[OSL | CCIE_Voice] Call-manager-fallback question
Hello, I don't know if it's possible to replicate in call-manager-fallback the busy trigger setting from the directory number in ucm. I have the following configuration: call-manager-fallback timeouts interdigit 4 ip source-address 10.1.1.200 port 2000 max-ephones 4 max-dn 4 octo-line transfer-pattern .T voicemail 3600 huntstop channel 1 call-forward pattern .T call-forward busy 3600 call-forward noan 3600 timeout 6 I though that the command "hunstop channel 1" will force the second call to go to voicemail if the line is busy, but it does not work. Any idea? Thank you ___ 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 translation rule
On Thu, Sep 6, 2012 at 3:27 PM, Chrysostomos Christofi < ch.christ...@logicom.net> wrote: > Hi > > ** ** > > voice translation-rule 30 > > rule 1 /\+12025552002/ /2002/ > > ** ** > > LOGPRIGW#test voice translation-rule 30 +12025552002 > > Matched with rule 1 > > Original number: +12025552002 Translated number: 2002 > > Original number type: none Translated number type: none > > Original number plan: none Translated number plan: none > > ** ** > > OR > > ** ** > > ** ** > > ** ** > > ** ** > > ** ** > > *From:* ccie_voice-boun...@onlinestudylist.com [mailto: > ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *The Masterplan > *Sent:* Πέμπτη, 6 Σεπτεμβρίου 2012 3:05 μμ > *To:* ccie_voice@onlinestudylist.com > *Subject:* [OSL | CCIE_Voice] Voice translation rule > > ** ** > > Hello, > > I want to strip the + sign followed by a number of digits from a number > using a voice translation rule. > For example, let's say that the input number is +12025552002 and I want > the output to be 2002. > > Thank you > ___ 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] Voice translation rule
Hello, I want to strip the + sign followed by a number of digits from a number using a voice translation rule. For example, let's say that the input number is +12025552002 and I want the output to be 2002. Thank you ___ 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] NTP daylight savings
Hello, I have the following scenario where ucm is syncing time from the router loopback address. On the router I have the following: ntp server 10.10.100.2 ntp source lo0 clock timezone PST -8 clock summer-time PDT reccuring Given the above configuration, when I create the date/time group in UCM, do I have to select the entry with the daylight time? And if i not configure daylight savings time on the router (no clock summer-time PDT reccuring) and I choose the entry with daylight time in UCM is that ok? Thank you ___ 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] SRST hunt group
Hi, I have an issue regarding srst. First requirement was that I should not have static mappings within the running configuration but to use Unified CME. So I used telephony-service with srst mode autoprovision none. The second requirement was to use bacd and to forward calls to 2 extensions based on longest idle. So i configured the following hunt group while in srst: ephone-hunt 1 longest-idle pilot list 4001, 4002 All worked fine in srst. When I was out of SRST I saved the config and then reloaded the router. After reload, the hunt group looks like this: ephone-hunt 1 longest-idle pilot If I put the phones back in srst, there are no extensions associated with that pilot, so hunt group don't function anymore. Is there a way to solve this without using autoprovision dn/all ? Thank you ___ 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] Overlapping route patterns
Hi, I have the following route patterns on CUCM that have a mgcp gateway in route list: 911 - marked as urgent priority 9.XXX - discard digit predot 91112.XXX- discard digit predot Because those are overlapping route patterns, their behaviour is different depending if enbloc dialing is used or not: 1)With digit by digit dialing if I dial 923942123 it will go to 911 because this route pattern have urgent priority enabled. If I dial 91112345 it will also go to 911 route pattern. 2)With en bloc dialing if I dial 923942123 it will go to the third route pattern as desired. If I dial 91112345 it will go to the second route pattern. So en bloc dialing will work good in this scenario. But if the routes above are configured in a CUCME, as dial-peers, I cannot find a way to force CUCME to have the behaviour as in the second point (en bloc dialing). CUCME uses logest match in dial-peers by default and even if I put preferences on those three dial-peers and force CUCME to use explicit preference first I did'n get the desired behaviour. Thank you ___ 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] CUC imported users behaviour
Awesome Kevin. I got burned 1 day by this bug. Can you give me the cisco
troubleshoot link from where you get those commands?
Thank you once again.
On Wed, Jul 11, 2012 at 8:17 PM, Kevin Spicer wrote:
> There's a bug in cuc that prevents mail from pstn being delivered if the
> smtp domain has been changed. Perhaps you hit that (affects the 5 lab
> workbook lab rentals).
