SPAN and SIP Trunk recording in parallel
I'm looking to get away using a dictaphone SPAN and use SIP automatic trunk of the record (by using a recorder DMS Verint pool UDP) calls.
My question is, if I apply SIP trunk recording simultaneously with recording SPAN, this mind? I need to make a CEP of the solution, but cannot stop the current recording.
Thanks for any help you can give me!
With the traditional port SPAN record in the Callmanager ignores the phone is registered, so when you activate the current record there will be no impact.
However you can finish by double flow at the end of voice recording, so not sure what verint would do that.
Tags: Cisco Support
Similar Questions
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CallManager Express SIP trunk problem
Hi all here.
I have a problem with my SIP trunk CallManager Express (version 10.0). On my site already configured trunk SIP between CME and CUE. I have configured DN for SIP phone and SIP phones recorded on CME successfully, but... As soon as SIP phone registered on CME, DIAL-PEER for CME and CUE connection changes. Especially 'session target ipv4' automatically changes to my IP phone SIP and all calls go to my SIP phone and does not reach the auto attendant.
How can I solve this problem?
Hello
Following is the error of CUCM;
WARNING: 399 UnicompCM "cannot find a Device Manager for the request received on port 55549 leave 192.168.10.2.
This probably means that CME use 192.168.10.2 for source packages SIP CUCM however SIP trunk in CUCM is not directed to this IP address, it is useful to point out to another IP interface of the GUY and therefore dismiss this appeal.
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Hello world
I created a SIP Trunk between control VCS (X7.1) and a CUCM 8.6.
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Yes, it's true! It seems at first glance a problem with codec negotiation when I see the reason 47 code.
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is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?
Dear all,
Need help :)
1.
is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?
my customers will not buy TP - lic :)
Topology:
8900 (jabber) - CUCM-(SIP) - SX10
2.
Customer, have MCU machine.
If not possible. is it possible, meeting point of MCU use?
Topology:
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1. Yes, but it is not that simple and straightforward that the SX10 can be reached by calling is the public IP address and CUCM does not support the IP address numbering is not a SIP address format...
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Please note the answers and score the questions as "answered" as appropriate.
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!
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Well, we can set up incoming and outgoing calls with no problems during this test phase, but we will succeed voice entering.
We don't have voices coming out of the voice gateway.
We checked with the ISP and we see the RTP of ISPS to Cisco 2911Voice gateway traffic, but we did not see packets RTP voice to the ISP gateway.
In fact, it was not all RTP packets arriving at the voice gateway on the internal network.
Might be a routing problem?
Internal CUCM and phones require Ip routing SIP from the ISP server access? If I understand correctly the devices internal only need to know how to get to the voice gateway Cisco2911, so it can function as a Proxy traffic and route to the SIP server?
Thank you
In addition to the comments of Chris,
1. There is a routing problem: IP phones should see the route to the ISP, even if they are inside a NAT.
2. If you want that:
-Just IP phones reach the 2911 and IP of 2911 present the call to the ISP.
-the Loopback0 bring the H323
- And the int GigabitEth 0/0 for the SIP
then
Configure the 2911 as a CUBE in path mode
Use the redirection ip2ip
Configure dspfarm on the 2911
3 also check this:
If you have not seen all the voice gateway to ISP RTP packets
Then
-Check if the transport of the ISP session is TCP or UDP.
-Set up a GUY on the 2911 to check the communication between the {2911 and ISP} and {2911 and CCM}
Kind regards
Antra
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Password 08114342101A0A1A43 7 authentication username 5555555555
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Hello
Is it possible to use a SIP trunk to a provider SIP ITSP having the CUBE / router gateway behind a firewall using a NAT?
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Hello
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Here is the RFC for "NAT Traversal practices for Client - Server SIP"
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HTH
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RG is composed of the following elements:
LYNC SIP TRUNK 1 (1.1.1.1)
LYNC SIP TRUNK 2 (2.2.2.2)
LYNC SIP TRUNK 3 (3.3.3.3)
The route group was built with "Top Down" the algorithm of distribution. The first SIP trunk knows congestion and some calls are never routed to secondary and tertiary SIP trunks.
Based on all the forum posts I've seen - it seems that I have to configure the algorithm of group distribution of ranges as 'circular '.
