Re: [OSL | CCIE_Voice] Gatekeeper trunk
Try to add: ! voice service voip sip midcall-signaling passthru ! Peter - Original Message - From: Mohd Baqari baqari.voic...@gmail.com To: san r luv...@gmail.com Cc: ccie_voice@onlinestudylist.com; The Masterplan winmasterp...@gmail.com Sent: Wednesday, May 23, 2012 9:15 PM Subject: Re: [OSL | CCIE_Voice] Gatekeeper trunk Hi, If the command emptycapability then it has to work assuming that u kept gk in media flow through mode. Plz share the output of debug ipipgw on cube Regards, Mohammed Al Baqari Sent from my iPhone On May 23, 2012, at 7:50 PM, san r luv...@gmail.com wrote: emptycapability ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CUCM auto-register questions
Hi all, I have some q about auto-registering in CUCM: - If default auto-registering protocol is changed than the whole cluster is needed to be restarted or only the node on which this function is enabled? Ex.sub obly. - After migrating a phone from SCCP to SIP via BAT enabling auto-register SIP is mandatory to register the phone as SIP device instead of rejecting the registration? - If question states not to use auto-registering function but need to migrate SCCP to SIP than what to do? Thanks, Peter___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME busy-trigger-button Problems
In CME busy-trigger is the same common value for the shared line. Let it be only one then if either instance serve a call it does not hunt for an idle. This differs in UCM where all line is independent in case of busy-trigger. Peter - Original Message - From: Roig Borrell, Francesc Xavier To: ccie_voice@onlinestudylist.com Sent: Thursday, April 21, 2011 9:36 PM Subject: [OSL | CCIE_Voice] CME busy-trigger-button Problems Hi all!! I am having a behavior that It does not make any sense with this easy scenario ephone-dn 11 octo-line number 4500 ephone 1 mac-address 0024.97AA.1B49 busy-trigger-per-button 1 type 7945 button 1:11 ephone 1 mac-address 0024.14B3.765E busy-trigger-per-button 1 type 7965 button 1:11 First call to 4500, phone 1 and 2 rings, answer one of them o Second call to 4500 gives busy !! Why? It is a shared octo-line and with this config I should expect a second incoming call rings the idle phone, shouldn't it?! If I change to ephone 1 mac-address 0024.97AA.1B49 busy-trigger-per-button 2 type 7945 button 1:11 ephone 1 mac-address 0024.14B3.765E busy-trigger-per-button 1 type 7965 button 1:11 I can have only 2 incoming calls to 4500. It seems like the bigger busy-trigger-per-button value configured always triggers the busy What am I missing? I have spent several hours troubleshooting this and I can't understand Thanks in advance!! Francesc -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Simple transfer to VM issue
also set 'transfer-pattern .T' under telephony-service to allow transfer to VM - Original Message - From: Phillip Day To: ccie_voice@onlinestudylist.com Sent: Wednesday, June 23, 2010 12:13 PM Subject: [OSL | CCIE_Voice] Simple transfer to VM issue Has anyone got a sensible idea for this issue? its really simple - I can't transfer to a CUE VM pilot number from any phone on a CUCME system. The first though for me was the IP-IP GW config, it does allow H323 to SIP and vice versa, so I'm sure thats not the fault. I can dial directly to the VM pilot from any registered phone but can't transfer to it from anywhere. There is no COR on the dial-peer for the CUE module. I have set the max redirects to a high number, but now I'm rather clutching at straws! any ideas?? Thanks in advance Phill Day __ This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This footnote also confirms that this email message has been swept by a content checking tool for the presence of computer viruses. Nettitude Limited is a Company registered in England Registered Address Nettitude Limited, Fosse House, Fosse Way, Leamington Spa, Warwickshire, CV31 1XG Company Registration Number: 4705154 VAT Number: 812 4539 44 www.nettitude.com __ -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] voice mail port AAR CSS
