[cisco-voip] SIP Session Refresh During CUCM Upgrade
Folks: Wondering if anyone else has observed an issue where a CUCM node fails to send a SIP session refresh while an upgrade is in progress? I noticed this behavior when running through a mock upgrade for a customer. While upgrading the Publisher node, a call was active where the SIP dialog was established between the Publisher and CUBE. The active call dropped and looking at CCM traces showed no refresh was sent at half the Session Expires value. I lowered the Session Expires timer to recreate the issue in a shorter timeframe (SE of 240 seconds) and failed to see a refresh from CUCM after 120 seconds - the call again disconnected once the SE expired. Forcing the SIP dialog to source from the Subscriber node results in no issues and SIP refresh requests are sent as expected. Back to the Publisher, another call was placed after the upgrade completed and the issue cannot be recreated. With the Pub out the way, the subscriber's upgrade was started and the issue can now be recreated there as well. Basically, it seems the CM node in process of upgrading isn't sending a reINVITE at half the SE value. If this is expected behavior, I can make sure the customer's Session Expires value is temporarily increased during the upgrade window, but I can't seem to locate any documentation stating this is a known or expected caveat. This is going from 8.6.2.24090-1 to 8.6.2.25131-1 and I've yet to test this using other upgrade paths, but I plan on testing this once more once time permits in order to set proper expectations w/ the customer. Thanks! - Dan ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] CUCM 9+ and SIP trunk provider w/o CUBE
I am planning on replacing an again CallManager deployment with SCCP phones with a new CUCM 10.x and SIP phones. Our service provider can provide all of our trunking and PSTN connectivity via a direct SIP trunk, so I no longer need a PSTN gateway or PRI card, etc. (or at least I should not). We will be peering SIP from the CUCM to our providers SBC, so the question begs to ask on our side is a CUBE absolutely required to make this work? Pros/Cons? I'd rather not have to add a CUBE in if I don't have to. Since the phones are native SIP as is the CallManager I'm trying to understand why a CUBE is required? (or is it not?) TIA -Robert ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Possible to force SIP refer transfer from IP phone
I have a unique requirement between a new 10.5 CUCM and a third party dialer/ACD. CUCM is integrated to PSTN via SIP trunk to 2911 PRI gateway. CUCM is also integrated via SIP trunk to the dialer/ACD via SIP trunk directly between CUCM and the dialer/ACD. CUCM takes inbound call to IP phone. User needs to be able to transfer the external caller to the dialer/ACD. However we need the initial invite sent to the dialer to have the Calling Party Number set to the external PSTN user, so the dialer can properly do a database lookup of the external party and match the caller ID. When we do this transfer via IP phone, it always starts off as a warm transfer so the dialer/ACD gets the internal IP phone number for the DB lookup. Is there anyway for a IP phone (or jabber) to utilize SIP refer to complete the transfer so the dialer/ACD sees the external party number as caller ID on the transferred leg? As a secondary option, I have considered possibly delaying the actual transfer to the dialer/ACD to give the IP phone user enough time to complete the supervised transfer. But not sure if unity connection call handler is the best way or if there is another way, perhaps in CUCM dialplan. Thanks Justin ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCM 9+ and SIP trunk provider w/o CUBE
If your provider is acting as a SIP registrar you will certainly need a CUBE, and to be honest I'd recommend it anyway just so you don't have to allow UCM and phones direct connectivity outside your network. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/trunks.html#pgfId-1363704 When connecting to a service provider's IP PSTN network, Cisco strongly recommends the use of the Cisco Unified Border Element as an enterprise edge Session Border Controller to provide a controlled demarcation and security point between your enterprise and the service provider's network. -Ryan On Aug 12, 2014, at 7:41 PM, Robert Blayzor rblayzor.b...@inoc.net wrote: I am planning on replacing an again CallManager deployment with SCCP phones with a new CUCM 10.x and SIP phones. Our service provider can provide all of our trunking and PSTN connectivity via a direct SIP trunk, so I no longer need a PSTN gateway or PRI card, etc. (or at least I should not). We will be peering SIP from the CUCM to our providers SBC, so the question begs to ask on our side is a CUBE absolutely required to make this work? Pros/Cons? I'd rather not have to add a CUBE in if I don't have to. Since the phones are native SIP as is the CallManager I'm trying to understand why a CUBE is required? (or is it not?) TIA -Robert ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Possible to force SIP refer transfer from IP phone
Justin, Since these phones do not support blind transfer, it won't be possible to do something like that. Can you add a delay before checking the number so that it checks the connected number after the transfer is completed? Brian On Tue, Aug 12, 2014 at 7:12 PM, Justin Steinberg jsteinb...@gmail.com wrote: I have a unique requirement between a new 10.5 CUCM and a third party dialer/ACD. CUCM is integrated to PSTN via SIP trunk to 2911 PRI gateway. CUCM is also integrated via SIP trunk to the dialer/ACD via SIP trunk directly between CUCM and the dialer/ACD. CUCM takes inbound call to IP phone. User needs to be able to transfer the external caller to the dialer/ACD. However we need the initial invite sent to the dialer to have the Calling Party Number set to the external PSTN user, so the dialer can properly do a database lookup of the external party and match the caller ID. When we do this transfer via IP phone, it always starts off as a warm transfer so the dialer/ACD gets the internal IP phone number for the DB lookup. Is there anyway for a IP phone (or jabber) to utilize SIP refer to complete the transfer so the dialer/ACD sees the external party number as caller ID on the transferred leg? As a secondary option, I have considered possibly delaying the actual transfer to the dialer/ACD to give the IP phone user enough time to complete the supervised transfer. But not sure if unity connection call handler is the best way or if there is another way, perhaps in CUCM dialplan. Thanks Justin ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCM 9+ and SIP trunk provider w/o CUBE
I agree with Ryan on this one. If you do not have the CUBE, your provider will have basically have full IP connectivity into your network. Regards, Nathan Richie CCIE No. 27910 Consulting Engineer boice.net 700 Pearl Street/New Albany/ IN/ 47150 (502)271-2139 Single Number Reach (502)271-2100 24x7x365 Emergency Assistance On 8/12/14, 21:47, Ryan Ratliff (rratliff) rratl...@cisco.com wrote: If your provider is acting as a SIP registrar you will certainly need a CUBE, and to be honest I'd recommend it anyway just so you don't have to allow UCM and phones direct connectivity outside your network. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/coll ab10/trunks.html#pgfId-1363704 When connecting to a service provider's IP PSTN network, Cisco strongly recommends the use of the Cisco Unified Border Element as an enterprise edge Session Border Controller to provide a controlled demarcation and security point between your enterprise and the service provider's network. -Ryan On Aug 12, 2014, at 7:41 PM, Robert Blayzor rblayzor.b...@inoc.net wrote: I am planning on replacing an again CallManager deployment with SCCP phones with a new CUCM 10.x and SIP phones. Our service provider can provide all of our trunking and PSTN connectivity via a direct SIP trunk, so I no longer need a PSTN gateway or PRI card, etc. (or at least I should not). We will be peering SIP from the CUCM to our providers SBC, so the question begs to ask on our side is a CUBE absolutely required to make this work? Pros/Cons? I'd rather not have to add a CUBE in if I don't have to. Since the phones are native SIP as is the CallManager I'm trying to understand why a CUBE is required? (or is it not?) TIA -Robert ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip The information transmitted is intended only for the person or entity to which it is addressed and may contain CONFIDENTIAL material. If you receive this material/information in error, please contact the sender and delete or destroy the material/information. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
