NexVortex SIP trunk and UC500 default timeout settings?

Hey guys,
I'm doing a little SIP trunk testing to determine a good provider for my customer base, and had some general questions as I can't seem to get outgoing or incoming phone calls to work at all.
To keep things simple, I'm using an 8user UC540W with 3 IP phones - a 525G, a 524G, and a 7937 conference phone.  I have a static IP on the UC540, have run through the telephony wizard and everything seems to be working on the LAN/PBX side of things.  The big difference, and the major variable that we are working with (I believe), is that we're working with Satellite internet connectivity rather than terrestrial Internet connectivity.  This is an Enterprise satellite connection, and we have run voice over the connection without problems, but this is our first attempts at SIP trunking from a UC500.  Due to the latency involved inherent in satellite (ping times around 550-700ms), I believe that either UC540 or NexVortex server/switch is timing out.  Is there any way to determine what the default setting is for a SIP acknowledgement on the UC540 and change this if it is too small?
Here is what I have found, if it is helpful:
Outgoing calls:
1. The SIP provider, NexVortex, says that they are seeing an invite from the UC540, but not on port 5060.  On the two calls that we tested, it first saw an invite on 63452, and then on 51677.  Is there any reason why this would not be sent out on 5060?
Incomign calls:
1. On incoming calls, Nexvortex is routing the calls to the proper IP, but is then receiving an "error 500 reason Q850" from the UC540.  What does this error mean?
I am also attaching my config in the event that it helps.  When I look at the SIP trunk status in CCA, it does not show that registration is working, so I assume that's a good place to start.
Lastly, the guys over at NexVortex don't seem to run across the UC500 very often.  If anybody has setup their UC500 to work with NexVortex and wouldn't mind posting a screenshot from CCA (feel free to remove usernames and passwords), I'd appreciate it.  I'm not certain that I have all of the information in the right places.
Thanks,
Seth

Hi Steven,
Thanks for the continued help.
I was able to make the changes in the config.  Here are snapshots from the current config:
dial-peer voice 1000 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3000 voip
description IncomingSIP
translation-profile incoming IncomingSIP_Called_4
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 14068906254$
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3001 voip
description IncomingSIP2
translation-profile incoming IncomingSIP2_Called_5
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 1406890624[2-3]$
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
dial-peer voice 3002 voip
incoming called-number 14068906254$
no dial-peer outbound status-check pots
sip-ua
authentication username nomadgcs password 7 *removed*
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:66.23.129.253:5060 expires 3600
sip-server ipv4:66.23.129.253:5060
connection-reuse
host-registrar
We are calling from within the 406 area code, so when we dial the number with the leading 406, we get a message saying "You don't need the area code" from the telephone company.  When we dial this from a cell, we get the following:
1. 4068906254 - "All circuits are busy, please try your call again..."
2. 8906254 - rings once, then no sound, then disconnects after about 10 seconds.
I don't know if this would factor in at all, but our NexVortex account is setup to deliver 14068906254 to the UC500, but would NexVortex deliver the entire string of characters if it is only receiving 4068906254 or 8906254?
Thanks,
Seth

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    ASA blocks ports associated to the new media endpoints
    I can resolve this by allowing the ports in the ACL, but suprised this is not working as re-invites to change media endpoints is to be expected in SIP conversation.
    Regards,
    AJ

    Below is the script you can use to reproduce this. Points worth mentioning.
    Initial invite sets up the media between SIP Trunk and a media device ( 10.1.2.150) in the inside network, SIP signalling will be with 10.1.2.100. At this poit RTP flows freely between the SIP Trunk and the media device.
    If the call is fax, a re-invite will occur and this will cause the IP address to change in the SDP. The new media endpoint becomes 10.1.2.151 (This device is SIP and Media (T38) capable).
    For every SIP call we establish 10.1.2.150 will be used for media, we do not want to change this behaviour.
    ASA 8.3 (2)
    conf t
    interface Ethernet0/0
    nameif Inside_Voice
    security-level 100
    ip address 10.1.2.11 255.255.255.0 standby 10.1.2.12
    exit
    interface Ethernet0/1
    nameif Outside_SIP_Trunk
    security-level 0
    ip address 10.1.60.254 255.255.255.0 standby 10.1.60.253
    exit
    object-group network SIP_trunks
    network-object 1.2.3.0 255.255.255.0
    exit
    object-group service SIP_service
    service-object tcp destination eq sip
    service-object udp destination eq sip
    exit
    object-group network SIP_inside_servers
    network-object host 10.1.2.100
    exit
    access-list Outside_SIP_in extended permit object-group SIP_service object-group SIP_trunks object-group SIP_inside_servers
    access-group Outside_SIP_in in interface Outside_SIP_Trunk
    route Outside_SIP_Trunk 0.0.0.0 0.0.0.0 10.1.60.1
    class-map inspection_default
    match default-inspection-traffic
    exit
    policy-map global_policy
    class inspection_default
    inspect dns preset_dns_map
    inspect ftp
    inspect h323 h225
    inspect h323 ras
    inspect ip-options
    inspect netbios
    inspect rsh
    inspect rtsp
    inspect skinny
    inspect esmtp
    inspect sqlnet
    inspect sunrpc
    inspect tftp
    inspect sip
    inspect xdmcp
    inspect icmp
    inspect icmp error
    lass class-default
    set connection decrement-ttl
    exit
    service-policy global_policy global
    end

