SIP Redundancy Question

Hello,
I have set up 2 CUBEs with sip trunks to a provider from my CUCM 8.0.3. everything is working fine with exception of fall over to the second sip trunk in the route list. Meaning, if i bring the first CUBE down the second one never routes the outbound calls even though it is second in the route group. Does anyone know how to get this working properly? or...does this set up just not work? How can I get the call manager to notice that its first outbound choice died?
Thanks,
Ed Vazquez

Hi,
Wrong forum. You need to move this question to IP Telephony forum...
Regards
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MDGDP, CCNA, CCNA Voice certified

Similar Messages

UC500- Outbound SIP Routing Question

We have a UC540 that is receiving its trunking from a cloud based sip server. They have two trunks on the server, one for voice traffic and one for fax traffic. Inbound this works fine but all outbound traffic goes over the 'voice' trunk. Is it possible to route outbound traffic, from the fxs ports that the fax machines are connected to, onto the 'fax' trunk of the sip server? The sip server only provides one ip address, with unique registration to each trunk.

Hey guys,
Thanks for the response. Let me clarify my question a bit. I get the translation routing part and that is very helpful but I am still stuck on how to point a dial peer to the 'fax' trunk on the sip server.
A typical dial peer on the switch looks like this:
dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*Generic Locale*Long Distance**
translation-profile outgoing SIP-Trunk-Out
preference 1
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
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and the sip-ua looks like this:
sip-ua
credentials username 7xxxxx5 password xxxxx realm xxx.xxx.xxx.236 (voice trunk/user ID on the sip server)
credentials username 7xxxxx6 password xxxxx realm xxx.xxx.xxx.236 (fax trunk/user ID on the sip server)
keepalive target ipv4:xxx.xxx.xxx.236:5060
authentication username 7xxxxx5 password xxxxxx
no remote-party-id
retry invite 2
retry register 10
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timers keepalive active 100
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connection-reuse
host-registrar
Thanks,
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WLC Redundancy Question

Hi Community,
I have one question regarding the redundancy feature for two WLC 2106. We have installed on a customer site 2 WLC's 2106 and 6 AP's. Primary and Backup WLC works fine. No the customer asks the question if there is a option to copy the guest access wlan user from the primary controller to the backup controller? The lobby user activates a new user on the primary controller and if the primary controller fails, then the user did not get access to the wlan guest network.
Any ideas?
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Hi Jerry,
Mobility - it's used for client so that they can roam without any issue and one tunnle will be stablish between wlc-1 and wlc-2. if wlc-1 goes down the client will be local to the wlc-2 and if the guest user is not present on wlc-2 then it will not be able to access network.
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Primary SIP domain Question

Hi,
We’re in strange situation where our AD domain is abc.net but we want primary sip domain to be lync.abc.com & smtp domain is abc.com so email addresses are
[email protected]
So my question is, if I publish the topology for primary sip domain as
lync.abc.com then what could be our challenges in terms of client logon and lync access …? What do I have to have in place to make this work?
Note: Our Lync deployment will be two central sites and each will handle their own specific lync traffic.

You won't hit anything that would won't be able to overcome.
But what is the justification for knowingly making your SIP domain different from your SMTP name space? It's considered best practice to align the two in order to create a simplified user experience and minimise the amount of additional consideration (such
as exchange integration).
Fully appreciate you'll have your own circumstances though, and if this is a requirement - so be it =)
Kind regards
Ben
Note: If you find a post informative, please mark it so using the arrow to the left. If it answers a question you've asked, please mark the thread as answered to aid others when they're looking for solutions to similar problems or queries.
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Nexus Open Systems

IDSM placement and redundancy question

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I can't see anything in the config guide.
If I wanted to place a redundant pair on the outside of a pair of firewalls, how would I manage the redundancy of them.
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These are two IDSM-2s connected to slot four and give of the same chassis. We are running FWSM >> MSFC OUTSIDE setup. All InterVLAN traffic is evaluated first by the IDSM than by the FWSM. Users default gateway is the FWSM.
Here you go:
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intrusion-detection module 5 data-port 1 channel-group 5
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Regards
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Box-to-Box redundancy question ?

