Fax and Voice calls use the same DID?
I know there used to be a Fax Onramp TCL script, but I thought that was for TDM circuits (T1-PRI).
In this scenario, the PSTN comes from a SIP provider to a CUBE router. The provider does support T.38.
Call flow is: Remote Fax machine -> PSTN -> SIP Provider -> CUBE ->SIP -> CUCM (or Fax server.)
Ideally, I would like to send Fax calls directly to a 3rd party Fax server without going through CallManager (using dial-peers).
I'm hoping there is some secret SDP message that says "This is a Fax call, route me to the Fax server".
Thanks, RandyHi Stephan,
there are several timeouts and timings on voice ports which you can try out to "speed up" your disconnect but from my experience fax machines have an internal timing they need to get ready again - for example it needs often ~30sec to make a second ingoing fax after receiving the first one. For outgoing faxes the machine has usually a buffer where it stores the pages till the line is ready again. So it depends on the type & brand of fax machine you use, doesn't it?
Notice that some timeouts don't work since you control them with UCM (eg. ringing timeout => should be no answer config on line configuration)
show voice port 2/0:
Initial Time Out is set to 2 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Supervisory Disconnect Time Out is set to 750 ms
Ringing Time Out is set to infinity
Wait Release Time Out is set to 30 s
Hookflash-in Timing is set to max=1000 ms, min=150 ms
Hookflash-out Timing is set to 400 ms
Furthermore I would suggest you to use actual 12.4(24)T release since i hade several problems with the 12.4(25c).
Regards,
Christoph -
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.
-
Voice gateways, SLT and PGW
Hi everyone,
I dont know if its the right place to start this discussion so forgive me if i am wrong.( I also have opened a discussion in IP telephony portion)
The company in which i am currently wokring is running a call center and IPT as well. i will try to explain the scenario
voicegateways (AS5400 and AS5350 routers with IOS Version 12.4(15)T7)
SLT (2651XM with IOS Version 12.2(8)T10)
PGW (PGW 2200 with SunOS 5.10, MGC)
I was under the impression that if you want to connect PSTN and a voip network you need a PSTN gateway. So why are we using 3 different types of hardware.
Follwing are the explainations in a doucment given to me
the Cisco PGW 2200 provides service providers with the capability to seamlessly route voice and data calls between the PSTN and New World packet networks.
Cisco 2611 Signaling Link Terminals
(E1 terminates, Take Signaling part and send B n D channels to Voice gateway, Signaling part is resolved by it self and PGW)
Voice Gateways
allows terminals of one type, such as H.323, to communicate with terminals of another type, such as a PBX, by converting protocols. Gateways connect an organization’s network to the PSTN
Any information is much appreciatedhave you read this
End-of-Sale and End-of-Life Announcement for the Cisco PGW 2200 Softswitch and Software
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2027/end_of_life_notice_c51-676990.html
hope this help -
Analog interface between voice gateway and pbx system, please suggest too
Hi
Im a fresh in telephony and voip technology. I didnt find information about design analog interface between voice gateway and pbx system such as fxo-to-fxo, fxo-to-fxs, fxs-to-fxs, fxo-to-e&m, fxs-to-e&m, or e&m-to-e&m. Could you please suggest any resource for me?
Ps. my networks have 1 HQ and 2 branch offices connect to the HQ.
Thanks,
NitassHello,
you can find lots of info about analog voice interfaces here:
http://www.cisco.com/en/US/tech/tk652/tk653/tk754/tech_protocol_home.html
I've found this particular document very useful when trying to understand how telephone signaling works. This applies to any telephone system where interfacing to analog phones or the PSTN is involved.
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800a6210.shtml
Regards. -
ILBC calls via SIP Trunk with CUBE and CUCM7
hi there,
our SIP Provider offers the IBLC codec which promises to provide better quality compard to G.729.
I'm using this scenario:
IP-Phone(G711) --- CUCM7 --- (SIP-Trunk1) --- CUBE --- (SIP-Trunk2) --- Provider
Everything workes unless I'm configuring IBLC at the provider and on trunk2.
I have the CUBE router acting as a trancoding device and also specified IBLC as codec to be handled.
SIP trunk 2 was placed in a region with IBLC as codec.
On the trunk configuration in CUCM the media ressource group with XCODE capability is configured
Transcoding workes between two IP Phones in different regions with different codecs within the intranet.
