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- About dfokin
08-18-2017
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Member since
12-23-2005
83
Posts
5
Helpful
3
Solutions
Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
12-06-2012
09:12 AM
12-06-2012
09:12 AM
Well, it seems, I finally found the root cause of such behaviour. In document http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmsys/a03rp.html (It is for 8.6, but older documents are similar) is stated following about rerouting: When the system initially presents a call to a member of a route list, Cisco Unified Communications Manager reroutes for all cause codes other than Out of Bandwidth, User Busy, and Unallocated Number. The value of the associated service parameters for the Cisco CallManager service determines the rerouting decision for those cause codes. The Clusterwide Parameters (Route Plan) grouping includes the Stop Routing on Out of Bandwidth Flag, Stop Routing on User Busy Flag, and Stop Routing on Unallocated Number Flag service parameters. You can set each service parameter to True or False. After a route list locks onto a trunk, no rerouting occurs. The media connect time of the endpoints and the Stop Routing service parameters determine when a route list stops hunting for the next route group. When media negotiation begins, the route list or hunt list loses the ability to reroute. Since my trunks have "MTP requred" checked, the media path is established at the first stage of SIP negotiation. This blocks the ability to reroute.
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Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
12-06-2012
07:41 AM
12-06-2012
07:41 AM
Chris, We are working with mature, well-documented product. When documentation says: "Changing parameter "A" will affect behaviour "B"", it does not mean, that setting this parameter will affect also behaviour "X". I hope, you understand my idea. It's somewhat like trying to cure headache by using plaster bandage on the leg. Moreower, setting "Stop Routing on User Busy Flag" to False will increase unwanted trunk usage. When remote user is busy, we must try to call them later, but not right now, just using another trunk.
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Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
12-06-2012
07:07 AM
12-06-2012
07:07 AM
No, because they are unrelated to my situation. Let me explain: Stop Routing on Unallocated Number Flag: When the parameter is set to True and a call that is being routed to a remote Cisco cluster through a route list is released by a remote Cisco CallManager because of the unallocated number, a local Cisco CallManager will stop routing the call to a next device in the route list. When the parameter is set to False, the local Cisco CallManager will route the call to the next device. SIP message for Unallocated Number is "SIP/2.0 404 Not Found" Stop Routing on User Busy Flag: When the parameter is set to True and a call that is being routed to a remote Cisco cluster through a route list is released by a remote Cisco CallManager because a remote user (phone) is busy, a local Cisco CallManager will stop routing the call to the next device in the route list. When the parameter is set to False, the local Cisco CallManager will route the call to the next device. SIP message for User Busy is "SIP/2.0 486 User Busy" In my case I receive "SIP/2.0 503 Service Unavailable" message from trunk, that's why turning these setting to True will not help in my situation.
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Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
12-05-2012
11:54 PM
12-05-2012
11:54 PM
There is no CUBE. SIP link from provider is directly connected to CUCM.
... View more
Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
02-29-2012
04:14 AM
02-29-2012
04:14 AM
Hi, Barry and thank you for reply. As far as I know, this solution will envolve additional CPU resources on call processing nodes, and it must be avoided in situations with high call rate. May I accomplish this task using external HW, a 28xx router, for example?
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Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
02-29-2012
01:44 AM
02-29-2012
01:44 AM
Hi all! Currently we need to establish a VoIP link between our CUCM and CUCM of one of our partners. We plan to use an intercluster trunk, but there is one problem: our IP addressing overlaps with IP addressing of our partner, so there is a problem with RTP traffic. Is is possible to establish IP connectivity between two clusters without change of IP addressing? TIA.
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Labels:
11-28-2011
07:36 AM
Hi! Currently we are running CUCM 7.1.3 without DNS enabled and are quite happy with that. But suddenly we have decided to use SMTP for purposes of alerting. I have configured SMTP address on my servers, but they refused to send e-mails in lack of configured DNS addresses. I understand, that configuring DNS addresses on my servers will solve my e-mail problem, but also I am aware of db replication issues, when DNS is enabled. Also I am afraid, that my cluster will start use DNS names instead of addresses. Had anybody successfull experience in providing cluster with DNS addresses without enabling DNS?