I get the sccp service down on HQ which is an RSVP agent. After this when the CUC would reach (by MWI or transfer) a phone at BR1 AAR is activated. In my opinion AAR CSS of a VM port has to take effect in this case. However this CSS is None here but MWI still works. In addition a call can be transfered from AA at CUC to a phone at BR1. All DNs at BR1 site are in pt-internal partition. Any suggestion? Or what is the function of AAR CSS of VM port?___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Fw: how to forward calls from UCM to an Auto Attendant at CUC
Any consideration why the recommended method is using a CTI port instead of a hunt pilot? If a hunt pilot is configured for AA then a user sees the call as a directed rather than forwarded which happens in case of CTI. - Original Message - From: Peter Farkas To: ccie list Sent: Monday, June 14, 2010 2:56 PM Subject: [OSL | CCIE_Voice] how to forward calls from UCM to an Auto Attendant at CUC Supposed method to call in a CUC from UCM is via a CTI RP which is forwarded to the Voicemail. However my solution was to set up a new hunt pilot that points to the same hunt list that was created for VM. I've also created a new route to the AA's Call handler in CUC Direct Call Routing Rules table based on dialed number parameter. I'd just to know if my soultion is still acceptable or what is the CTI RP's benefit over? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE caller id
You can record a spoken name via AvT menu. It works for me. - Original Message - From: Angel Perez To: osl osl Sent: Tuesday, June 15, 2010 2:23 PM Subject: [OSL | CCIE_Voice] CUE caller id Hi: I was trying to setup CUE to say voicemail user name instead of phone number when somebody left a message at voicemail, (like in CUC) but the most i can do is just to hear phone number (voicemail callerid), after some tests my conclusions is that it is not possible Anybody has tried this? Regards -- Hotmail: Trusted email with powerful SPAM protection. Sign up now. -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] how to forward calls from UCM to an Auto Attendant at CUC
Supposed method to call in a CUC from UCM is via a CTI RP which is forwarded to the Voicemail. However my solution was to set up a new hunt pilot that points to the same hunt list that was created for VM. I've also created a new route to the AA's Call handler in CUC Direct Call Routing Rules table based on dialed number parameter. I'd just to know if my soultion is still acceptable or what is the CTI RP's benefit over?___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] show vlan-s brief
I also cheked mine. Further interesting these different output of the same
config.
BR2-RTR#sh vlan-switch
VLAN Name StatusPorts
- ---
1default activeFa0/1/2, Fa0/1/3
200 DATA active
400 PHONES active
1002 fddi-default act/unsup
1003 token-ring-default act/unsup
1004 fddinet-default act/unsup
1005 trnet-defaultact/unsup
VLAN Type SAID MTU Parent RingNo BridgeNo Stp BrdgMode Trans1 Trans2
- -- - -- -- -- --
1enet 11 1500 - - ---1002 1003
200 enet 100200 1500 - - ---0 0
400 enet 100400 1500 - - ---0 0
1002 fddi 101002 1500 - - ---1 1003
1003 tr101003 1500 1005 0 --srb 1 1002
1004 fdnet 101004 1500 - - 1ibm -0 0
1005 trnet 101005 1500 - - 1ibm -0 0
BR2-RTR#sh run int Fas 0/1/0
Building configuration...
Current configuration : 119 bytes
!
interface FastEthernet0/1/0
switchport trunk native vlan 200
switchport mode trunk
switchport voice vlan 400
end
BR2-RTR#sh vlan-switch id 400
VLAN Name StatusPorts
- ---
400 PHONES activeFa0/1/0, Fa0/1/1
VLAN Type SAID MTU Parent RingNo BridgeNo Stp BrdgMode Trans1 Trans2
- -- - -- -- -- --
400 enet 100400 1500 - - ---0 0
- Original Message -
From: Angel Perez
To: osl osl
Sent: Wednesday, June 02, 2010 7:53 PM
Subject: [OSL | CCIE_Voice] show vlan-s brief
Hi:
When I configure the swich port of my hwic-esw with the old method:
interface range fas 0/3/0 - 3
swicht mode trunk
swicht trunk encap dot1q native vlan 200
swicht voice 300
I get the following result:
sh vlan-s bri
VLAN Name StatusPorts
-
---
1default activeFa0/3/1, Fa0/3/3
300 voiceactiveFa0/3/1, Fa0/3/3
200 data active
Everything works as expected, it is just a problem in the show comand, but I
wonder if the proctor wants to check the vlans with this command he/she could
think that it is wrong...