Unfortunately the CUBE router doesn't seem to use the transcoder to change internal G711u calls into IBLC codec
so calls are blocked by the CUBE device:
deb ccsip calls
for incoming call:
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4AE7AC98
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0237892992
Called Number : 036677725231
Source IP Address (Sig ): 10.100.100.50
Destn SIP Req Addr:Port : <IP SIP Provicer>
Destn SIP Resp Addr:Port : <IP SIP Provicer>:5060
Destination Name : <IP SIP Provicer>
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : ilbc
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): <IP CUBE>
Source IP Port (Media): 0
Destn IP Address (Media): <IP SIP Provicer>
Destn IP Port (Media): 22022
Orig Destn IP Address:Port (Media): [ - ]:0
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 488
(Output lookes similar to outgoing calls)
I set up ccm on cube and assigned dsp ressources without success:
Here are the relevant configuration parts:
voice class codec 1
codec preference 1 iblc
voice service voip
address-hiding
allow-connections sip to sip
allow-connections h323 to sip
allow-connections sip to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
h323
sip
header-passing error-passthru
no update-callerid
midcall-signaling passthru
privacy-policy passthru
voice-card 0
dspfarm
dsp services dspfarm
dial-peer voice 40991 voip
description *** Incoming from SIP-Provider
destination-pattern 03667772523.%
session protocol sipv2
session target ipv4:<IP_of_CUCM>
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
ip qos dscp cs5 media
ip qos dscp cs5 signaling
sccp local GigabitEthernet0/0
sccp ccm 10.100.100.50 identifier 11 version 4.1
sccp
sccp ccm group 11
description *** lokaler CCM fuer Codec-Konvertierung von SIP/DUS.NET
associate ccm 11 priority 1
associate profile 21 register DE_WGT_MTP02
dspfarm profile 21 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec ilbc
maximum sessions 10
associate application SCCP
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 10
sdspfarm tag 1 DE_WGT_MTP02
max-ephones 30
max-dn 30
ip source-address 10.100.100.50 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 14 2010 02:10:34
sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.100.100.50
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.100.100.50, Port Number: 2000
Priority: N/A, Version: 4.1, Identifier: 11
Trustpoint: N/A
Call Manager: 10.1.1.55, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 10
Trustpoint: N/A
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.100.100.50, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 21
Reported Max Streams: 20, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
sh dspfarm dsp all
SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED
0 1 26.3.4 UP N/A FREE xcode 1 - - -
0 1 26.3.4 UP N/A FREE xcode 1 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 1 - - -
1 1 26.3.4 UP N/A FREE xcode 1 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
Thanks in advance,
DavidHi there,
Just wondering whether you ever got this resolved? I seem to have a very similiar problem.
Regards
Karen -
Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP
Hi Cisco Community,
I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
Below is an example of a call that is connected with the current setup:
Note:
IP: 10.18.81.2 (CUBE)
IP: 10.18.81.11 (CUCM SUB)
IP: 10.111.111.254 (ITSP SBC)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
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,X-cisco-original-called
Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1417347869
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
s=SIP Call
c=I
PM-HO-VG-01#N IP4 10.18.81.2
t=0 0
m=audio 22256 RTP/AVP 18 0 8 101
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 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
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf9
PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,application/xml
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 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
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 236
v=0
o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.80.40
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
PM-HO-VG-01#
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
PM-HO-VG-01#sh sip
PM-HO-VG-01#sh sip-ua call
PM-HO-VG-01#sh sip-ua calls
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 27218091323
Called Number : 0862000000
Bit Flags : 0xC04018 0x10000100 0x0
CC Call ID : 64511
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.111.111.254]:5060
Destn SIP Resp Addr:Port: [10.111.111.254]:5060
Destination Name : 10.111.111.254
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64511
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22256
Media Dest IP Addr:Port : [10.111.111.254]:20074
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 0218091323
Called Number : 0862000000
Bit Flags : 0xC0401E 0x10000100 0x80004
CC Call ID : 64510
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.18.81.11]:5060
Destn SIP Resp Addr:Port: [10.18.81.11]:5060
Destination Name : 10.18.81.11
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64510
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22350
Media Dest IP Addr:Port : [10.18.80.40]:21928
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
PM-HO-VG-01#
PM-HO-VG-01#
PM-HO-VG-01#
As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22256 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 102 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 360
v=0
o=BroadWorks 316169737 2 IN IP4 10.111.111.254
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=inactive
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22350 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Length: 0
Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 103 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 306
v=0
o=BroadWorks 316169737 3 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 2
PM-HO-VG-01#00 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 213
v=0
o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.81.10
t=0 0
m=audio 4000 RTP/AVP 18
a=X-cisco-media:umoh
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=sendonly
Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 104 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Race Condition
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
Timestamp: 1417347889
CSeq: 104 BYE
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 101 BYE
Content-Length: 0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 200
PM-HO-VG-01#Hi Manish,
Again, excellent feedback. Much appreciated.