guys,
i have two CSS 11506 configured as Box-to-Box Master/Backup mode.
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associate rsakey
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please check the attached file that will clarify my point !!
please advice,
Thanks in advance

Dear Jose,
i have exported the files from CSS1 and importes them to CSS2,
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here you have the output that i got:
CSS1# commit_redundConfig "10.0.207.18 -d -s"
Checking the disk space locally before continuing with the script.
COMMIT_REDUNDANCY Version: 4.6
Verifying that ip redundancy is activated on Master switch.
Verifying that app session is up with backup switch.
Making sure app session is up.
Seconds to wait before calling it quits: 120
Checking the disk space remotely before continuing with the script.
Config Sync Failed.
CSS1# commit_redundConfig "10.0.207.18"
Checking available disk space on systems ...
Verifying app and redundancy configs ... /|-\-
Previous synchronization still occurring on remote switch.
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CSS redundancy question

Hello,
I have a query regarding box-to-box redundancy - What will be the status of circuits in redundant CSS? Currently I have couple of CSS configured for box-to-box redundancy. The primary works just fine, while all the circuits on secondary are showing disabled while the physical ports are up. Is this normal behaviour to have or do we have any problem here?
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Ranganath

Found a note in CSS redundant configuration guide under 'Configuring Redundant circuits' -
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Does that mean the circuit enabled for redundancy will be "down-ipDisabled" when acting as redundant?
Thanks.

SIP Trunk Question - Outbound Calls Fail

Hi Folks,
I am using a Cisco 2821 as a router that will convert a SIP trunk to an E1 PRI. Si my setup is:
SIp-Trunk > 2821 Router > E1 port on 3900> CUCM
Inbound calls are working fine, but outbound are failing. I am starting to think its due to transcoding issue on the SIP-GW maybe (there is nothing configured on it for XCODE etc).
I think my configuration is fine as I am able to recieve calls inbound. Just outbound fail.
Here are the debugs from the SIP-GW:
"Debug CCSIP calls"
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4BB4F194
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 1528xxxx
Called Number : 909
Source IP Address (Sig ): 172.29.x.xxx
Destn SIP Req Addr:Port : 10.200.7.157:5060
Destn SIP Resp Addr:Port : 10.200.7.157:5060
Destination Name : 10.200.7.157
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 172.29.5.210
Source IP Port (Media): 16786
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 28
Disconnect Cause (SIP) : 484
Can anyone shed some light on the area that I need to focus on? This is my first attempt at SIP and I am confused :)
Thanks.

Hi
909 is what I meant to dial as that is the help desk for the telco.
I tried mobile numbers as well getting the same error codes. And international numbers.
If it's based on the called number being wrong then I guess I will have to play with the calling party ID and call type as well... Maybe this is causing it to fail??

UCCX Redundancy questions

Dear NetPro gurus,
I'm a newbie to UCCX so would appreciated anyone's help on this.
My customer currently has 2 x UCCXs 7.0 servers (on the same LAN), and they are wondering what do they have to do to make the system 'redundant'.
All scripts, prompts and CM Telephony Triggers have been created and uploaded onto the Primary UCCX Server and everything is working. Do they have to manually "upload" all of these onto the Secondary UCCX Server as well? Or is the backend sync between two UCCX Servers?
If all the sync process is 'automatic' between 2 UCCX servers. My customer have tried to *power off* the primary UCCX Server, however, all of the CSQs dies. Is there anywhere i should get them to look to see why that is the case and how to fix?
Would greatly appreciated anyone helps on this.
Cheers,
Hunt