  • Design Question Sip Trunk

    Hi,
    I have a requirement to move from a h323 environment to a SIP environment. I am looking for best practises especially around security. I have 2 CUCM servers (8.5) located in separate cities for redundancy. I also have 2 voice gateways which at the moment are h323 to the PSTN, each located at different cities.
    My  requirements are:
    1. Creat a sip trunk to the provider instaed of using PRI.
    2. If the Wan link fails on one gateway to provider, the alternate router in the other location should be able to receive the setup messages and if a user logs on via extension mobility, should be able to answer the call.
    Are there any simplified design docos about for this? I am hesitant to create a SIP trunk straight to the provider for security, so thinking of terminating the call on the voice routers with CUBE. I'm pretty sure this is run of the mill and would appreciate some input.
    Cheers!
    Pieter

    +5 to Chris..Always use CUBE...
    Here are more ideas..
    1. Create two sip trunks, first one to Cube 1, second to CUBE 2
        on your sip trunk set DTMF as no preference
    2. Assign the trunks to CUCM group with your two servers in it
    3. Configure route groups with Circular algorithm distribution (this way you have  load balancing on your two cubes)
    4. Configure dial-peers on your CUBE gateway to point in preferential order to your cucm servers
    5. Use dtmf-relay rtp-nte on your dial-peer (ensure you have sip as the protocol on your dial-peers)
    6. configure your codec selection properly on your dial-peers
    7. configure your region settings properly between your sip trunk and phones.
    8. Evaluate if you need xcoders, provide one if you do and ensure you set your region correctly between your xcoder and CUBE
    9. Is there voicemail involved? Ensure you set your region settings between cube and voicemail correctly, otherwise calls to voicemail may invoke xcoder (from experience)
    10. Provision adequate bandwidth for your calls. Once you move to sip, you loose the luxury of e1 channels. You are solely relying on Bandwidth. Ensure you have adequate bandwidth for your concurrent calls
    11. Provision QoS for your calls
    12. Is there fax involved? You need to think carefully on this one. What fax method do you want to use. T.38/pass through. Does your provider support T.38?
    13. Have a thorough test plan. Test call transfers, call on hold, call forward etc
    Just a few pointers..Careful planning and implementation is required for a successful SIP implementation

  • Best Practice to Integrate CER with RedSky E911 Anywhere via SIP Trunk

    We are trying to integrate CER 9 with RedSky for V911 using a SIP trunk and need assistance with best practice and configuration. There is very little documentation regarding "best practice" for routing these calls to RedSky. This trunk will be handling the majority of our geographically dispersed company's 911  calls.
    My question is: should we use an IPsec tunnel for this? The only reference I found was this: http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/virtual-office/deployment_guide_c07-636876.htmlm which recommends an IPsec tunnel for the SIP trunk to Intrado. I would think there are issues with an unsecure SIP trunk for 911 calls. Looking for advice or specifics on how to configure this. Does the SIP trunk require a CUBE or is a CUBE only required for the IPsec tunnel?
    Any insight is appreciated.
    Thank you.

    you can use Session Trace in RTMT to check who is disconnecting the call and why.

  • Unable to perform call transfer & call park through SIP Trunk (SKYPE)

    The Scenario is:
    I have set up a SIP trunk to SKYPE and we are able to make outbound call to a number via SIP Trunk.
    After the call is established, when we tried to make call transfer, the call DROP and the phone at the other end shows error "Temp Fail".
    I tried to "enable MTP" in SIP Trunk and We are able to perform call-transfer but it limits the call session to 1.
    Anyone has facing the same issue?

    MTP is needed to invoke supplementary functions like hold, transfer etc. Make sure that the MTP is checked on SIP trunk, MTP is assigned to the MRGL of the device pool on SIP trunk and has sufficient resources.
    HTH
    Manish

  • Keep alive a SIP Trunk?