> run cuc dbquery unitydirdb select * from tbl_alias
> to see if domains all match.
> If not fix with
> run cuc dbquery unitydirdb EXECUTE PROCEDURE csp_SmtpAddressMigrate('
> new.com','old.com')
> (substitute the correct old and new domain)
>
> On 11 Jul 2012 18:03, "The Masterplan" wrote:
>
>> Hi,
>>
>> I am struggling with a strange CUC behaviour and I am interested if
>> anyone experienced this.
>>
>> For users imported from CUC, no message is being recorded in user
>> mailbox, altough the system play (after the user press # for more options
>> and then 1 to send this message ) "your message has been sent". On that
>> imported user, if I go to edit-> Mailbox it shows that the number of
>> messages is 0.
>>
>> If I create users from CUC administration everything is working fine, I
>> receive MWI and I see the corresponding number of messages in user mailbox.
>>
>> In order to solve this issue, I deactivated/activated all the unity
>> services and then I restarted CUCM and CUC and still nothing. I usually
>> activate all the CUC services and then I start to configure it (Phone
>> System, Port group, Port, User templates etc) and in this case I've done
>> the same.
>>
>> Thank you experts,
>>
>> ___
>> 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] CUC imported users behaviour
Hi, I am struggling with a strange CUC behaviour and I am interested if anyone experienced this. For users imported from CUC, no message is being recorded in user mailbox, altough the system play (after the user press # for more options and then 1 to send this message ) "your message has been sent". On that imported user, if I go to edit-> Mailbox it shows that the number of messages is 0. If I create users from CUC administration everything is working fine, I receive MWI and I see the corresponding number of messages in user mailbox. In order to solve this issue, I deactivated/activated all the unity services and then I restarted CUCM and CUC and still nothing. I usually activate all the CUC services and then I start to configure it (Phone System, Port group, Port, User templates etc) and in this case I've done the same. Thank you experts, ___ 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] LAB8 - Question 7.1
I think you could use dialplan-pattern command. ___ 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] Gatekeeper trunk
Still nothing. I give up for the moment. The behaviour is the same in all cases. Thank you guys for your time and advices. If you have other ideas... On Thu, May 24, 2012 at 11:10 AM, Kevin Spicer wrote: > I've seen supplementary service fail when CUCM invokes network hold and > MOH doesn't support the codec of the call (real world CUCM 6 intercluster > trunk), so might be worth disabling MOH to check. > > On Thu, May 24, 2012 at 6:56 AM, The Masterplan > wrote: > >> Supplementary services still fails with all these commands. >> Results of debug voip ipipgw are below. In case of call from cme to ucm: >> >> May 24 06:13:49.148: >> //1515/761E975F86AF/H323/cch323_set_h245_state_mc_mode_outgoing: >> call_spi_mode = 3 >> May 24 06:13:49.148: >> //1515/761E975F86AF/H323/cch323_set_h245_state_mc_mode_outgoing: h245 state >> m/c mode=0x4F0, h323_ctl=0x0 >> May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_get_peer_info: Entry >> May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_get_peer_info: Have >> peer >> May 24 06:13:49.148: //-1//H323/cch323_get_dp_pref_mask: >> cch323_get_dp_pref_mask:IPIPGW(1515):setting mask for 729ar8also as 729 >> is configured >> May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_set_pref_codec_list: >> Peer channel present: dp pref mask=C >> May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_set_pref_codec_list: >> Not a single match to filter: dp pref mask=C >> May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_set_pref_codec_list: >> First preferred codec(bytes)=-1(0) >> May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_get_peer_info: Flow >> Mode set to FLOW_THROUGH >> May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_caps_chn_info: >> SIP->H323 xcoding, try to