If I change the algorithm group of "Circular" lines - can I expect the following results:
1. first call will go through LYNC SIP TRUNK 1
2. second call will cross LYNC SIP TRUNK 2
3. third call will cross LYNC SIP TRUNK 3
When I change the algorithm of distribution of the route to the 'circular' group and click 'SAVE', I am invited on "RESET". This service assigns to the existing calls through SIP Trunks?
TIA,
Amir
Hello
the circular algorithm will work the way you mentioned, but I suggest to try and make sure that is not perform integration
to reset the SIP trunk actually active calls should not be made because the gose media directly between endpoint unles syou use something like trust rely wher evous enforce calls to go to the SIP
Therefore, in most cases is should be fine just try configuration at the time of the calls will face service disruption
hope this helps
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Hello
I have a requirement to pass an h323 to SIP environment environment. I'm looking for good practices, especially around security. I have 2 servers CUCM (8.5) in cities separated for redundancy. I have also 2 voice gateways which, at the present time, h323 to the PSTN, are each located in different cities.
My requirements are:
1. create a sip trunk instead of the supplier of the use of PRI.
2 If the Wan link fails on a gateway provider, router replacing in the other location should be able to receive installation messages and if a user connects via extension mobility, should be able to answer the call.
Is there a simplified design docos on for this? I hesitate to create a SIP trunk directly to the supplier for safety, thus thinking to end the call on the routers of voice with the CUBE. I am sure that it is managed from the factory and would appreciate comments.
See you soon!
Pieter
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Source IP address of the originating call to sip trunk
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I also undergoes several changes i - e DP and list route but think not worked for me. Can someone help to sortout this problem?
Let's say if your ip phone registered in sub - and "run on all nodes" enabled, then the source ip address is always sup - A s ip address regardless of the device pool config on the sip trunk.
Could please explain you the device pool config in phones IP & SIP trunk?
you have enabled the "run on all nodes" for a list of course also?
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Hello
I have a problem in case of detection of the DTMF
We have a SIP of the ITSP Trunk and everything is ok except DTMF.
The sip trunk is between ITSP and router 3945
ITSP <->3945 <->CUCM 10.5
I have try all method such as rtp - nte, h245 alphanumeric and h245-signal, info-sip, sip-kpml,... in line-peer to ITSPs
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16 August 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:INVITE sip: [email protected] / * /: 5060. user = phone SIP/2.0Via: SIP/2.0/UDP 10.105.40.34:5060; branch = z9hG4bKejhh8jixkobhnb7ykvi87vuj8Call ID: [email protected]/ * /.From: sip: [email protected] / * />; tag = c3cx3vcw-CC-40To: sip: [email protected] / * /; user = phone >CSeq: 1 INVITEAllow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, seeMax-Forwards: 69Supported: 100rel, timerUser-Agent: Huawei SoftX3000 V300R010Session time-out: 300Min - SE: 90Contact: sip: [email protected] / * /: 5060; user = phone >Content-Length: 374Content-Type: application/sdpv = 0o = HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34s = call Sipc = IN IP4 10.105.40.34t = 0 0m = audio 27762 RTP / AVP 8 0 18 4 2 98 99 102a = rtpmap:8 PCMA/8000a = rtpmap:0 PCMU/8000a G729/8000 rtpmap:18 =a = rtpmap:4 G723/8000a = rtpmap:2 G726-32/8000a = rtpmap:98 G726-40/8000a = rtpmap:99 G726-32/8000a = rtpmap:102 G726-24/8000a = ptime:20a = fmtp:18 annex b = No.It is a message to guest (with sdp) of ITSPAs you can see the line with red color must have a code with number of 101 but rather a code with number of 18In my "ccsip media debug' output show that the method of negotiation between me and itsp 'incoming voice. 'It's my router config:voip phone service
No IP trust to authenticate
allow connections h323 to SIP
allow connections sip h323
allow sip to sip connections
SIP
interface FastEthernet0/0/1 source control binding
bind media source interface FastEthernet0/0/1
min - to 300 session expires-300
!Dial-peer voice 2 voip---> router CUCM and vice versa
translation-profile outgoing toos
destination-model 42584...