    Hi!
    I'm trying to register a SIP Trunk to a SIP server. The trunk registration is done, but not keep alive. The trunk register with SIP server when an outgoing call starts, but when this call ends, the SIP trunk closes the connection with SIP server. Then, the
    outgoing calls work OK, but the incoming calls doesn't work because the SIP Trunk is unregistered while no active outgoing calls.
    Then, can i keep alive the SIP Trunk registration with SIP Server?
    Thanks!!!

    You need to talk to the SIP provider and get them enable OPTIONS on the SIP Trunk and enable OPTIONS on the PSTN Gateway. Check the registration interval of the SIP trunk on the Gateway and try increasing it to a higher value.
    http://thamaraw.com

  • Telco Messages via PRI (VG) connected to CUCM 9.X via SIP Trunk??

    Hello 
    Questions:
    Should I hear tel-co message on an IP Phone if a call that is meant to be long distance is sent to the  gateway without a one (1).  Currently these calls simply ring until disconnect, but work properly if the user dials a 1, the user expects to get a message from the provider and I am wondering if the SIP trunk between GW and CUCM is not allowing it?
    Scenario:
    IP Phone --> CUCM (SIP Trunk) --> ISR 2901(PRI) --> PSTN
    In CUCM we have a local pattern 9.[2-9]XX[2-9]XXXXXX
    If the IP Phone dials a number that is actually a long distance number, which would require a 1, but they dial it as a 10 digit local number, the call gets to the gateway and IP Phone hears ringing but it, rings until it eventually disconnects.  At this point I believe the gateway is sending a cause code back to CUCM and I would expect error message back from the telco informing the caller of the issue, such as "Please dial 1 before a long distance number".  Is there a specific setting on the SIP trunk and or Gateway to achieve this?
    See Q931 Debug 
         Bearer Capability i = 0x8090A2 
                    Standard = CCITT 
                    Transfer Capability = Speech  
                    Transfer Mode = Circuit 
                    Transfer Rate = 64 kbit/s 
            Channel ID i = 0xA9838F 
                    Exclusive, Channel 15 
            Display i = 0xB1, 'London Hydro' 
            Calling Party Number
    UCS5-GW-02# i = 0x2181, '5193334444' 
                    Plan:ISDN, Type:National 
            Called Party Number i = 0xA1, '5191112222' 
                    Plan:ISDN, Type:National
    *Jul 23 15:33:28.363: ISDN Se0/0/1:23 Q931: RX <- STATUS pd = 8  callref = 0xC3FA 
            Cause i = 0x80AB28 - Access information discarded 
            Call State i = 0x01
    *Jul 23 15:33:28.411: ISDN Se0/0/1:23 Q931: RX <- CALL_PROC pd = 8  callref = 0xC3FA 
            Channel ID i = 0xA9838F 
                    Exclusive, Channel 15
    *Jul 23 15:33:28.415: ISDN Se0/0/1:23 Q931: RX <- PROGRESS pd = 8  callref = 0xC3FA 
            Cause i = 0x80FF - Interworking error; unspecified 
            Progress Ind i = 0x8088 - In-band info or appropriate now available 
    Thanks
    Richard

    Yes, Early media needs to be turned on the SIP Profile that is associated to the SIP trunk. The setting is SIP Rel 1xx options. It needs to be set for Send PRACK for all 1XX messages. If you have CUCM 7.x , this setting is a service parameter.

  • CUCM 10.5.1 and Exchange 2010 Unified Messaging (UM) SIP Trunk Problem

    This is more a comment if you're migrating from a lower version to 10.x.  Hopefully Google will pickup this post so others don't spend too much time (I got lucky and found this in about 30 minutes).
    There are many more SIP options than in the past.  If you configured your integration as per the integration doc, all settings are relevant, however there are some new defaults that need to change.
    SYMPTOM: Dialing another number or AA pilot and being redirected internally works, but the call drops on calls from external phones.  Exchange logs an Event ID 1079 from UMCore and also informational 1084 and 1172 events.  A capture yields a status 200 OK, then an ACK, two BYEs and a status 481 that the call leg does not exist.  The call is then dropped.
    RESOLUTION:
    In the SIP Trunk, modifiy the following:
    Device Information. . . Check Media Termination Point Required.  I've had some that have needed this checked and some where I needed to leave unchecked.  In this case going from 7.1 to 10.5 necessitated enabling.
    Call Routing Information. . . SIP Privacy:  Need to change from Default to None. 
    Any comments how the SIP Privacy might affect security and functionality would be appreciated.

    Just to inform everyone. I rebuilt the edge server from scratch. 
    Now everything is working as expected. 
    I cannot work out how the edge was not passing on calls to exchange or not communicating correctly with exchange. 
    Anyway it is resolved now. 

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