reserve transcoder for codec mismatch >> May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_caps_chn_info: >> Xcoding needed, setup pref codec >> May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_peer_info: Entry >> May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_peer_info: Have >> peer >> May 24 06:13:49.152: //-1//H323/cch323_get_dp_pref_mask: >> cch323_get_dp_pref_mask:IPIPGW(1515):setting mask for 729ar8also as 729 >> is configured >> May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_peer_info: Flow >> Mode set to FLOW_THROUGH >> May 24 06:13:49.152: >> //1515/761E975F86AF/H323/cch323_set_h323_control_options_outgoing: h245 sm >> mode = 1264 >> May 24 06:13:49.152: >> //1515/761E975F86AF/H323/cch323_set_h323_control_options_outgoing: >> h323_ctl=0x20 >> May 24 06:13:49.160: //1515/761E975F86AF/H323/cch323_rotary_validate: No >> peer_ccb available >> May 24 06:13:49.160: >> //1515/761E975F86AF/H323/cch323_build_local_encoded_fastStartOLCs: >> state_mc_mode=0x4F0 on outbound leg >> May 24 06:13:49.160: >> //1515/761E975F86AF/H323/cch323_build_local_encoded_fastStartOLCs: >> srcAddress = 0xA010123, h245_lport = 0, flow mode = 1, minimum_qos=0 >> May 24 06:13:49.160: >> //1515/761E975F86AF/H323/cch323_generic_open_logical_channel: current codec >> = 16:20:20 >> May 24 06:13:50.461: >> //1515/761E975F86AF/H323/cch323_receive_fastStart_cap_response: Send cap >> ind to peer leg >> May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_build_olc_for_ccapi: >> audioFastStartArray=0x45F99E9C >> May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_build_olc_for_ccapi: >> channel_info ptr=0x46044A78, ccb ptr=0x46011CF4 >> May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_build_olc_for_ccapi: >> Channel Information: >> Logical Channel Number (fwd): 1 >> Logical Channel Number (rev): 1 >> Channel address (fwd/rev):10.1.1.235 >> RTP Channel (fwd/rev): 17668 >> RTCP Channel (fwd/rev): 17669 >> QoS Capability (fwd/rev): 0 >> Symmetric Audio Codec:16 >> Symmetric Audio Codec Bytes: 20 >> Flow Mode:0 >> Silence Suppression: 1 >> May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_build_olc_for_ccapi: >> NumOfElements = 1 idx = 1 >> May 24 06:13:50.461: >> //1515/761E975F86AF/H323/cch323_receive_fastStart_cap_response: set >> have_lcl_caps to 1 >> May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_do_open_channel_ind: >> Sending event CC_EV_H245_OPEN_CHANNEL_IND, channelInfo pointer 0x46044A78 >> May 24 06:13:50.465: //1515/761E975F86AF/H323/h245_address_ind: ev=6 >> May 24 06:13:50
Re: [OSL | CCIE_Voice] Gatekeeper trunk
I have added this command also and the same behaviour. This problem drives me nuts. On Thu, May 24, 2012 at 9:46 AM, Peter Farkas wrote: > Try to add: > > ! > voice service voip > sip > midcall-signaling passthru > ! > > Peter > > - Original Message - From: "Mohd Baqari" > > To: "san r" > Cc: >; > "The Masterplan" > 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 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
Re: [OSL | CCIE_Voice] Gatekeeper trunk
Supplementary services still fails with all these commands. Results of debug voip ipipgw are below. In case of call from cme to ucm: May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_set_h245_state_mc_mode_outgoing: call_spi_mode = 3 May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_set_h245_state_mc_mode_outgoing: h245 state m/c mode=0x4F0, h323_ctl=0x0 May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_get_peer_info: Entry May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_get_peer_info: Have peer May 24 06:13:49.148: //-1//H323/cch323_get_dp_pref_mask: cch323_get_dp_pref_mask:IPIPGW(1515):setting mask for 729ar8also as 729 is configured May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_set_pref_codec_list: Peer channel present: dp pref mask=C May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_set_pref_codec_list: Not a single match to filter: dp pref mask=C May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_set_pref_codec_list: First preferred codec(bytes)=-1(0) May 24 06:13:49.148: //1515/761E975F86AF/H323/cch323_get_peer_info: Flow Mode set to FLOW_THROUGH May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_caps_chn_info: SIP->H323 xcoding, try to reserve transcoder for codec mismatch May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_caps_chn_info: Xcoding needed, setup pref codec May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_peer_info: Entry May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_peer_info: Have peer May 24 06:13:49.152: //-1//H323/cch323_get_dp_pref_mask: cch323_get_dp_pref_mask:IPIPGW(1515):setting mask for 729ar8also as 729 is configured May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_get_peer_info: Flow Mode set to FLOW_THROUGH May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_set_h323_control_options_outgoing: h245 sm mode = 1264 May 24 06:13:49.152: //1515/761E975F86AF/H323/cch323_set_h323_control_options_outgoing: h323_ctl=0x20 May 24 06:13:49.160: //1515/761E975F86AF/H323/cch323_rotary_validate: No peer_ccb available May 24 06:13:49.160: //1515/761E975F86AF/H323/cch323_build_local_encoded_fastStartOLCs: state_mc_mode=0x4F0 on outbound leg May 24 06:13:49.160: //1515/761E975F86AF/H323/cch323_build_local_encoded_fastStartOLCs: srcAddress = 0xA010123, h245_lport = 0, flow mode = 1, minimum_qos=0 May 24 06:13:49.160: //1515/761E975F86AF/H323/cch323_generic_open_logical_channel: current codec = 16:20:20 May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_receive_fastStart_cap_response: Send cap ind to peer leg May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_build_olc_for_ccapi: audioFastStartArray=0x45F99E9C May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_build_olc_for_ccapi: channel_info ptr=0x46044A78, ccb ptr=0x46011CF4 May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_build_olc_for_ccapi: Channel Information: Logical Channel Number (fwd): 1 Logical Channel Number (rev): 1 Channel address (fwd/rev):10.1.1.235 RTP Channel (fwd/rev): 17668 RTCP Channel (fwd/rev): 17669 QoS Capability (fwd/rev): 0 Symmetric Audio Codec:16 Symmetric Audio Codec Bytes: 20 Flow Mode:0 Silence Suppression: 1 May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_build_olc_for_ccapi: NumOfElements = 1 idx = 1 May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_receive_fastStart_cap_response: set have_lcl_caps to 1 May 24 06:13:50.461: //1515/761E975F86AF/H323/cch323_do_open_channel_ind: Sending event CC_EV_H245_OPEN_CHANNEL_IND, channelInfo pointer 0x46044A78 May 24 06:13:50.465: //1515/761E975F86AF/H323/h245_address_ind: ev=6 May 24 06:13:50.465: //1515/761E975F86AF/H323/cch323_h245_connection_sm: state=0, event=1, ccb=46011CF4, listen state=0 May 24 06:13:50.485: //1515/761E975F86AF/H323/cch323_h245_connection_sm: state=1, event=2, ccb=46011CF4, listen state=0 May 24 06:13:50.485: //1515/761E975F86AF/H323/h245_copy_preferred_codec_list: Copying Preferred codec into caps table May 24 06:13:50.485: //1515/761E975F86AF/H323/h245_send_generic_audio_caps: [trans]audio mask after operation=0xC May 24 06:13:50.493: //1515/761E975F86AF/H323/cch323_h245_cap_ind: Masks au=0x100C data=0x4 uinp=0x32 May 24 06:13:50.493: //-1//H323/cch323_get_dp_pref_mask: cch323_get_dp_pref_mask:IPIPGW(1515):setting mask for 729ar8also as 729 is configured May 24 06:13:50.493: //1515/761E975F86AF/H323/cch323_set_extended_caps: Copying codec list into extended caps structure May 24 06:13:50.493: //1515/761E975F86AF/H323/cch323_set_extended_caps: G729IETF May 24 06:13:50.493: //1515/761E975F86AF/H323/cch323_set_extended_caps: G729a May 24 06:13:50.497: //1515/761E975F86AF/H323/cch323_iwf_cap_notify: Mask sent to other leg=C May 24 06:13:50.497: //1515/761E975F86AF/H323/cch323_iwf_cap_notify: ../voip/cch323/gw/src/cch323_h245_iwf_util.c:cch323_iwf_cap_notify:1048 Pos
Re: [OSL | CCIE_Voice] Gatekeeper trunk
