Hi all, I have an issue with AAR to VM over the PSTN.
Problem is i just get a busy tone after 21 seconds when it trys to forward the call onto VM. If i press the messages button it dials out to VM so the config seems right. Config below: CSS-ALL - pt-internal - \+! AAR GROUP - VM - no prefix BR1 PH2 device aar css - CSS-ALL line aar group VM VM HUNTPILOT aar group VM ext number mask - +12123945XXX Call flow below: 1001 -> 1002 cfw VM (5600) -> ext mask +12123945XXX \+ -> LRG Thanks Kev On 13 July 2010 04:17, <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 (Jeff Price (jeffpric)) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 12 Jul 2010 19:53:15 -0700 > From: Mark Holloway <m...@markholloway.com> > To: Graham Hopkins <ghopk...@wolf-rock.co.uk> > Cc: CCIE Voice Maillist <ccie_voice@onlinestudylist.com> > Subject: Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue > Message-ID: <51a752b1-a902-466f-8cfc-9145df827...@markholloway.com> > Content-Type: text/plain; charset="us-ascii" > > Ok, so this is how set my H.323 gateway to operate. For example, a single > POTS dial peer to handle Local calls (7 digit called, 7 digit calling > number) for normal operation with UCM and when the router is in SRST mode. > > dial-peer voice 4 voip > description Calls from UCM add 9 > translation-profile incoming ADD9 > incoming called-number . > > voice translation-profile ADD9 > translate called 50 > > voice translation-rule 50 > rule 1 /\(.*\)/ /9\1/ > > > > dial-peer voice 920 pots > description LOCAL > translation-profile outgoing LOCAL > destination-pattern 9[2-9]......$ > port 0/0/0:23 > > voice translation-profile LOCAL > translate calling 11 > translate called 10 > > voice translation-rule 10 > rule 1 // // type unknown subscriber plan unknown isdn > ! > voice translation-rule 11 > rule 1 /\(^2...$\)/ /222\1/ > > > > On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote: > > > 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 > > > > _______________________________________________ > > 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/20100712/a519770e/attachment-0001.html> > > ------------------------------ > > Message: 2 > Date: Mon, 12 Jul 2010 22:17:10 -0500 > From: "Jeff Price (jeffpric)" <jeffp...@cisco.com> > To: "Mark Holloway" <m...@markholloway.com>, "Graham Hopkins" > <ghopk...@wolf-rock.co.uk> > Cc: CCIE Voice Maillist <ccie_voice@onlinestudylist.com> > Subject: Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue > Message-ID: > <b2de0afa86565c47bd3a8435550f9553014df...@xmb-rcd-201.cisco.com> > Content-Type: text/plain; charset="us-ascii" > > I still think the easiest way to do this is to have all dial-peers with > a 9. As you are configuring, you may run into issues with TEHO, but if > you do a debug voip dialpeer you can see the incoming number and add any > "extra" dial-peers you may need. I have noticed for some reason when I > configure TEHO, sometimes CUCM doesn't add the 9 even though it is in > the prefix box of the RL. In this event, as I said, it would be easier > to do the debug and figure out what is being sent. > > > > This simplifies adding the SRST functionality, which you can pretty much > guarantee you will have to configure in some fashion. As you do these > configurations over and over in this method, you will start to just > think naturally about sending the 9. It is easier to have the 1 > dial-peer than have to create 2 for each type of dialing pattern. > > > > Just my opinion, > > > > Jeff > > > > From: ccie_voice-boun...@onlinestudylist.com > [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark > Holloway > Sent: Monday, July 12, 2010 6:23 PM > To: Graham Hopkins > Cc: CCIE Voice Maillist > Subject: Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue > > > > I proceeded to use the method where all my H323 dial-peers start with 9 > in the destination-pattern. I imagine it's more work to have UCM keep > the 9 on the dialed number because of TEHO to multiple gateways, it gets > very "busy" to know when to prepend and not prepend in UCM route lists. > Assuming 9 is stripped on UCM and the H323 gateway is adding 9 before > sending the call to a POTS dial peer, is a VoIP dial-peer being created > to match any incoming call and then it is sent through a > translation-profile so it can match a POTS dial peer? > > > > > > > > On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote: > > > > > > 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 <http://www.ipexpert.com/> > > > > _______________________________________________ > For more information regarding industry leading CCIE Lab training, > please visit www.ipexpert.com <http://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/20100712/3e4e9018/attachment.html> > > ------------------------------ > > _______________________________________________ > CCIE_Voice mailing list > CCIE_Voice@onlinestudylist.com > http://onlinestudylist.com/mailman/listinfo/ccie_voice > > > End of CCIE_Voice Digest, Vol 53, Issue 67 > ****************************************** >
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