session protocol sipv2
session target ipv4:10.20.30.70
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DTMF-relay rtp - nte
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VoIP voice 10 Dial - peer---> router for ITSP and vice versa
destination-model. T
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session target ipv4:10.105.40.34
incoming called-number. T
DTMF-relay rtp - nte
Codec g711ulawI have configured cucm with a sip section to my favorite router with active PSG and RFC2833BUT HE DIDN'T THERE WAS NO DETECTION OF DTMF IN INCOMING CALLS AND OUTGOINGI even test dial-position 10 without any method of dtmf-relay because I want to configure it with the incoming voice method, but it does not workI change the codec but does not solve the problemThere is an interesting point and that is, if use Elastix 3945 of the router and configure dtmf among elastix and itsp as "inbound" method and method dtmf among elastix and cucm as 'rfc2833' everything is OK (ITSP<--Inbound--> Elastix <--rfc2833-->CUCM)--rfc2833--> --Inbound-->Please give me a solution to solve the problem between Cisco 3945 and ITSPConcerning->->It would be more accurate to say that the problem has been bypassed by using a transcoding resource. The correct resolution here would be to open a ticket to trouble with the ITSP. I agree that the PROMPT message you show doesn't include advertising RFC2833 in the SDP offer. It is a problem that only the carrier can rectify.
-
Hi all
Is it possible to register a VCS as a SIP endpoint? I want to be able to do the voice only calls destined for the PSTN to VCS by registering the VCS as a SIP proxy endpoint.
Thank you
AA
Hi AA!
If you look more closely the mechanisms used in venture capital, the required functionality
is somehow already implemented with the OCS-Rleay and X 7 the B2BUA OCS.
(who actually sign up on the OCS/Lync server).
So in theory, it could be an application feature and some future versions now.
But currently, I don't see that this is possible without an external host playing the role of SIP - UA.
your registration to your proxy sip and offering VCS cases as a SIP trunk.
Most of the Services SIP / PSTN GW also has the capablity to implement to your
IP/domain of VCS, his less interesting to know.
Please vote the answers and check them in as solved if they are.
-
SIP trunk CUBE with Callcentric - incoming unanswered call
I'm doing some tests with a Sip trunk with a provider called Callcentric.It is a CUBE scenario. I use a SIP to the CUCM trunk.
I have a Cisco Callmanager and a virtual router 7200 (GNS3) as a gateway (C7200-ADVENTERPRISEK9-M), Version 12.4 (24).A CPIC connected to Callmanager, I call out to PSTN and it works perfectly.
When I do an incoming call to the PSTN to the softphone destination, she sounds, but when I press the answer button, the call is not connected and the analog phone in RTC can listen back ringtone.
Do you have any idea what it could be?
Some relevant configurations:voip phone service
allow sip to sip connections
Fax protocol cisco
SIP
interface FastEthernet0/0 source control binding
bind media source interface FastEthernet0/0
Registrar Servervoice class codec 1
g711ulaw codec preference 1translation of the voice-rule 1
rule 1 / ^ 8 / /0056/
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voice translation-rule 2
rule 1 5.0 / /17772114zzz/
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voice translation-rule 3
rule 1 /17772114zzz/ /500/voice translation-profile IN
definition of 3 called
!
FLIGHT voice translation-profile
definition of call 2
translate 1 calledDial-peer voice 1 voip
CALLCENTRIC description
entrants IN translation-profile
translation-profile outgoing OUT
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session protocol sipv2
session target sip-Server
incoming called-number 17772114zzz
SIP DTMF-relay-notify rtp - nte
!
Dial-peer voice 2 voip
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media stream-autour
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session target ipv4:192.168.10.116
incoming called number 8.T
SIP DTMF-relay-notify rtp - nteSIP - ua
credentials of username, password 7 17772114zzz 094F471B100928 callcentric.com Kingdom
authentication username 17772114zzz password 7 121A0C051B0803 callcentric.com Kingdom
no remote-party-id
Registrar dns:callcentric.com expires 3600
DNS:callcentric.com SIP server
Home-Office
Thank you guys.Hello
Please configure the sip requires low profile to remove the header and apply this profile in the dial-peer 1 outgoing.
SIP-class voice profiles 1
response header 200 sip requires DELETE
If this does not work under Dial-peers, try to apply globally.
voip phone service
SIP
SIP profiles 1
Suresh
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