CME: assign extension sip trunk

Similar Questions

• How a dial-peer SIP Trunk using a Registrar?

  Hi all

  I'll put up a GUY using two records.  With the generous help of people on this forum, I got records in doubles to work.  But now I need to know how to configure the dial-peer to use registration information.  For example, I have this set up:

  SIP - ua

  Password 08114342101A0A1A43 7 authentication username 5555555555

  ...

  IPv4:11.11.11.11:6034 at the office 1 expires 3600

  Registrar 2 ipv4:22.22.22.22:6035 expires 3600

  How to configure a dial-peer to send traffic to one of the recordings?  I tried this, and it does not work:

  Dial-peer voice voip 105

  Description * outgoing SIP Trunk call *.

  translation-profile outgoing PSTN_Outgoing

  destination-model 91%...

  session protocol sipv2

  Registrar of target session? WHAT SHOULD I USE HERE?

  codec voice-class 2

  DTMF-relay rtp - nte

  No vad

  Thanks in advance.

  Hi Tod,

  Please go through this post, hope that it answers your question:

  http://tekcert.com/blog/2011/02/03/CME-configuration-example-SIP-trunks-ViaTalk-and-voipms

  Specify the article accordingly.

  Kind regards

  Kevin

• 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.

  Can you please check what is the IP of CMF address you configured on the SIP trunk in CUCM?

  -Vivek

• Configure (fxs) analog phones with Sip trunk

  Hi all

  IS - this configuration FXS possiple to knit on SIP TRUNK? I HAVE 16 FXS ports and voip gateway 2921 cisco.

  Configuring fxs to work with her?

  any help thanks.

  Is there a PBX at stake here, i.e. CUCM or CME?

  In both cases, you have set good dial-peers to point to the FXS ports to match the destination of the SIP trunk, which is pretty simple.

• CUCM v8.5 with 3 SIP Trunks to the Lync Server - Route algorithm of Distribution for the Group

  I CUCM connected to three different Lync server via 3 different SIP trunks.

  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

• Sip Trunk design question

  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

  Simple answer use ALWAYS the CUBE.  With IOS 15.1 T and more you have security against fraud free of charge that you can use to restrict which can address IP contacted the CUBE, that's all you need.

  HTH,

  Chris

• 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:

  8900 (jabber) - CUCM - MCU - SX10

  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...

  2. Yes, using the MCU would be much simpler, easier and, in my opinion, a better solution to be implemented. Here is an example of how a stand-alone SX10 could dial the IP address of the MCU standard automatic.

  You can also assign a SIP URI address to the meeting point which the SX10 can connect to.

  /Jens

  Please note the answers and score the questions as "answered" as appropriate.

• SIP trunk behind a router using NAT

  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?

  Does anyone do this?

  I ask because I'm having problems to make my SIP trunk to work and my router for cube is behind my generic service provider router, which makes the NAT. I just want to rule this out as a problem.

  Has anyone else done this? Or is it really impossible?

  Thank you very much

  Tom

  Hello

  As NAT works fine SIP would work properly as the Protocol.

  Here is the RFC for "NAT Traversal practices for Client - Server SIP"

  https://Tools.ietf.org/html/rfc6314

  HTH

  JB

• Source IP address of the originating call to sip trunk

  I'm unable to set address ip source from trunk sip call. I read somewhere that the source ip address of the call must be the ip address that call device registered to. I also checked "run in all active nodes such as suggested by cisco. We have 3 Sub cluster and 1 pub however still call orgianate to a particular Sub.

  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?

  Suresh

  Please note all useful messages.

• DTMF in SIP Trunk problem

  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

  ITSP say that he send with standard RFC2833 dtmf (it equals to rtp-net, I know), but when I am 'debug ccsip message', the results show the dtmf with rfc2833 sends us