With the new method:
switch mode acc
switch acc vlan 200
swith voice vlan 300
I get this output, that looks better:
sh vlan-s bri
VLAN Name StatusPorts
-
---
1default active
300 voiceactiveFa0/3/1, Fa0/3/2, Fa0/3/3
200 data activeFa0/3/1, Fa0/3/2, Fa0/3/3
What do you think about it?
Thanks
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Re: [OSL | CCIE_Voice] show vlan-s brief
In case of ESW the suggested method is trunk mode.
http://www.cisco.com/en/US/docs/ios/lanswitch/configuration/guide/lsw_hwic_ethsw_ic_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1051730
- Original Message -
From: Peter Farkas
To: Angel Perez
Cc: osl osl
Sent: Wednesday, June 02, 2010 8:36 PM
Subject: Re: [OSL | CCIE_Voice] show vlan-s brief
I also cheked mine. Further interesting these different output of the same
config.
BR2-RTR#sh vlan-switch
VLAN Name StatusPorts
-
---
1default activeFa0/1/2, Fa0/1/3
200 DATA active
400 PHONES active
1002 fddi-default act/unsup
1003 token-ring-default act/unsup
1004 fddinet-default act/unsup
1005 trnet-defaultact/unsup
VLAN Type SAID MTU Parent RingNo BridgeNo Stp BrdgMode Trans1 Trans2
- -- - -- -- -- --
1enet 11 1500 - - ---1002 1003
200 enet 100200 1500 - - ---0 0
400 enet 100400 1500 - - ---0 0
1002 fddi 101002 1500 - - ---1 1003
1003 tr101003 1500 1005 0 --srb 1 1002
1004 fdnet 101004 1500 - - 1ibm -0 0
1005 trnet 101005 1500 - - 1ibm -0 0
BR2-RTR#sh run int Fas 0/1/0
Building configuration...
Current configuration : 119 bytes
!
interface FastEthernet0/1/0
switchport trunk native vlan 200
switchport mode trunk
switchport voice vlan 400
end
BR2-RTR#sh vlan-switch id 400
VLAN Name StatusPorts
-
---
400 PHONES activeFa0/1/0, Fa0/1/1
VLAN Type SAID MTU Parent RingNo BridgeNo Stp BrdgMode Trans1 Trans2
- -- - -- -- -- --
400 enet 100400 1500 - - ---0 0
- Original Message -
From: Angel Perez
To: osl osl
Sent: Wednesday, June 02, 2010 7:53 PM
Subject: [OSL | CCIE_Voice] show vlan-s brief
Hi:
When I configure the swich port of my hwic-esw with the old method:
interface range fas 0/3/0 - 3
swicht mode trunk
swicht trunk encap dot1q native vlan 200
swicht voice 300
I get the following result:
sh vlan-s bri
VLAN Name StatusPorts
-
---
1default activeFa0/3/1, Fa0/3/3
300 voiceactiveFa0/3/1, Fa0/3/3
200 data active
Everything works as expected, it is just a problem in the show comand, but
I wonder if the proctor wants to check the vlans with this command he/she could
think that it is wrong...
With the new method:
switch mode acc
switch acc vlan 200
swith voice vlan 300
I get this output, that looks better:
sh vlan-s bri
VLAN Name StatusPorts
-
---
1default active
300 voiceactiveFa0/3/1, Fa0/3/2, Fa0/3/3
200 data activeFa0/3/1, Fa0/3/2, Fa0/3/3
What do you think about it?
