Re: [OSL | CCIE_Voice] Route Patterns
Not sure what you would like explained. 9011.! and 9011.!# are for international dialing from North America. Both patterns include a dot (.) which will allow you to apply a digit transform action of pre-dot, if you want. The exclamation (!) is a wild card and instructs the digit analysis process to continue accepting dialed digits (0-9,*). The hash is treated as a termination character. Since ! says I'll keep taking digits until inter-digit timeout expires you sometimes want to provide a way for users to expedite the digit analysis. The # gets you there and you need a route pattern to handle that digit. The 9.1(2-9)xx[2-9]xx is an invalid pattern. Parens are not acceptable characters for route patterns. -Bill -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Oct 10, 2013, at 9:30 PM, Anthony Nwachukwu wrote: Hi, Can someone explain the Route Patterns below. 9.1(2-9)XX[2-9]XX 9011.! 9011.!# 9.1(2-9)XX[2-9]XX Cheers ___ 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] MVA confusion and quesiton
hi All, Thanks for your responses. One of the things I noticed was ifnbsp; I do not manipulate the calling number from 525 to 9525nbsp; then while placing a call directlynbsp; from the pstn line 525 tonbsp; SBPh1nbsp; goes to voicemail or fails . Is this expected behavior? -Ramesh From: Bill Lake lt;whl...@gmail.comgt; Sent: Mon, 07 Oct 2013 18:17:34 To: Martin Sloan lt;martinsloa...@gmail.comgt; Cc: ramesh lt;rameshdol...@rediffmail.comgt;, ccie_voice lt;ccie_voice@onlinestudylist.comgt; Subject: Re: [OSL | CCIE_Voice] MVA confusion and quesiton On Number 2 you should remember in 7.0.x there is a bug with partial match and you might not want to risk hitting it Bill Sent from my iPhone On Oct 7, 2013, at 7:43 AM, Martin Sloan lt;martinsloa...@gmail.comgt; wrote: Hi Ramesh, Here's some answers based on my approach to configuring MVA. 1)nbsp; I would use the 4 digit number for my dial-peer and CUCM MVA number (3300).nbsp; Since you probably already have a translation-profile in place on the voice port or inbound dial-peer to chop the called number down to 4 digits, it makes sense to use that. 2) I don't change the calling party number and I use 'complete match' on the service parameter.nbsp; I set my remote destination to that full number (either 7 or 10 digits). 3) No manipulation required, just set the remote destination to the full number. Marty On Sun, Oct 6, 2013 at 9:52 AM, ramesh lt;rameshdol...@rediffmail.comgt; wrote: Hi San, Thanks for your reply. 1) So you're suggestion is to use 3300 or 3033300 ? 2)At the dial-peer level are you using 3300 or 3033300?nbsp; I way I use it is as given below : - = (a) If I use 3300 at the dial-peer levelnbsp; and on the callmanger as MVA numbernbsp; with 525 as the calling party numbernbsp; then Inbsp; amnbsp; ablenbsp; to havenbsp; MVA functionality .nbsp; (b)nbsp; Inbsp; normally call from the pstn using 3033300 from line 525( pstn phone)nbsp; thennbsp;nbsp; on my h323 gatewaynbsp; I strip the called number to last 4 digits and send to the callmanger .nbsp; (c) On the callmanger my MVA number is 3300. Arenbsp; the above steps ( a to b )nbsp; correct? Regards, Ramesh From: san r lt;luv...@gmail.comgt; Sent: Sun, 06 Oct 2013 13:37:52 To: ramesh lt;rameshdol...@rediffmail.comgt; Subject: Re: [OSL | CCIE_Voice] MVA confusion and quesiton if you're stripping number for MVA , then mostly it wont work. Should use the exactly same number in Dial peer amp; and CCM MVA configurations. I had the same issue in lab On Sat, Oct 5, 2013 at 8:12 PM, ramesh lt;rameshdol...@rediffmail.comgt; wrote: Hello Guys, I have the following questions for MVA. 1) I am followingnbsp; a 4 digitnbsp; internal dial-plan fornbsp; my site B phonesnbsp; and there is a requirement that I usenbsp; 3033300nbsp; ( 7 digit number )nbsp; as my MVA numbernbsp;nbsp; thennbsp; can strip this 7 digit number tonbsp; the last 4 digitnbsp; number (nbsp; 3300 ) as my MVA number ? 2) Alsonbsp; my calling number isnbsp; a 7 digit number coming from pstnnbsp; as 525nbsp; then do I change it to 9525? 3) If incase calling number isnbsp; a 10 digit numbernbsp; then It would come into site B as 972525 ( which is 10 digits) is manipulation required for this or can I just use the complete match with 10 digits on the service parameter level? -Ramesh Dollar Get your own FREE website, FREE domain amp; FREE mobile app with Company email. nbsp; Know More gt; ___ 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 Get your own FREE website, FREE domain amp; FREE mobile app with Company email. nbsp; Know More gt; ___ 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 ___ 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] Voice Lab routing issue