  16 August 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  INVITE sip: [email protected] / * /: 5060. user = phone SIP/2.0
  Via: SIP/2.0/UDP 10.105.40.34:5060; branch = z9hG4bKejhh8jixkobhnb7ykvi87vuj8
  Call ID: [email protected]/ * /.
  From: sip: [email protected] / * />; tag = c3cx3vcw-CC-40
  To: sip: [email protected] / * /; user = phone >
  CSeq: 1 INVITE
  Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see
  Max-Forwards: 69
  Supported: 100rel, timer
  User-Agent: Huawei SoftX3000 V300R010
  Session time-out: 300
  Min - SE: 90
  Contact: sip: [email protected] / * /: 5060; user = phone >
  Content-Length: 374
  Content-Type: application/sdp
  v = 0
  o = HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34
  s = call Sip
  c = IN IP4 10.105.40.34
  t = 0 0
  m = audio 27762 RTP / AVP 8 0 18 4 2 98 99 102
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a G729/8000 rtpmap:18 =
  a = rtpmap:4 G723/8000
  a = rtpmap:2 G726-32/8000
  a = rtpmap:98 G726-40/8000
  a = rtpmap:99 G726-32/8000
  a = rtpmap:102 G726-24/8000
  a = ptime:20
  a = fmtp:18 annex b = No.
  It is a message to guest (with sdp) of ITSP
  As you can see the line with red color must have a code with number of 101 but rather a code with number of 18
  In 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
  Codec g711ulaw
  DTMF-relay rtp - nte
  !
  VoIP voice 10 Dial - peer---> router for ITSP and vice versa
  destination-model. T
  session protocol sipv2
  session target ipv4:10.105.40.34
  incoming called-number. T
  DTMF-relay rtp - nte
  Codec g711ulaw
  I have configured cucm with a sip section to my favorite router with active PSG and RFC2833
  BUT HE DIDN'T THERE WAS NO DETECTION OF DTMF IN INCOMING CALLS AND OUTGOING
  I 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 work
  I change the codec but does not solve the problem
  There 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)
  Please give me a solution to solve the problem between Cisco 3945 and ITSP
  Concerning

  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.

• 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.

• Problem with a SIP Trunk between VCS and CUCM

  Hello world

  I created a SIP Trunk between control VCS (X7.1) and a CUCM 8.6.

  I followed this Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_6-1_7_8_and_X7 - 0.pdf deployment guide.

  The trunk is in place, but I have the following problem: I can ring since an EX90 recorded on the VCS for a registered on the CUCM 9951 but when I take the call, I have a busy signals (audio and video). When I on the EX90 9951 ring, I have the following message on VC: can not reach the contact.

  I have a 503 service unavailable status on the call history and "call rejected" on the event log.

  My transformation is OK and the call is sent to my neighbor CUCM zone

  You have an idea to solve this problem?

  Thanks for your help

  Best regards

  Bruno Z.

  Hello

  Yes, it's true! It seems at first glance a problem with codec negotiation when I see the reason 47 code.

  in any case set the region to G.711 trunk phone in case if you have different device pool and the trunk region and ip-phones.

  make G.711 end to end and test again.

  Thank you

  Alok

• VCS SIP Trunk

  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 Server

  voice class codec 1
  g711ulaw codec preference 1

  translation of the voice-rule 1
  rule 1 / ^ 8 / /0056/
  !
  voice translation-rule 2
  rule 1 5.0 / /17772114zzz/
  !
  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 called

  Dial-peer voice 1 voip
  CALLCENTRIC description
  entrants IN translation-profile
  translation-profile outgoing OUT
  destination-model 8.T
  codec voice-class 1
  session protocol sipv2
  session target sip-Server
  incoming called-number 17772114zzz
  SIP DTMF-relay-notify rtp - nte
  !
  Dial-peer voice 2 voip
  CUCM description
  destination-model 500
  media stream-autour
  codec voice-class 1
  session protocol sipv2
  session target ipv4:192.168.10.116
  incoming called number 8.T
  SIP DTMF-relay-notify rtp - nte

  SIP - 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

  Please note all useful posts

• CUCM SIP-trunk multiple Ports

  I would like to know if it is possible to create a Script of SIP standardization in CUCM which change the Destination SIP port based on the phone which makes the phone.

  Current issue:

  The ITSP provider asked that I send calls from different regions to the same address IP SBC but a different port.

  Example:

  Phone (A) makes a call

  RP - RL - RG---> SIP Trunk (5060)---> CUBE (5060<-->5001)---> ITSP1

  Telephone (B) makes a call

  RP - RL - RG---> SIP Trunk (5060)--->---> CUBE ITSP1 (5060<-->5002)

  I can do it on the CUBE, but which require several dial-Exchange and SIP profiles, is it possible in the CUCM to change the sip by device-pool port?.

  Or is there another I can match and send calls via the same destination with different dial-peer sip ports.?

  Thank you.

  Zakiab,

  You can do this, its all just impossible. Your signage ports are defined not on the endpoints, but on the B2BUA (CUCM or CUBE). The CUBE, you have the most flexibility, because you can change the port of signs based on your dial-peers. However, as you pointed out quite rightly, need you an insurmountable amount of dial-peers to have this for each endpoint. So I suggest tell you your ITSP to change their design or moving to a new.