I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
I will be doing some intensive test again later on this week and will send the logs.
Here is my question to both of you:
Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
Thanks again for your support fellows. -
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 -
SG-300 28P switches problem with VLAN Data and Voice, working all the time as Voice VLAN
Hi Everyone,
Thank you very much for your help in advance. I’m pulling my hair to fix the problem.
I just got the new SG-300 28P switches. My Bios ordered for me. I did not know how it runs until now... not an IOS based. I really do not know how to configure it.
I have 2 VLAN are Data and Voice.
- Data VLAN ID is 2 IP 192.168.2.X/255.255.255.0
- Voice VLAN ID is 200 IP 192.168.22.X/255.255.255.0
- I created two vlans, in switch, Data and Voice.
- On the port number 28, it is trunk by default, so I add Data vlan ID 2 tagged.
- On the port number 26, it is trunk by default, so I add Voice vlan ID 200 tagged.
- On the port number 27, I add Data vlan ID 2 tagged for Data vlan out.
- Port settings No.1
I set it up as Trunk with Data vlan 2 untagged, and 200 Tagged (voice vlan). I plugged in a phone with a pc attached. But the PC will get to the vlan 200 to get the DHCP address, but no from vlan 2. The Phone works with correct vlan ip.
- Port settings No.2
Trunk with vlan 1UP, 2T, and 200T. The phone is even worse. Would never pick up any IP from DHCP.
- Port settings No.3
Access with 200U...of course the phone will work... and the PC could not get to its own vlan. Instead, the PC got an ip from the voice vlan. Not from VLAN 2.
I have Linksys phone I’m not sure if this help.
For more information I setup in switch,
- enable voice vlan
- set the port on auto voice vlan
- enable LLDP-MED globally
- create a network policy to assign VLAN 200
- assign this network policy to the port the phone is connected to.
I hope this information help to help me to setup Data and Voice vlans, to plug the phone to work with vlan Voice 200 (IP rang 192.168.22.X), from phone to Pc and pc work as Data vlan 2 (IP rang 192.168.2.X).I just got done setting up voice VLANs on an SF 300-24P and verified working. This was working with Cisco 7900 series phones connected to a Cisco UC setup.
Here's my sample config.
Note that I edited this by hand before posting, so doing a flat out tftp restore probably won't work. However, this should give you a clue. Also, don't take this as 100% accurate or correct. I've only been working with these things for about a week, though I've worked with the older Linksys SRW switches for a couple of years. I'm a CCNP/CCDP.
VLAN 199 is my management VLAN and is the native VLAN on 802.1q trunks.
VLAN 149 is the data/computer VLAN here.
VLAN 111 is the voice/phone VLAN here.
VLAN 107 does nothing.
interface range ethernet e(1-24)
port storm-control broadcast enable
exit
interface ethernet e1
port storm-control include-multicast
exit
interface ethernet e2
port storm-control include-multicast
exit
interface ethernet e3
port storm-control include-multicast
exit
interface ethernet e4
port storm-control include-multicast
exit
interface ethernet e5
port storm-control include-multicast
exit
interface ethernet e6
port storm-control include-multicast
exit
interface ethernet e7
port storm-control include-multicast
exit
interface ethernet e8
port storm-control include-multicast
exit
interface ethernet e9
port storm-control include-multicast
exit
interface ethernet e10
port storm-control include-multicast
exit
interface ethernet e11
port storm-control include-multicast
exit
interface ethernet e12
port storm-control include-multicast
exit
interface ethernet e13
port storm-control include-multicast
exit
interface ethernet e14
port storm-control include-multicast
exit
interface ethernet e15
port storm-control include-multicast
exit
interface ethernet e16
port storm-control include-multicast
exit
interface ethernet e17
port storm-control include-multicast
exit
interface ethernet e18
port storm-control include-multicast
exit
interface ethernet e19
port storm-control include-multicast
exit
interface ethernet e20
port storm-control include-multicast
exit
interface ethernet e21
port storm-control include-multicast
exit
interface ethernet e22
port storm-control include-multicast
exit
interface ethernet e23
port storm-control include-multicast
exit
interface ethernet e24
port storm-control include-multicast
exit
interface range ethernet g(1-4)
description "Uplink trunk"
exit
interface range ethernet g(1-4)
switchport default-vlan tagged
exit
interface range ethernet e(21-24)
switchport mode access
exit
vlan database
vlan 107,111,149,199
exit
interface range ethernet g(1-4)
switchport trunk allowed vlan add 107
exit