Site A router f0/0 is connected to switch port 1/0/1 right? The switch has the native vlan set but the router does not. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Dane Warner Sent: Friday, October 11, 2013 4:09 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Voice Lab routing issue All, I've put together my voice lab and I'm having a routing issue which is preventing me from completing the setup. From SiteA-rtr, I cannot ping 10.10.100.3 on the 3750. The switch cannot ping 10.10.100.1 on the router. I have the PSTN-WAN router connected to f1/0/24 on the switch. From PSTN router I can ping 10.10.100.3 but not 10.10.100.1. The servers can all talk to each other, and the SiteA router can ping the servers, but the servers cannot ping 10.10.100.2 (NTP). It seems to be the connection from Switch port 1 to SiteA router subinterface for VLAN 101 only. The initial configurations came directly from Proctorlabs and I've edited from there. Can anyone take a look and see what I'm missing. It must be something obvious but I'm not seeing it. Thanks much, Dane Warner, CCVP Sr. Network Engineer Epoch Universal, Inc. (909)226-0755 dwar...@epochuniversal.commailto:dwar...@epochuniversal.com [Epoch_Logo_Smaller_Transparent] From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson Sent: Monday, October 07, 2013 10:20 AM To: Ramcharan Arya; Josh Petro Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; sanity insanity Subject: [OSL | CCIE_Voice] ccie written hi Arya and all, when you have 2 ccie specialization, do you need to write WRITTEN exam for both or just one ? K From: Ramcharan Arya ramcharan.a...@gmail.commailto:ramcharan.a...@gmail.com To: Josh Petro josh.pe...@gmail.commailto:josh.pe...@gmail.com Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; sanity insanity networksanitytoinsan...@gmail.commailto:networksanitytoinsan...@gmail.com Sent: Wednesday, October 2, 2013 6:59:08 PM Subject: Re: [OSL | CCIE_Voice] Presence - on hook and off hook status Hi Josh, I do not believe it is related to vmware environment. I am assuming your CUPS is integrated with CUCM using SIP trunk. Can you enable SIP debug level to detail and run collect SIP debug logs ( on primary call processing engine i.e. Sub) and check SIP logs why there is delay in status.? Im my home lab I never had this issue it works almost instantly my CUPS client is installed on UCCX server. Regards, Ramcharan Arya CCIE # 28926 ( Voice/RS) On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro josh.pe...@gmail.commailto:josh.pe...@gmail.com wrote: I have a huge delay in my presence updates on my system. Im assuming thatapos;s from CUPS being installed in my lab vmware environment though. Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone knows how to fix the lag, please let me know. Im assuming its related to vmware. Josh On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.commailto:ovi.p...@gmail.com wrote: Hi You could try a reboot of the CUPS server. Worked for me a couple of times... Cheers, Ovidiu On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.commailto:martinsloa...@gmail.com wrote: Hi MJ, Is the end user assigned on the line level of the hard phone? That assignment is unique per line appearance so if you make the association on the CUPC device it does not automatically populate to the hard phone/any other line appearance. When the phone goes off-hook CUCM checks the end user assignment for that appearance and if there is an end user assigned it check whether that end user is assigned CUP licensing to decide if the publish message is sent over the CUPS SIP trunk. BR, Marty On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity networksanitytoinsan...@gmail.commailto:networksanitytoinsan...@gmail.com wrote: Hello all, I have configured presence and both softphone and deskphone modes , IM and voicemail is working fine on the clients However I have a question when I lift the handset of the phone ( hard phone ) that is assoicated with the CUPC clients . I see that the presence status does not showOn the phone and does not turn yellow. I have tried reseting my sip trunk pointing to the presence server yet I see the same issue. Please let me know what can be done to fix this ? Also is this a major issue ? -MJ ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training,