Hi
1. All the stuff you upload to UCCX actually goes into the 'repository', essentially a replicated DB between the two servers.
2. Aside from checking the replication status of the databases from AppAdmin, you also need to take a look at your scripts. It's very common for engineers to have not understood the system properly, and written (for example) XML files containing settings that the scripts need to the c:\ drive or wfavvid folder. These are not replicated, as they are local file system writes - the correct procedure is to use 'upload' steps to write them to the repository.
In terms of diagnosing this, you can go through the scripts which is easy if you know what you are looking for. The other option is during a maintenance period, fail over the system, and then do some reactive debugs of the scripts. Step through them, and see where they fail...
Aaron

SIP UUI Question

Hi,
Could somebody please advise as how would we retrieve the UUI (User to User Info) in ICM ? The UUI is being set at the customer PBX (Nortel) and getting passed on to ICM through the PG. I`m trying to retrieve that through either through UserToUserInfo variable ,with no luck.
Call flow:
Nortel PBX ---->SIP Trunk -----> Cisco CallManager ---->CTI Route Point----->CCM PG----->ICM
Raj

Yes, same trunk will be used for both inbound and outbound calls.
Let say you create one SIP trunk between CUCM and Asterisk, same SIP trunk here will be used to deliver all calls from CUCM to Asterisk and vice versa.
When you look at SIP trunk configuration in CUCM, you will get both inbound and outbound call related parameters for appropriate configuration of outbound and incoming call routing.
Thanks
Vivek

Single Server DHCP environment - redundancy question

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of our computers are lease refurbs and more than 4 years old. Additional hardware is not a solution).
I am going to move dhcp to router. It is small and cheap and I have a spare configured as a mirror of our production router. If my cheap but functional router dies, I can replace it in less than 60 seconds. If my server suffers failure,
I cannot replace it in 60 seconds.
I welcome other suggestions, or configuration changes of my network. We have had several physical infrastructure incidences over the last 18 months that have resulted in server failure, meaning we lack the access to the internet, meaning we cannot sell product,
meaning we do not make money to pay (my) salary. I understand ZERO down time comes at a cost. I'm trying to get the best reliability on the funds I have available.
Thanks

Thank you for not providing a single answer. WHY does moving DHCP from DC solve nothing? If server fails, workstations would still have access to internet, which allows my credit card processing, the problem I'm describing. So, what am
I missing that makes moving DHCP from the DC so bad?
How does a good backup solve my internet connection issue? I have a good backup system.
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IDH

Quorum Server Redundancy Question

Hi All,
I'm just investigating my options for a new cluster configuration and was trying to find out about multiple quorum servers hosts. All the examples I have come across in the documentation have 1 physical host acting as a quorum server for an n+1 node cluster. I'm ssuming that there will be quorum issues in the event the physical host, hosting the quorum server, is down and the cluster nodes performed a reconfiguration/switch when the quorum server was unavailable.
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Correct, the QS is only used if the cluster changes state, i.e. nodes leave or join. However, having more than 1 QS for a single cluster does not help. You simply lower your overall availability because there are more failure scenarios where one of these is down, leading to insufficient votes for the remaining cluster node to obtain.
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---

