mls qos srr-queue output cos-map queue 1 threshold 3 5 mls qos srr-queue output cos-map queue 2 threshold *1* 4
mls qos srr-queue output cos-map queue 2 threshold 3 6 7 mls qos srr-queue output cos-map queue 3 threshold 3 2 3 mls qos srr-queue output cos-map queue 4 threshold 3 0 *mls qos queue-set output 2 threshold 2 **60** 100 100 100* mls qos interface FastEthernet2/0/6 switchport trunk encapsulation dot1q switchport trunk allowed vlan 101,103,203 switchport mode trunk srr-queue bandwidth share 1 30 40 30 srr-queue bandwidth shape 4 0 0 0 queue-set 2 priority-queue out On Sat, Jul 10, 2010 at 5:50 AM, ghulam jilani <jilani.ghu...@gmail.com>wrote: > hi , > did u try to make another translation rule to strip 9 and it will be ok for > CUCM mode and > SRST mode. for example > > voice translation rule 9 > rule 1 /^3033...$/ /\0/ [it means you are sending XXXXXXX calling mask > from CUCM for Globalize] > rule 2 /^3...$/ /303\0/ [it means you are sending only XXXX from CUCM for > normal mode] > rule 3 /9/ // type any subscriber plan any isdn > rule 4 /^.*/ /\0/ type any subscriber plan any isdn > > voice translation-profile SUBSCRIBER > translate calling 9 > translate called 9 > > dial-peer voice 9 pots > destination-pattern 9[2-9]...... > translation-profile outgoing SUBSCRIBER > port X/Y/Z:23 > > i did it this way. can u please check. > On Fri, Jul 9, 2010 at 5:04 PM, <ccie_voice-requ...@onlinestudylist.com>wrote: > >> Send CCIE_Voice mailing list submissions to >> ccie_voice@onlinestudylist.com >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://onlinestudylist.com/mailman/listinfo/ccie_voice >> or, via email, send a message with subject or body 'help' to >> ccie_voice-requ...@onlinestudylist.com >> >> You can reach the person managing the list at >> ccie_voice-ow...@onlinestudylist.com >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of CCIE_Voice digest..." >> >> >> Today's Topics: >> >> 1. Re: Globalisation/Localisation Issue (Mark Holloway) >> 2. Re: Globalisation/Localisation Issue (Graham Hopkins) >> 3. RTMT on Mac (Mark Holloway) >> 4. Re: RTMT on Mac (Tanner Ezell) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Fri, 9 Jul 2010 11:21:46 -0700 >> From: Mark Holloway <m...@markholloway.com> >> To: Joaquim Fernandes <joa_...@yahoo.com> >> Cc: ccie_voice@onlinestudylist.com >> Subject: Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue >> Message-ID: <56c5648a-48b3-46f1-98f1-38ecc6a08...@markholloway.com> >> Content-Type: text/plain; charset="us-ascii" >> >> Sounds like you have the PSTN to CUCM part working ok. >> >> This is what I have been doing. >> >> On the H323 router create the following dial-peer >> >> dial-peer voice 10 pots >> destination-pattern [2-9]......$ >> port 0/0/0:23 >> >> On CUCM have a Route Pattern that handles \+1414.[2-9]XXXXXX for calls >> originated by BR1 phones and strip the predot. This way you can assign the >> call type as Subscriber within the Route Pattern and if local calls are >> supposed to send a 7 digit calling number you can set the calling party >> transformation mask within the Route Pattern to XXXXXXX. >> >> >> You could have a second dial-peer on your H323 router for SRST >> >> dial-peer voice 910 pots >> destination-pattern 9[2-9]......$ >> port 0/0/0:23 >> translation-profile outgoing LOCAL >> >> >> There are really two different ways to handle H323 gateway dial-peers. >> You can strip the 9 in CUCM then add it back on the H323 gateway through a >> translation-profile and only have one set of dial-peers. Or, build your >> dial-peers for local, LD, international, and 911 without the 9, copy/paste >> in notepad and put a 9 in front of the dial-peer number and the >> destination-pattern then paste it into your router. You will have two sets >> of dial-peers for SRST and normal operation. >> >> >> >> >> On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: >> >> > HI Team, >> > >> > I have an issue with this question. >> > >> > Question >> > ======= >> > when pstn number 4143638888 call phones at site b they should display 7 >> digits on the phone display. >> > For example when pstn calling ph 1 or ph 2 at branch B it should display >> 3638888 on the screen. >> > >> > >> > My solution >> > ========= >> > >> > I have added +1 in Device pool of Branch B to make it globalised when >> the call comes in the H323 Branch B router. >> > >> > I have created \+1414.3638888 calling party transformation mask. >> > >> > I have created \+1414.3638888 route pattern with Branch