Re: [OSL | CCIE_Voice] Voice Lab routing issue
Without seeing your config it sounds like a vlan mismatch between switch and router. Make sure your switchport facing the router is trunking and the router has subinterfaces with the correct vlan tag. Native vlan number will also need to match. If you post those relevant parts of the config I can review. On Oct 11, 2013 5:15 PM, Dane Warner dwar...@epochuniversal.com wrote: All, ** ** I’ve put together my voice lab and I’m having a routing issue which is preventing me from completing the setup. ** ** From SiteA-rtr, I cannot ping 10.10.100.3 on the 3750. The switch cannot ping 10.10.100.1 on the router. I have the PSTN-WAN router connected to f1/0/24 on the switch. From PSTN router I can ping 10.10.100.3 but not 10.10.100.1. The servers can all talk to each other, and the SiteA router can ping the servers, but the servers cannot ping 10.10.100.2 (NTP). It seems to be the connection from Switch port 1 to SiteA router subinterface for VLAN 101 only. The initial configurations came directly from Proctorlabs and I’ve edited from there. Can anyone take a look and see what I’m missing. It must be something obvious but I’m not seeing it. ** ** Thanks much, ** ** ** ** *Dane Warner, CCVP* *Sr. Network Engineer* *Epoch Universal, Inc.* *(909)226-0755* *dwar...@epochuniversal.com * *[image: Epoch_Logo_Smaller_Transparent]* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Karen Johnson *Sent:* Monday, October 07, 2013 10:20 AM *To:* Ramcharan Arya; Josh Petro *Cc:* ccie_voice@onlinestudylist.com; sanity insanity *Subject:* [OSL | CCIE_Voice] ccie written ** ** hi Arya and all, when you have 2 ccie specialization, do you need to write WRITTEN exam for both or just one ? K ** ** *From:* Ramcharan Arya ramcharan.a...@gmail.com *To:* Josh Petro josh.pe...@gmail.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; sanity insanity networksanitytoinsan...@gmail.com *Sent:* Wednesday, October 2, 2013 6:59:08 PM *Subject:* Re: [OSL | CCIE_Voice] Presence - on hook and off hook status** ** ** ** Hi Josh, I do not believe it is related to vmware environment. I am assuming your CUPS is integrated with CUCM using SIP trunk. Can you enable SIP debug level to detail and run collect SIP debug logs ( on primary call processing engine i.e. Sub) and check SIP logs why there is delay in status.? Im my home lab I never had this issue it works almost instantly my CUPS client is installed on UCCX server. Regards, Ramcharan Arya CCIE # 28926 ( Voice/RS) ** ** On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro josh.pe...@gmail.com wrote:** ** I have a huge delay in my presence updates on my system. Im assuming thatapos;s from CUPS being installed in my lab vmware environment though. Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone knows how to fix the lag, please let me know. Im assuming its related to vmware. Josh On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.com wrote: Hi ** ** You could try a reboot of the CUPS server. Worked for me a couple of times... ** ** Cheers, Ovidiu ** ** On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.com wrote: Hi MJ, Is the end user assigned on the line level of the hard phone? That assignment is unique per line appearance so if you make the association on the CUPC device it does not automatically populate to the hard phone/any other line appearance. When the phone goes off-hook CUCM checks the end user assignment for that appearance and if there is an end user assigned it check whether that end user is assigned CUP licensing to decide if the publish message is sent over the CUPS SIP trunk. BR, Marty ** ** On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hello all, I have configured presence and both softphone and deskphone modes , IM and voicemail is working fine on the clients However I have a question when I lift the handset of the phone ( hard phone ) that is assoicated with the CUPC clients . I see that the presence status does not show On the phone and does not turn yellow. I have tried reseting my sip trunk pointing to the presence server yet I see the same issue. Please let me know what can be done to fix this ? Also is this a major issue ? -MJ ** ** ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ** ** ___ For more information
Re: [OSL | CCIE_Voice] Voice Lab routing issue