interface range ethernet e(21-24)
switchport access vlan 111
exit
interface range ethernet g(1-4)
switchport trunk allowed vlan add 111
exit
interface range ethernet e(1-20)
switchport trunk native vlan 149
exit
interface range ethernet g(1-4)
switchport trunk allowed vlan add 149
exit
interface range ethernet g(1-4)
switchport trunk native vlan 199
exit
voice vlan aging-timeout 5
voice vlan oui-table add 0001e3 Siemens_AG_phone________
voice vlan oui-table add 00036b Cisco_phone_____________
voice vlan oui-table add 00096e Avaya___________________
voice vlan oui-table add 000fe2 H3C_Aolynk______________
voice vlan oui-table add 0060b9 Philips_and_NEC_AG_phone
voice vlan oui-table add 00d01e Pingtel_phone___________
voice vlan oui-table add 00e075 Polycom/Veritel_phone___
voice vlan oui-table add 00e0bb 3Com_phone______________
voice vlan oui-table add 108ccf MyCiscoIPPhones1
voice vlan oui-table add 40f4ec MyCiscoIPPhones2
voice vlan oui-table add 8cb64f MyCiscoIPPhones3
voice vlan id 111
voice vlan cos 6 remark
interface ethernet e1
voice vlan enable
exit
interface ethernet e1
voice vlan cos mode all
exit
interface ethernet e2
voice vlan enable
exit
interface ethernet e2
voice vlan cos mode all
exit
interface ethernet e3
voice vlan enable
exit
interface ethernet e3
voice vlan cos mode all
exit
interface ethernet e4
voice vlan enable
exit
interface ethernet e4
voice vlan cos mode all
exit
interface ethernet e5
voice vlan enable
exit
interface ethernet e5
voice vlan cos mode all
exit
interface ethernet e6
voice vlan enable
exit
interface ethernet e6
voice vlan cos mode all
exit
interface ethernet e7
voice vlan enable
exit
interface ethernet e7
voice vlan cos mode all
exit
interface ethernet e8
voice vlan enable
exit
interface ethernet e8
voice vlan cos mode all
exit
interface ethernet e9
voice vlan enable
exit
interface ethernet e9
voice vlan cos mode all
exit
interface ethernet e10
voice vlan enable
exit
interface ethernet e10
voice vlan cos mode all
exit
interface ethernet e11
voice vlan enable
exit
interface ethernet e11
voice vlan cos mode all
exit
interface ethernet e12
voice vlan enable
exit
interface ethernet e12
voice vlan cos mode all
exit
interface ethernet e13
voice vlan enable
exit
interface ethernet e13
voice vlan cos mode all
exit
interface ethernet e14
voice vlan enable
exit
interface ethernet e14
voice vlan cos mode all
exit
interface ethernet e15
voice vlan enable
exit
interface ethernet e15
voice vlan cos mode all
exit
interface ethernet e16
voice vlan enable
exit
interface ethernet e16
voice vlan cos mode all
exit
interface ethernet e17
voice vlan enable
exit
interface ethernet e17
voice vlan cos mode all
exit
interface ethernet e18
voice vlan enable
exit
interface ethernet e18
voice vlan cos mode all
exit
interface ethernet e19
voice vlan enable
exit
interface ethernet e19
voice vlan cos mode all
exit
interface ethernet e20
voice vlan enable
exit
interface ethernet e20
voice vlan cos mode all
exit
interface ethernet e1
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e2
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e3
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e4
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e5
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e6
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e7
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e8
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e9
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e10
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e11
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e12
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e13
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e14
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e15
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e16
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e17
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e18
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e19
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e20
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e21
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e22
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e23
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e24
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet g1
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet g2
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet g3
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet g4
lldp optional-tlv port-desc sys-name sys-desc sys-cap 802.3-mac-phy 802.3-lag 802.3-max-frame-size
exit
interface ethernet e1
lldp med notifications topology-change enable
exit
interface ethernet e2
lldp med notifications topology-change enable
exit