B as the >> gateway. (branch b is the H323 gateway). >> > >> > So on the Route pattern i have just done predot and in the branch b >> route list i have done NANP-Predot and prefix 9. I have done vice versa as >> well but things doesnt work. >> > >> > IN the branch B router i have a dial-peer for the local calls. >> > >> > dial-peer voice 1 pots >> > destination-pattern 9[2-9]...... >> > port 0/0/0:23 >> > translation-profile outgoing local >> > >> > translation-rule 1 >> > rule 1 /^8.../ /363\0/ >> > >> > translation-rule 2 >> > rule 1 // // type any sub plan any isdn >> > >> > translation-profile lcoal >> > translate called 2 >> > translate calling 1 >> > >> > Note: If i make a dial-peer without 9 i.e (.......) >> > Then the display is perfect. but i dont feel this would be the solution. >> > >> > because in srst this would be an issue. >> > >> > >> > Issue >> > ===== >> > >> > The issue is when PSTN phone 4143638888 calls Brach B ph1 or ph2 the >> caller id is 3638888 and in the missed call its globalized number >> +14143638888 >> > as per the question. >> > >> > But when i do redial using missed calls from Branch B ph1 or ph2 the >> calling number on the ip phones is displayed as 93638888 (9 is the secondary >> dial tone) and the call goes through. Evrything works fine except for the >> display on ph1 or ph2, there is 9. >> > >> > How do i get rid of it 9. >> > >> > I hope i have made my point very clear of what issue i am facing. The >> question state the display on the phone should be only 3638888 and not >> 93638888. >> > >> > Regards, >> > JF >> > >> > _______________________________________________ >> > For more information regarding industry leading CCIE Lab training, >> please visit www.ipexpert.com >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> </archives/ccie_voice/attachments/20100709/f1b10d52/attachment-0001.html> >> >> ------------------------------ >> >> Message: 2 >> Date: Fri, 9 Jul 2010 20:22:11 +0100 >> From: Graham Hopkins <ghopk...@wolf-rock.co.uk> >> To: CCIE Voice Maillist <ccie_voice@onlinestudylist.com> >> Subject: Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue >> Message-ID: <65ec46dd-6c8c-47b1-8489-16fb490cb...@wolf-rock.co.uk> >> Content-Type: text/plain; charset="us-ascii" >> >> With the two sets of dial-peers you do need to take care that overlapping >> patterns don't cause problems in SRST for example I hit issues with >> >> [2-9]...... >> >> and >> >> 91[2-9]..[2-9]...... >> >> I decided to go with the translation pattern to put the 9 back on to the >> digits sent by CUCM, but this 9 will still show up on the phone unless you >> use >> >> voice service voip >> no supplementary-service h225-notify cid-update >> >> Regards >> >> Graham Hopkins >> >> >> >> >> On 9 Jul 2010, at 19:21, Mark Holloway wrote: >> >> > Sounds like you have the PSTN to CUCM part working ok. >> > >> > This is what I have been doing. >> > >> > On the H323 router create the following dial-peer >> > >> > dial-peer voice 10 pots >> > destination-pattern [2-9]......$ >> > port 0/0/0:23 >> > >> > On CUCM have a Route Pattern that handles \+1414.[2-9]XXXXXX for calls >> originated by BR1 phones and strip the predot. This way you can assign the >> call type as Subscriber within the Route Pattern and if local calls are >> supposed to send a 7 digit calling number you can set the calling party >> transformation mask within the Route Pattern to XXXXXXX. >> > >> > >> > You could have a second dial-peer on your H323 router for SRST >> > >> > dial-peer voice 910 pots >> > destination-pattern 9[2-9]......$ >> > port 0/0/0:23 >> > translation-profile outgoing LOCAL >> > >> > >> > There are really two different ways to handle H323 gateway dial-peers. >> You can strip the 9 in CUCM then add it back on the H323 gateway through a >> translation-profile and only have one set of dial-peers. Or, build your >> dial-peers for local, LD, international, and 911 without the 9, copy/paste >> in notepad and put a 9 in front of the dial-peer number and the >> destination-pattern then paste it into your router. You will have two sets >> of dial-peers for SRST and normal operation. >> > >> > >> > >> > >> > On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: >> > >> >> HI Team, >> >> >> >> I have an issue with this question. >> >> >> >> Question >> >> ======= >> >> when pstn number 4143638888 call phones at site b they should display 7 >> digits on the phone display. >> >> For example when pstn calling ph 1 or ph 2 at branch B it