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Labels:
Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
09-27-2011
10:41 PM
09-27-2011
10:41 PM
Hi! There is no way to put your own files to the internal web-server. On Linux-based CUCM appliances you have no direct access to the filesystem. Maybe your http server does not support javascript? What happens if you access your custom authentication URL using web browser?
... View more
Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
09-27-2011
07:03 AM
09-27-2011
07:03 AM
Hi, The file may be located on any web-server in your network. Of course, IP connectivity must exist between IP phone and this server. Also I need to say, that this authentication scheme is a big security issue. Anyone in your network will be able to see Screenshots of all your IP phones and send to them XML commands for pressing buttons and even establishing calls. I suggest you to re-define authentication URL only for testing purposes on per-phone basis.
... View more
Re: Dead air when connected to PSTN lin...
in IP Telephony and Phones
09-27-2011
06:39 AM
09-27-2011
06:39 AM
Hi, guys. Following commands will help you to avoid this issue: mgcp bind control source-interface voice_interface mgcp bind media source-interface voice_interface Just enter this commands after each use of "no mgcp/mgcp" combination.
... View more
Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
09-27-2011
06:13 AM
09-27-2011
06:13 AM
To bypass phone authentication you need to re-define "URL Authentication" parameter in Enterprise Parameters Configuration. This parameter must point to the URL, which answers with "AUTHORIZED" string to any HTTP rquest. For example, you may create .jsp file with following content: response.addHeader("Content-Type", "text/html"); out.print("AUTHORIZED");
... View more
Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
09-27-2011
04:35 AM
09-27-2011
04:35 AM
Hi! I had same problem. Restarting TFTP service is not enough for CUCM to start recognize 3905 phones. You need to reload your cluster to allow them to register. I have tested this solution in my lab environment. The phone has successfully registered after CUCM reload. cin-sip.jar is only responsible for the localization of phone's user interface and the ability to register to the cucm does not depend on the existence of this file on the TFTP server.
... View more
05-26-2011
12:00 PM
Hi! Currenly we are migrating from CUCM 5.0 to 7.1.5. 1200 phones are spread across 20 WAN sites, each site is connected to central site with 2Mbps link. The cluster is located at the central site. Central site has 20 Mbps link to the WAN. During migration process the "Load Information" parameter in "Device Defaults" section was cleared for certain phone models to prevent FW upgrade issues during migration pocess. All phones currently are provisioned with FW version 8.3.2 and registered on the new cluster. I need to upgrade phone FW to most current 9.0.2.SR1 version.To accomplish this, I have developed few scenarios of upgrade process and i want to discuss with you most perfect (in my opinion, of course) scenario: Since I am unable to do direct upgrade from 8.3.2 to 9.0.2, I need to perform interim upgrade to the 8.5.2 version of FW. To minimize WAN link usage, I will use "Peer firmware sharing" option. Each site is associated with the Device Pool. Further steps are: 1. Upload to all TFTP servers in cluster 8.5.2 FW bundles for my 7911 and 7941 phones 2. Set "Load Information" option in "Device Defaults" to 8.5.2 3. For each device pool: 3.1 Enable "Peer Firmware Sharing" option for all phones 3.2 Reset phones 4. Set "Load Information" option in "Device Defaults" to 9.0.2.SR1 5. For each device pool: 5.1 Reset phones For the first look all things are easy, but when I tried to test this scenario in the lab, I have met very discouraging issue: when the phone upgrades its firmware from version 8.3.2 to 8.5.2, it does it TWICE. After the step 3.2 of my scenario the phones are rebooting, one of them loads FW files from the TFTP server and distributes FW files across their neighbors. They all are updating their firmwares and after update they all are rebooting, but instead of further booting, they all are upgrading their firmwares one more time. This time they do not use peer firmware sharing (since they are not configured with this option) and all phones are requesting FW files directly from the central TFTP server. I think, this is very bad, because it is real pain, to distribute phone firmware through WAN link for 60-70 phones at the same time with TFTP.To avoid this mess, I have developed following plan: 1. Upload 8.5.2 FW bundles to the TFTP server at each site (I have used Cisco router) 2. Update DHCP option 150 at each site to point to the local TFTP server 3. Reset all phones at the site 4. Monitor TFTP server events to understand, that the firmware upgrade process is finished 5. Set DHCP option 150 back to the addresses of TFTP servers at the central site 6. After that phones with FW version 8.5.2 will be registered on CUCM and I will perform step 5 from my original plan. I have tested this plan in my lab and it worked, but I think, this procedure is very complex. Maybe there is more easy way to upgrade 8.3.2 to 8.5.2? Thank you for reading! =)
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Labels:
Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
04-20-2011
05:34 AM
04-20-2011
05:34 AM
I have met once this problem. The phone with this problem was in a remote site and CUCM server was at the central site. The cause of the problem was unreliable link between HO and remote site without QoS support. Some packets were dropped or lost in transit -- so the pone did not received a "StopTone" SCCP message. After implementing QoS on the link and increasing bandwidth between sites the problem disappeared.