Thank you very much. One word makes all the difference: Encapsulation dot1q 'native' Back on track again. Kind regards, Dane Warner, CCVP Sr. Network Engineer Epoch Universal, Inc. (909)226-0755 dwar...@epochuniversal.commailto:dwar...@epochuniversal.com [cid:image001.png@01CEC691.B7BC94B0] From: Fredenberg, Cliff [mailto:cliff.fredenb...@corebts.com] Sent: Friday, October 11, 2013 2:28 PM To: Dane Warner; ccie_voice@onlinestudylist.com Subject: RE: Voice Lab routing issue Site A router f0/0 is connected to switch port 1/0/1 right? The switch has the native vlan set but the router does not. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Dane Warner Sent: Friday, October 11, 2013 4:09 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Voice Lab routing issue All, I've put together my voice lab and I'm having a routing issue which is preventing me from completing the setup. From SiteA-rtr, I cannot ping 10.10.100.3 on the 3750. The switch cannot ping 10.10.100.1 on the router. I have the PSTN-WAN router connected to f1/0/24 on the switch. From PSTN router I can ping 10.10.100.3 but not 10.10.100.1. The servers can all talk to each other, and the SiteA router can ping the servers, but the servers cannot ping 10.10.100.2 (NTP). It seems to be the connection from Switch port 1 to SiteA router subinterface for VLAN 101 only. The initial configurations came directly from Proctorlabs and I've edited from there. Can anyone take a look and see what I'm missing. It must be something obvious but I'm not seeing it. Thanks much, Dane Warner, CCVP Sr. Network Engineer Epoch Universal, Inc. (909)226-0755 dwar...@epochuniversal.commailto:dwar...@epochuniversal.com [cid:image001.png@01CEC691.B7BC94B0] From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson Sent: Monday, October 07, 2013 10:20 AM To: Ramcharan Arya; Josh Petro Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; sanity insanity Subject: [OSL | CCIE_Voice] ccie written hi Arya and all, when you have 2 ccie specialization, do you need to write WRITTEN exam for both or just one ? K From: Ramcharan Arya ramcharan.a...@gmail.commailto:ramcharan.a...@gmail.com To: Josh Petro josh.pe...@gmail.commailto:josh.pe...@gmail.com Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com; sanity insanity networksanitytoinsan...@gmail.commailto:networksanitytoinsan...@gmail.com Sent: Wednesday, October 2, 2013 6:59:08 PM Subject: Re: [OSL | CCIE_Voice] Presence - on hook and off hook status Hi Josh, I do not believe it is related to vmware environment. I am assuming your CUPS is integrated with CUCM using SIP trunk. Can you enable SIP debug level to detail and run collect SIP debug logs ( on primary call processing engine i.e. Sub) and check SIP logs why there is delay in status.? Im my home lab I never had this issue it works almost instantly my CUPS client is installed on UCCX server. Regards, Ramcharan Arya CCIE # 28926 ( Voice/RS) On Wed, Oct 2, 2013 at 7:47 PM, Josh Petro josh.pe...@gmail.commailto:josh.pe...@gmail.com wrote: I have a huge delay in my presence updates on my system. Im assuming thatapos;s from CUPS being installed in my lab vmware environment though. Try to go off hook or call a pstn number and let it sit for 1-2 minutes. If anyone knows how to fix the lag, please let me know. Im assuming its related to vmware. Josh On Oct 1, 2013 11:20 AM, Ovidiu Popa ovi.p...@gmail.commailto:ovi.p...@gmail.com wrote: Hi You could try a reboot of the CUPS server. Worked for me a couple of times... Cheers, Ovidiu On Tue, Oct 1, 2013 at 4:43 PM, Martin Sloan martinsloa...@gmail.commailto:martinsloa...@gmail.com wrote: Hi MJ, Is the end user assigned on the line level of the hard phone? That assignment is unique per line appearance so if you make the association on the CUPC device it does not automatically populate to the hard phone/any other line appearance. When the phone goes off-hook CUCM checks the end user assignment for that appearance and if there is an end user assigned it check whether that end user is assigned CUP licensing to decide if the publish message is sent over the CUPS SIP trunk. BR, Marty On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity networksanitytoinsan...@gmail.commailto:networksanitytoinsan...@gmail.com wrote: Hello all, I have configured presence and both softphone and deskphone modes , IM and voicemail is working fine on the clients However I have a question when I lift the handset of the phone ( hard phone ) that is assoicated with the CUPC clients . I see that the presence status does not showOn the phone and