UC320 questions

Hello,
I have installed 2 UC320s. On my 2nd install the customer wants to be able to use an external transfer on the analog trunk. This requires a hook flash. I have not been able to figure out how to do this.
3 analog trunks
uc320 in blend mode
5 spa 504G sets
I have tried with direct line appearances and with no line appearances (dial 9 to access trunks). The spa 504 does not show a hook/flash button or key anywhere that I can find. Is the UC320 able to produce a hook/flash ?
also, is it possible for the auto attendant to produce a hook/flash.. for instance, customer wants after hours calls to transfer to an on call techs cell phone, can the UC320 produce a hook/flash in the instance to access the central office external transfer function on the analog trunk ? This is to not tie up 2 lines, but access the external transfer feature on the analog trunk itself
The customer has a 4th line (fax line) they would like this line to be accessible from the UC320 to make a 4th call only if the other 3 lines are busy. How do I setup the UC320 to call out on line 4 only when the other 3 lines are busy? I can not make 4 line appearances as the spa504 sets require one intercom set button.
Also, is it possible to have line appearances, and still have those line accessible from the dial out group when 9 is pressed?
Also, is it possible to break up the dial plan, for instance, have dial 8 go to analog trunks 1 or 2 and have dial 9 go to analog trunks 3 and 4 ?
I like the simplicity of the interface, however so far, it looks like the system is too dumbed down... please make an advanced tab or something that allows for dial plan manipulation (trunk call out order ect...), set VLANS, DSCP marks, cos etc... and other higher level functions. This thing works great for basic vannila stuff but if you need to do anything else good luck.
How do you check which voip encoding is being used, g.711 or g.729... can you force the phones to use one of the other? Does it support T.38... g.729 with VAD ?
Does the UC320 support Central office voice mail idication lamp on the analog trunks ? How about call waiting on the analog trunks... how do you do a hook/flash (redundant question, I know im just sort of ranting at this point)
Also, how do you create call restrictions.. for example, limit a lobby phone from dialing long distance, 900 #s, but still allow calling cards and 800 numbers ?
Any help would be great,
CTL Jake

Jake,
Let me categorize your questions and answer them
1. Hook Flash Support: Currently the UC320 / SPA Phone do not have any hook flash support or Centrex feature integration. Centrex capability is likely in the next release at the end of this year, however hook flash would still not be supported in the near term.
2. Fax Line: You could put a splitter on the 4th line and send one end to Fax and the other end to UC320. In the PBX mode ( 9+number) the 4th line would be selected last for outbound calls. On Shared FXOs offcourse it is user choice. They would need to be told not to use the Fax Line.
3. Simplicity: Yes this is intentional for a High Volume product. If you like to tweak things we would recommend proposing the UC500.
4. Codec: Yes preffered codec can be chose on the SIP trunk. T38 is supported.
5. MWI form CO: Again unsupported Centrex feature. In Key system mode you should be able to hear stutter dial tone.
6. Call Restriction: It is likely that this feature will be in our next major release at the end of this year.

Ethernet and WIFI Question

This maybe a redundant question, I want to back up the stuff in my Imac using TC. I do not want to use this Imac wireless nor do I want my backups done wireless but can I still use the wifi system on TC for Ipad ,Nook, etc. without having to configure my Imac everytime to use a wifi devise?

Yes. Just connect the iMac to your Time Capsule with an Ethernet cable, then go into System Preferences->Network and either set Ethernet to a higher priority than AirPort, or turn off AirPort completely.
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Hello.
We are currently using both Linux (RHEL5) and Solaris (5.10) for all of our Oracle 10.2.0.5.0 RAC DBs. Weare NOT using ASM currently. So, we are going to upgrade our DBs to the latest version of 11gR2 - and this obviously will include our CRS. I know the ASM and CRS under 10g are now combined under the 11gR2 Grid Infracstructure. I have been revieiwing the Oracle Grid Infrastructre Installation Guides for both OSs. While these guides are centered on new installations, I do notice that each has a small appendix tied to upgrading activities (Appendix E in Solaris and Appendix F in Linux).
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3.) Does anyone have any other additional document references/URLs that you might feel would be helpful in the overall 11gR2 upgrade process (both Infrastructure and/or DB)?
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Matt

Hi Matt;
Please check below notes:
Minimizing Downtime During Production Upgrade [ID 478308.1]
Master Note For Oracle Database Upgrades and Migrations [ID 1152016.1]
Different Upgrade Methods For Upgrading Your Database [ID 419550.1]
How To Upgrade ASM from 10.2 to 11.1 (RAC)? [ID 736127.1]
I suggest Please also check my blog
http://heliosguneserol.wordpress.com/2010/06/17/move-to-oracle-database-11g-release-2-wiht-mike-dietrich/
In this pdf you can see patch of to upgrade db from x to n wiht many senerios wiht all related metalinks notes which is created by Oracle worker Mike Dietrich
Regard
Helios

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