Do you have a sip trunk for CME built in CM? If so have you tried resetting the trunk
On Friday, September 14, 2012, William Affeldt wrote: > Hey Experts, > > I am having an issue with CME to CM presence. The CME can see the CM > presence but the CM can't see the CME presence. It never even send the > message to CM. The debug presence and important config is below. Any help > would be greatly appreciated. > > Sep 15 05:27:43.141: //PRESENCE:[0]:/presence_line_update: SCCP, no update > Sep 15 05:27:43.145: //PRESENCE:[0]:/presence_sip_line_update: SIP nothing > to update > > > **** ** > !**** > !**** > voice service voip**** > allow-connections h323 to h323**** > allow-connections h323 to sip**** > allow-connections sip to h323**** > allow-connections sip to sip**** > sip**** > bind control source-interface Loopback0**** > bind media source-interface Loopback0**** > !**** > !**** > !**** > presence**** > presence call-list**** > server 10.10.210.10**** > watcher all**** > allow subscribe**** > !**** > gateway**** > timer receive-rtp 1200**** > !**** > sip-ua**** > mwi-server ipv4:10.10.115.2 expires 3600 port 5060 transport udp > unsolicited**** > presence enable**** > !**** > !**** > telephony-service**** > sdspfarm units 2**** > sdspfarm transcode sessions 4**** > sdspfarm tag 1 sitectrans**** > no auto-reg-ephone**** > authentication credential admin cisco**** > em logout 0:0 0:0 0:0**** > max-ephones 10**** > max-dn 20**** > ip source-address 10.10.110.3 port 2000**** > system message Your current options**** > url services http://10.10.115.2/voiceview/common/login.do**** > url authentication http://10.10.110.3/CCMCIP/authenticate.asp**** > time-format 24**** > date-format yy-mm-dd**** > voicemail 4600**** > max-conferences 8 gain -6**** > web admin system name admin password cisco**** > dn-webedit**** > time-webedit**** > transfer-system full-consult**** > transfer-pattern .T**** > create cnf-files version-stamp 7960 Sep 15 2012 04:14:41**** > !**** > !**** > ephone-dn 1 octo-line**** > number 4001 no-reg primary**** > description +442077964001**** > name SiteCPhone1**** > allow watch**** > call-forward busy 4600**** > call-forward noan 4600 timeout 10**** > !**** > !**** > ephone-dn 2 octo-line**** > number 4002**** > description +442077964002**** > name SiteCPhone2**** > allow watch**** > call-forward busy 4600**** > call-forward noan 4600 timeout 10**** > !**** > !**** > ephone-dn 5 octo-line**** > number *4002**** > call-forward all 4600**** > !**** > !**** > ephone 1**** > mac-address 0021.A086.8796**** > username "sitecphn1"**** > type 7962**** > button 1:1**** > pin 12345**** > !**** > !**** > !**** > ephone 2**** > mac-address E8BA.70FA.A942**** > username "tjefferson"**** > presence call-list**** > blf-speed-dial 1 2002 label "2002"**** > type 7975**** > button 1:2 6m5**** > pin 12345**** > !**** > !**** > > > > > > > > > > > *From:* Bill Lake <[email protected] <javascript:_e({}, 'cvml', > '[email protected]');>> > *To:* John John <[email protected] <javascript:_e({}, 'cvml', > '[email protected]');>> > *Cc:* "[email protected] <javascript:_e({}, 'cvml', > '[email protected]');>" > <[email protected]<javascript:_e({}, 'cvml', > '[email protected]');>> > > *Sent:* Friday, September 14, 2012 3:11 PM > *Subject:* Re: [OSL | CCIE_Voice] sip trunk > > Sure you can, you know that any call incoming from Site A must have ANI of > 332211 and any call incoming from Site B must have 332222 so you just need > to enforce that and you could do that with a incoming voice translation > rule. > > So no matter what the customer sends you can change that to what you want > it to be > > We do this all the time in the work books > > voice translation-rule 1 > rule 1 /^3...$/ /408387\0/ > > this expands 3001 to 4083873001 > > but you could also do > > rule 1 /^3...$/ /4001/ > > This would take any 3XXX number and change it to 4001, why you ask, just > think you have 800 people calling out from your company call center and > this prevents 800 different numbers from going out. Everyone you call gets > 4001. not 3001-3799 or something like that. > > You get the idea? > > > > On Fri, Sep 14, 2012 at 4:35 PM, John John > <[email protected]<javascript:_e({}, 'cvml', > '[email protected]');> > > wrote: > > Dear Pavan, > > Thanks for your feedback,But my question is even if i implement cor still > I can't control that because I can't control what customer A and B and in > that case if customer A sent me the call with ANI for cutomer B then > customer B dial-peer will be match and the call will not be blocked. > > but during my google search I found some greating feature which is > incoming uri based on the IP so anybody working in that feature ???? > > > *From:* Pavan <[email protected] <javascript:_e({}, 'cvml', > '[email protected]');>> > *To:* John John <[email protected] <javascript:_e({}, 'cvml', > '[email protected]');>> > *Cc:* "[email protected] <javascript:_e({}, 'cvml', > '[email protected]');>" > <[email protected]<javascript:_e({}, 'cvml', > '[email protected]');>> > > *Sent:* Friday, September 14, 2012 6:43 AM > *Subject:* Re: [OSL | CCIE_Voice] sip trunk > > I think cor can offer a good solution for this. Assign a separate cor > group to each dialpeer say cust1 and cust2 in both inbound and outbound > directions > > -Pavan > > On Sep 13, 2012, at 17:45, John John > <[email protected]<javascript:_e({}, 'cvml', > '[email protected]');>> > wrote: > > Dear All, > > We have two sipt trunk for 2 comapany:- > > company A - DID range 332211XX > Company B - DID range 332222XX > > and each company has own PBX,Company A has AVAYA and company B has cisco > Call manager. > > and they have sip trunk to my gateway where is the E1 is connected. > > in my gateway there is two dial peer : > > dial-peer voice 1 voip > description ## Customer-A ## > answer-address 332211.. > destination-pattern 332211.. > modem passthrough nse codec g711ulaw > session protocol sipv2 > session target ipv4:192.168.10.10 > > dial-peer voice 2 voip > description ## Customer-B ## > answer-address 332222.. > destination-pattern 332222.. > modem passthrough nse codec g711ulaw > session protocol sipv2 > session target ipv4:192.168.20.10 > > > So in my case if customer A send me the ANI as 33222222 which is for > customer B then the call will hit dial-peer 2 and the call will go outside > with 33222222 as the ANI number so how we can block Site A calls if they > didn't use there own DID. > > > > > _______________________________________________ > 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, 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, 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