Thanks
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[OSL | CCIE_Voice] Conference by Select/Join (SIP)
Conference is supported at HQ site but CIPC(SIP) IP phone cannot use Select/Join softkey to bulid up a conference. It failes with Unavailable Feature message on the display. However Confrn softkey works as expected. CIPC or SIP supports buliding a conference by Select/Join method, at all?___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] COR List - SRST
If u do not attach any incoming COR to the SB2, as can be seen in the config, then it can access all dial-peers. For this reason it is a good practice to config a default incoming cor under call-manager-fallback to restrict. SB1 has correct configuration, it also can access dial-peer 9011. - Original Message - From: Ashar Siddiqui To: ccie_voice@onlinestudylist.com Sent: Monday, May 31, 2010 7:40 PM Subject: [OSL | CCIE_Voice] COR List - SRST Hello all, Help needed in COR Requirement: There are two phones at Site B, SB2 cannot make International calls while SB1 can. Both can make rest of the calls. I configured following COR, seems to be not working. SB2 can still make international even thou you can see an outgoing COR. To be honest, I am not good at COR and kind of hate it as well :) ! dial-peer cor custom name International ! ! dial-peer cor list InternationalCalls member International ! dial-peer cor list SB1 member International ! dial-peer voice 9011 pots corlist outgoing InternationalCalls translation-profile outgoing int-pstn destination-pattern 9011T port 0/0/0:23 prefix 011 ! call-manager-fallback cor incoming SB1 1 3001 Ash -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] COR List - SRST
If u do not attach any outgoing cor list to the dial-peer then all caller can use this dial-peer without any restriction. Right, it can be the easiest way if u configure corlist on the dial-peers on that you would like to have some restriction. But keep in mind that incoming COR also have to be configured on DNs. Ex. DP for 911 shouldn't have any corlist since all caller have to call this number. However a more secure way can be configuring out corlist for each dial-peer, default incoming for SRST and incoming for DNs that are not default. two comms: here, SB1 can only access international but not 911. :( the right syntax under call-manager-fallback: cor incoming corlist, here it is default default I hope it helps. - Original Message - From: Ashar Siddiqui To: Peter Farkas Cc: ccie_voice@onlinestudylist.com Sent: Monday, May 31, 2010 8:19 PM Subject: Re: [OSL | CCIE_Voice] COR List - SRST Thanks Peter. That would mean I will have to configure each one them under cor custom and then assign each one of them under each dial-peer...isnt't it? Somewhere I read that if you don't want to restrict access to specific dial-peer then you don't need outgoing cor on that dial-peer. But, as you are saying I will have to create a default incoming COR that means I will have to configure each dial peer with an outgoing COR as well? Correct me if I am wrong? This is what I understood .. dial-peer cor custom name emergency name local name national name International then... dial-peer cor list EmPt member emergency dial-peer cor list LocPt member local dial-peer cor list nationalPt member national dial-peer cor list InternationalCalls member International then in each dial-peer something like.. corlist outgoing InternationalCalls corlist outgoing Empt corlist outgoing LocPt etc.. and then dial-peer cor list default member emergency member local member national ! dial-peer cor list SB1 member International ! and at call-manager-fallback call-manager-fallback cor incoming default cor incoming SB1 2 3001 Instead of doing all this just to block access of International number from SB2, I am looking into what Vik Suggested.. Ash Peter Farkas wrote: If u do not attach any incoming COR to the SB2, as can be seen in the config, then it can access all dial-peers. For this reason it is a good practice to config a default incoming cor under call-manager-fallback to restrict. SB1 has correct configuration, it also can access dial-peer 9011. - Original Message - From: Ashar Siddiqui To: ccie_voice@onlinestudylist.com Sent: Monday, May 31, 2010 7:40 PM Subject: [OSL | CCIE_Voice] COR List - SRST Hello all, Help needed in COR Requirement: There are two phones at Site B, SB2 cannot make International calls while SB1 can. Both can make rest of the calls. I configured following COR, seems to be not working. SB2 can still make international even thou you can see an outgoing COR. To be honest, I am not good at COR and kind of hate it as well :) ! dial-peer cor custom name International ! ! dial-peer cor list InternationalCalls member International ! dial-peer cor list SB1 member International ! dial-peer voice 9011 pots corlist outgoing InternationalCalls translation-profile outgoing int-pstn destination-pattern 9011T port 0/0/0:23 prefix 011 ! call-manager-fallback cor incoming SB1 1 3001 Ash -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question