should >> display 3638888 on the screen. >> >> >> >> >> >> My solution >> >> ========= >> >> >> >> I have added +1 in Device pool of Branch B to make it globalised when >> the call comes in the H323 Branch B router. >> >> >> >> I have created \+1414.3638888 calling party transformation mask. >> >> >> >> I have created \+1414.3638888 route pattern with Branch B as the >> gateway. (branch b is the H323 gateway). >> >> >> >> So on the Route pattern i have just done predot and in the branch b >> route list i have done NANP-Predot and prefix 9. I have done vice versa as >> well but things doesnt work. >> >> >> >> IN the branch B router i have a dial-peer for the local calls. >> >> >> >> dial-peer voice 1 pots >> >> destination-pattern 9[2-9]...... >> >> port 0/0/0:23 >> >> translation-profile outgoing local >> >> >> >> translation-rule 1 >> >> rule 1 /^8.../ /363\0/ >> >> >> >> translation-rule 2 >> >> rule 1 // // type any sub plan any isdn >> >> >> >> translation-profile lcoal >> >> translate called 2 >> >> translate calling 1 >> >> >> >> Note: If i make a dial-peer without 9 i.e (.......) >> >> Then the display is perfect. but i dont feel this would be the >> solution. >> >> >> >> because in srst this would be an issue. >> >> >> >> >> >> Issue >> >> ===== >> >> >> >> The issue is when PSTN phone 4143638888 calls Brach B ph1 or ph2 the >> caller id is 3638888 and in the missed call its globalized number >> +14143638888 >> >> as per the question. >> >> >> >> But when i do redial using missed calls from Branch B ph1 or ph2 the >> calling number on the ip phones is displayed as 93638888 (9 is the secondary >> dial tone) and the call goes through. Evrything works fine except for the >> display on ph1 or ph2, there is 9. >> >> >> >> How do i get rid of it 9. >> >> >> >> I hope i have made my point very clear of what issue i am facing. The >> question state the display on the phone should be only 3638888 and not >> 93638888. >> >> >> >> Regards, >> >> JF >> >> >> >> _______________________________________________ >> >> 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 >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> </archives/ccie_voice/attachments/20100709/069c1ce0/attachment-0001.html> >> >> ------------------------------ >> >> Message: 3 >> Date: Fri, 9 Jul 2010 13:09:38 -0700 >> From: Mark Holloway <m...@markholloway.com> >> To: "ccie_voice@onlinestudylist.com osl" >> <ccie_voice@onlinestudylist.com> >> Subject: [OSL | CCIE_Voice] RTMT on Mac >> Message-ID: <61387254-9bf5-47da-96e1-8c91bfb94...@markholloway.com> >> Content-Type: text/plain; charset="us-ascii" >> >> I found an old thread about running RTMT on Mac but the blog that >> explained how to do it is down. Anyone know of a way to run RTMT on Mac? >> >> The old blog link was >> http://www.ciscomonkey.net/2009/10/13/real-time-monitoring-tool-on-mac-os-x >> >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: >> </archives/ccie_voice/attachments/20100709/7b7a04e6/attachment-0001.html> >> >> ------------------------------ >> >> Message: 4 >> Date: Fri, 9 Jul 2010 17:04:37 -0400 >> From: Tanner Ezell <tanner.ez...@gmail.com> >> To: Mark Holloway <m...@markholloway.com> >> Cc: "ccie_voice@onlinestudylist.com osl" >> <ccie_voice@onlinestudylist.com> >> Subject: Re: [OSL | CCIE_Voice] RTMT on Mac >> Message-ID: >> <aanlktilhayfhehlirhjiyeaamhrpno9zwukzbbixp...@mail.gmail.com> >> Content-Type: text/plain; charset=ISO-8859-1 >> >> Well, RTMT is just some [awful] java code, it can run on OS X. >> Regarding the blog, give it a day or so, looks like it's just down for >> maintenance. >> >> On Fri, Jul 9, 2010 at 4:09 PM, Mark Holloway <m...@markholloway.com> >> wrote: >> > I found an old thread about running RTMT on Mac but the blog that >> explained >> > how to do it is down. ?Anyone know of a way to run RTMT on Mac? >> > The old blog link >> > was? >> http://www.ciscomonkey.net/2009/10/13/real-time-monitoring-tool-on-mac-os-x >> > >> > >> > _______________________________________________ >> > For more information regarding industry leading CCIE Lab training, >> please >> > visit www.ipexpert.com >> > >> > >> >> >> ------------------------------ >> >> _______________________________________________ >> CCIE_Voice mailing list >> CCIE_Voice@onlinestudylist.com >> http://onlinestudylist.com/mailman/listinfo/ccie_voice >> >> >> End of CCIE_Voice Digest, Vol 53, Issue 54 >> ****************************************** >> > >
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