Yes, I have mtp on trunk. In the media resource group list assigned on trunk I have first a media resource group containing the hardware mtp and second a media resource group with the transcoder. On Wed, May 23, 2012 at 9:46 PM, san r wrote: > I believe it's supposed to be there on GK. Do you have Mtp on trunk? > On May 24, 2012 12:11 AM, "The Masterplan" > wrote: > >> No, only the gatekeeper configuration needed to route calls and the >> following: >> >> voice service voip >> allow-connections h323 to h323 >> allow-connections h323 to sip >> allow-connections sip to h323 >> allow-connections sip to sip >> >> >> On Wed, May 23, 2012 at 9:37 PM, san r wrote: >> >>> Do you have the same on GK as well? >>> On May 24, 2012 12:05 AM, "The Masterplan" >>> wrote: >>> >>>> Hi, >>>> >>>> This is the cme side relevant config: >>>> >>>> voice service voip >>>> allow-connections h323 to h323 >>>> allow-connections h323 to sip >>>> allow-connections sip to h323 >>>> allow-connections sip to sip >>>> h323 >>>> emptycapability >>>> h225 connect-passthru >>>> h245 passthru tcsnonstd-passthru >>>> sip >>>> bind control source-interface FastEthernet0/0 >>>> bind media source-interface FastEthernet0/0 >>>> registrar server >>>> no update-callerid >>>> >>>> voice register global >>>> mode cme >>>> source-address 10.1.1.235 port 5060 >>>> max-dn 4 >>>> max-pool 3 >>>> authenticate register >>>> create profile sync 0054221011645706 >>>> voice register dn 2 >>>> number 1006 >>>> call-forward b2bua busy 2000 >>>> call-forward b2bua mailbox 2100 >>>> call-forward b2bua noan 2000 timeout 20 >>>> ! >>>> voice register pool 1 >>>> id mac .. >>>> number 1 dn 2 >>>> dtmf-relay rtp-nte >>>> username 1006 password cisco >>>> codec g711ulaw >>>> >>>> interface FastEthernet0/0 >>>> ip address 10.1.1.235 255.255.0.0 >>>> h323-gateway voip interface >>>> h323-gateway voip id UCM ipaddr 10.1.1.171 1719 >>>> h323-gateway voip h323-id cme >>>> h323-gateway voip tech-prefix 1# >>>> h323-gateway voip bind srcaddr 10.1.1.235 >>>> >>>> dial-peer voice 5000 voip >>>> translation-profile incoming stripgk >>>> destination-pattern 5...$ >>>> session target ras >>>> incoming called-number . >>>> tech-prefix 1# >>>> dtmf-relay h245-alphanumeric >>>> no vad >>>> >>>> >>>> On Wed, May 23, 2012 at 9:08 PM, san r wrote: >>>> >>>>> Please paste your configs >>>>> On May 23, 2012 11:23 PM, "The Masterplan" >>>>> wrote: >>>>> >>>>>> I configured that, but the result is the same. >>>>>> >>>>>> On Wed, May 23, 2012 at 6:50 PM, san r wrote: >>>>>> >>>>>>> Try the following commands in voice class or H.323 voice-service >>>>>>> configuration mode - h323 end >>>>>>> >>>>>>> h225 connect-passthru >>>>>>> emptycapability >>>>>>> h245 passthru tcsnonstd-passthru >>>>>>> >>>>>>> On Wed, May 23, 2012 at 5:35 PM, The Masterplan < >>>>>>> winmasterp...@gmail.com> wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> I have the following setup: >>>>>>>> Sip IP phone <---> CME <--->h323<> GK <--->GK trunk<> >>>>>>>> CUCM<--> Sip IP phone >>>>>>>> On the cme side I have a transcoder and sip ip phone configured >>>>>>>> with codec g711ulaw. >>>>>>>> On the cucm side I also have a transcoder on mrgl of gk trunk that >>>>>>>> is in a device pool configured to speak g729 only. I have unchecked on >>>>>>>> trunk "Wait for Far End H.245 Terminal Capability Set" and checked the >>>>>>>> following: >>>>>>>> - Media termination point required >>>>>>>> - Inbound Fast Start >>>>>>>> - Outbound Fast Start with G729 codec >>>>>>>> Given this facts, the call is established succesfully (transcoders >>>>>>>> are used) but supplementary services are not working from either side. >>>>>>>> >>>>>>>> Thank you >>>>>>>> >>>>>>>> ___ >>>>>>>> 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] Gatekeeper trunk
No, only the gatekeeper configuration needed to route calls and the following: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip On Wed, May 23, 2012 at 9:37 PM, san r wrote: > Do you have the same on GK as well? > On May 24, 2012 12:05 AM, "The Masterplan" > wrote: > >> Hi, >> >> This is the cme side relevant config: >> >> voice service voip >> allow-connections h323 to h323 >> allow-connections h323 to sip >> allow-connections sip to h323 >> allow-connections sip to sip >> h323 >> emptycapability >> h225 connect-passthru >> h245 passthru tcsnonstd-passthru >> sip >> bind control source-interface FastEthernet0/0 >> bind