Display ID of RDP's DN is missing. When shared line is created then only the Alerting Name is copied from the line. Go to the DN Configuration of 5002 and select the RDP from Associated Devices list and use Edit Line Appaearance button to modify. - Original Message - From: Matthew Berry To: OSL Group Sent: Wednesday, May 26, 2010 3:11 AM Subject: [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question Fellow nerds, I am battling a single number reach (i.e. Mobile Connect) question on Lab 4. Question 3.1 says the call should appear to BR1 Phone 2 as if it is actually coming from HQ Phone 2 directly (Calling Name and Number). When I call in from the PSTN phone to BR1 Phone 2, the display on BR1 Phone 2 shows 5002. The calling number is represented just fine. However, I cannot get the calling nmae to be presented on the display. I have tinkered around with the partial/complete match and significant digits parameters under the mobility section of the Call Manager service parameters but nothing has changed. Any ideas? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Version UC Applications for CCIE Voice Lab
From Ben at ask the expert forum: Please see answers inline: 1-The version of Unified applications (Communication Manager, Presence, Unity Connection, IPCCX) is 7.0 or it is the latest service release. Ben: 7.0(1) 4-CUPC and CIPC Versions. Ben: 7.0(1) CUCME 7.0 - Original Message - From: Kevin Damisch kevin.dami...@vitalsite.com To: Alexis Muñoz M amuno...@hotmail.com; ccie_voice@onlinestudylist.com; alexis.mu...@telefonica.com Sent: Tuesday, May 25, 2010 4:39 AM Subject: Re: [OSL | CCIE_Voice] Version UC Applications for CCIE Voice Lab The last I heard (and saw for myself) was the early 7.0 releases. Does anyone know if 7.1 has been let loose in the lab yet? From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alexis Muñoz M [amuno...@hotmail.com] Sent: Monday, May 24, 2010 5:03 PM To: ccie_voice@onlinestudylist.com; alexis.mu...@telefonica.com Subject: [OSL | CCIE_Voice] Version UC Applications for CCIE Voice Lab Hello Somebody could share me which version of UC Applications included CME and also CUE i need to use for preparing for CCIE Voice Lab v3.0. I would appreciate the help. Thanks, Alexis Munoz Networking IP Telephony Engineer Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it!http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cannot dial from MVA
Thank you for the link however my case is different a bit. Finally I could step over by looking sdi traces in depth: H.323 gw put '+1' at the begining of the ANI to be in E.164 format but the RD was defined without that. The behaviour was strange to me since Mobile Connect works as expected so RD is reachable in this case. In the other hand a call from the remote destination can succesfully authenticate. I have a question: Mobile Connect use the original ANI received from the H.323 GW without using Incoming Calling Party Settings at gw level to match RD? - Original Message - From: Angel Perez To: wormh...@sch.hu ; osl osl Sent: Wednesday, May 19, 2010 9:10 AM Subject: RE: [OSL | CCIE_Voice] cannot dial from MVA Hi, check this topic: http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16572.html hth -- From: wormh...@sch.hu To: ccie_voice@onlinestudylist.com Date: Tue, 18 May 2010 20:24:30 +0200 Subject: [OSL | CCIE_Voice] cannot dial from MVA Gents, I have an issue with MVA. MVA collects PIN and I press 1 to dial but it does not proceed with any call instead the well known prompt sounds: The call cannot be completed... Even if the called number is local and placed in the None partition. This prompt suggests CSS issue however as Vik advised before I created a totally new CSS just for RDP but it does not solve the problem. Service Parameters: Complete Match and RDP+Line CSS. I have read near all the thread regarding MVA here, but the issue remains. I attached the vxml debug. Any suggestion? -- Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] incoming VoIP dial-peer matching not based on session protocol
Hi, I'm a bit confused why the session protocol command is not necessary for an incoming VoIP dial-peer. Even if it is H.323 dial-peer (session protocol cisco) a SIP call can match. Any ANI/DNIS matching rule is enough. Sb can makes me clear? thx P___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] incoming VoIP dial-peer matching not based onsession protocol
I was missed that there is no need to apply the 'session protocol cisco' command at all since it is the default. Can be checked by show dial-peer. - Original Message - From: Paul Kruger To: Peter Farkas Cc: ccie list Sent: Monday, May 03, 2010 2:05 PM Subject: Re: [OSL | CCIE_Voice] incoming VoIP dial-peer matching not based onsession protocol Not 100% sure, but think it is only applicable for outbound, and inbound will match on matched digits. From the Cisco IOS Voice Command Reference: session protocol cisco - The cisco keyword is applicable only to VoIP on the Cisco 1750, Cisco 1751, Cisco 3600 series, and Cisco 7200 series routers. 2010/5/3 Peter Farkas wormh...@sch.hu Hi, I'm a bit confused why the session protocol command is not necessary for an incoming VoIP dial-peer. Even if it is H.323 dial-peer (session protocol cisco) a SIP call can match. Any ANI/DNIS matching rule is enough. Sb can makes me clear? thx P ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