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Created by
dfokin
in IP Telephony and Phones
in IP Telephony and Phones
04-14-2011
07:18 AM
04-14-2011
07:18 AM
I think, you may achieve needed result in two steps: 1. Stream your audio through your network using multicast and g.711 codec 2. Create IP phone application, which will connect to this stream and play it using headset (RTPMRx function in IP Phone services Development guide) The person, who need audio must pick up handset, press application button and voila!
... View more
Created by
dfokin
in IP Telephony and Phones
12-06-2012
12-06-2012
Well, it seems, I finally found the root cause of such behaviour.In
document
http://www.cisco.com/en...
Created by
dfokin
in IP Telephony and Phones
12-06-2012
12-06-2012
Chris,We are working with mature, well-documented product. When
documentation says: "Changing parame...
Created by
dfokin
in IP Telephony and Phones
12-06-2012
12-06-2012
No, because they are unrelated to my situation.Let me explain:Stop
Routing on Unallocated Number Fla...
Created by
dfokin
in IP Telephony and Phones
12-05-2012
12-05-2012
There is no CUBE. SIP link from provider is directly connected to CUCM.
Created by
dfokin
in IP Telephony and Phones
02-29-2012
02-29-2012
Hi, Barry and thank you for reply.As far as I know, this solution will
envolve additional CPU resour...
Created by
dfokin
in IP Telephony and Phones
02-29-2012
02-29-2012
Hi all!Currently we need to establish a VoIP link between our CUCM and
CUCM of one of our partners. ...
11-28-2011
Hi!Currently we are running CUCM 7.1.3 without DNS enabled and are quite
happy with that. But sudden...
09-27-2011
Hi!There is no way to put your own files to the internal web-server. On
Linux-based CUCM appliances ...
09-27-2011
Hi,The file may be located on any web-server in your network. Of course,
IP connectivity must exist ...
09-27-2011
Hi, guys. Following commands will help you to avoid this issue:mgcp bind
control source-interface vo...
09-27-2011
To bypass phone authentication you need to re-define "URL
Authentication" parameter in Enterprise Pa...
Created by
dfokin
in IP Telephony and Phones
09-27-2011
09-27-2011
Hi!I had same problem. Restarting TFTP service is not enough for CUCM to
start recognize 3905 phones...
05-26-2011
Hi!Currenly we are migrating from CUCM 5.0 to 7.1.5. 1200 phones are
spread across 20 WAN sites, eac...
04-20-2011
I have met once this problem. The phone with this problem was in a
remote site and CUCM server was a...
Created by
dfokin
in IP Telephony and Phones
04-14-2011
04-14-2011
I think, you may achieve needed result in two steps:1. Stream your audio
through your network using ...
Public Statistics
| Date Registered | 12-23-2005 08:16 AM |
| Date Last Visited | 08-18-2017 03:56 AM |
| Total Messages Posted | 83 |
| Total Helpful Votes Received | 5 |