interface ethernet e3
lldp med notifications topology-change enable
exit
interface ethernet e4
lldp med notifications topology-change enable
exit
interface ethernet e5
lldp med notifications topology-change enable
exit
interface ethernet e6
lldp med notifications topology-change enable
exit
interface ethernet e7
lldp med notifications topology-change enable
exit
interface ethernet e8
lldp med notifications topology-change enable
exit
interface ethernet e9
lldp med notifications topology-change enable
exit
interface ethernet e10
lldp med notifications topology-change enable
exit
interface ethernet e11
lldp med notifications topology-change enable
exit
interface ethernet e12
lldp med notifications topology-change enable
exit
interface ethernet e13
lldp med notifications topology-change enable
exit
interface ethernet e14
lldp med notifications topology-change enable
exit
interface ethernet e15
lldp med notifications topology-change enable
exit
interface ethernet e16
lldp med notifications topology-change enable
exit
interface ethernet e17
lldp med notifications topology-change enable
exit
interface ethernet e18
lldp med notifications topology-change enable
exit
interface ethernet e19
lldp med notifications topology-change enable
exit
interface ethernet e20
lldp med notifications topology-change enable
exit
interface ethernet e21
lldp med notifications topology-change enable
exit
interface ethernet e22
lldp med notifications topology-change enable
exit
interface ethernet e1
lldp med enable network-policy poe-pse
exit
interface ethernet e2
lldp med enable network-policy poe-pse
exit
interface ethernet e3
lldp med enable network-policy poe-pse
exit
interface ethernet e4
lldp med enable network-policy poe-pse
exit
interface ethernet e5
lldp med enable network-policy poe-pse
exit
interface ethernet e6
lldp med enable network-policy poe-pse
exit
interface ethernet e7
lldp med enable network-policy poe-pse
exit
interface ethernet e8
lldp med enable network-policy poe-pse
exit
interface ethernet e9
lldp med enable network-policy poe-pse
exit
interface ethernet e10
lldp med enable network-policy poe-pse
exit
interface ethernet e11
lldp med enable network-policy poe-pse
exit
interface ethernet e12
lldp med enable network-policy poe-pse
exit
interface ethernet e13
lldp med enable network-policy poe-pse
exit
interface ethernet e14
lldp med enable network-policy poe-pse
exit
interface ethernet e15
lldp med enable network-policy poe-pse
exit
interface ethernet e16
lldp med enable network-policy poe-pse
exit
interface ethernet e17
lldp med enable network-policy poe-pse
exit
interface ethernet e18
lldp med enable network-policy poe-pse
exit
interface ethernet e19
lldp med enable network-policy poe-pse
exit
interface ethernet e20
lldp med enable network-policy poe-pse
exit
interface ethernet e21
lldp med enable network-policy poe-pse
exit
interface ethernet e22
lldp med enable network-policy poe-pse
exit
lldp med network-policy 1 voice vlan 111 vlan-type tagged
interface range ethernet e(1-22)
lldp med network-policy add 1
exit
interface vlan 199
ip address 199.16.30.77 255.255.255.0
exit
ip default-gateway 199.16.30.3
interface vlan 1
no ip address dhcp
exit
no bonjour enable
bonjour service enable csco-sb
bonjour service enable http
bonjour service enable https
bonjour service enable ssh
bonjour service enable telnet
hostname psw1
line console
exec-timeout 30
exit
line ssh
exec-timeout 30
exit
line telnet
exec-timeout 30
exit
management access-list Management1
permit ip-source 10.22.5.5 mask 255.255.255.0
exit
logging 199.16.31.33 severity debugging description mysysloghost
aaa authentication enable Console local
aaa authentication enable SSH tacacs local
aaa authentication enable Telnet local
ip http authentication tacacs local
ip https authentication tacacs local
aaa authentication login Console local
aaa authentication login SSH tacacs local
aaa authentication login Telnet local
line telnet
login authentication Telnet
enable authentication Telnet
password admin
exit
line ssh
login authentication SSH
enable authentication SSH
password admin
exit
line console
login authentication Console
enable authentication Console
password admin
exit
username admin password admin level 15
power inline usage-threshold 90
power inline traps enable
ip ssh server
snmp-server location in-the-closet
snmp-server contact [email protected]
ip http exec-timeout 30
ip https server
ip https exec-timeout 30
tacacs-server host 1.2.3.4 key spaceballz timeout 3 priority 10
clock timezone -7
clock source sntp
sntp unicast client enable
sntp unicast client poll
sntp server 199.16.30.1
sntp server 199.16.30.2
ip domain-name mydomain.com
ip name-server 199.16.5.12 199.16.5.13
ip telnet server
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