media source-interface FastEthernet0/0 >> registrar server >> no update-callerid >> >> voice register global >> mode cme >> source-address 10.1.1.235 port 5060 >> max-dn 4 >> max-pool 3 >> authenticate register >> create profile sync 0054221011645706 >> voice register dn 2 >> number 1006 >> call-forward b2bua busy 2000 >> call-forward b2bua mailbox 2100 >> call-forward b2bua noan 2000 timeout 20 >> ! >> voice register pool 1 >> id mac .. >> number 1 dn 2 >> dtmf-relay rtp-nte >> username 1006 password cisco >> codec g711ulaw >> >> interface FastEthernet0/0 >> ip address 10.1.1.235 255.255.0.0 >> h323-gateway voip interface >> h323-gateway voip id UCM ipaddr 10.1.1.171 1719 >> h323-gateway voip h323-id cme >> h323-gateway voip tech-prefix 1# >> h323-gateway voip bind srcaddr 10.1.1.235 >> >> dial-peer voice 5000 voip >> translation-profile incoming stripgk >> destination-pattern 5...$ >> session target ras >> incoming called-number . >> tech-prefix 1# >> dtmf-relay h245-alphanumeric >> no vad >> >> >> On Wed, May 23, 2012 at 9:08 PM, san r wrote: >> >>> Please paste your configs >>> On May 23, 2012 11:23 PM, "The Masterplan" >>> wrote: >>> >>>> I configured that, but the result is the same. >>>> >>>> On Wed, May 23, 2012 at 6:50 PM, san r wrote: >>>> >>>>> Try the following commands in voice class or H.323 voice-service >>>>> configuration mode - h323 end >>>>> >>>>> h225 connect-passthru >>>>> emptycapability >>>>> h245 passthru tcsnonstd-passthru >>>>> >>>>> On Wed, May 23, 2012 at 5:35 PM, The Masterplan < >>>>> winmasterp...@gmail.com> wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> I have the following setup: >>>>>> Sip IP phone <---> CME <--->h323<> GK <--->GK trunk<> >>>>>> CUCM<--> Sip IP phone >>>>>> On the cme side I have a transcoder and sip ip phone configured with >>>>>> codec g711ulaw. >>>>>> On the cucm side I also have a transcoder on mrgl of gk trunk that is >>>>>> in a device pool configured to speak g729 only. I have unchecked on trunk >>>>>> "Wait for Far End H.245 Terminal Capability Set" and checked the >>>>>> following: >>>>>> - Media termination point required >>>>>> - Inbound Fast Start >>>>>> - Outbound Fast Start with G729 codec >>>>>> Given this facts, the call is established succesfully (transcoders >>>>>> are used) but supplementary services are not working from either side. >>>>>> >>>>>> Thank you >>>>>> >>>>>> ___ >>>>>> 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] Gatekeeper trunk
Hi, This is the cme side relevant config: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 emptycapability h225 connect-passthru h245 passthru tcsnonstd-passthru sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 registrar server no update-callerid voice register global mode cme source-address 10.1.1.235 port 5060 max-dn 4 max-pool 3 authenticate register create profile sync 0054221011645706 voice register dn 2 number 1006 call-forward b2bua busy 2000 call-forward b2bua mailbox 2100 call-forward b2bua noan 2000 timeout 20 ! voice register pool 1 id mac .. number 1 dn 2 dtmf-relay rtp-nte username 1006 password cisco codec g711ulaw interface FastEthernet0/0 ip address 10.1.1.235 255.255.0.0 h323-gateway voip interface h323-gateway voip id UCM ipaddr 10.1.1.171 1719 h323-gateway voip h323-id cme h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 10.1.1.235 dial-peer voice 5000 voip translation-profile incoming stripgk destination-pattern 5...$ session target ras incoming called-number . tech-prefix 1# dtmf-relay h245-alphanumeric no vad On Wed, May 23, 2012 at 9:08 PM, san r wrote: > Please paste your configs > On May 23, 2012 11:23 PM, "The Masterplan" > wrote: > >> I configured that, but the result is the same. >> >> On Wed, May 23, 2012 at 6:50 PM, san r wrote: >> >>> Try the following commands in voice class or H.323 voice-service >>> configuration mode - h323 end >>> >>> h225 connect-passthru >>> emptycapability >>> h245 passthru tcsnonstd-passthru >>> >>> On Wed, May 23, 2012 at 5:35 PM, The Masterplan >> > wrote: >>> >>>> Hi, >>>> >>>> I have the following setup: >>>> Sip IP phone <---> CME <--->h323<> GK <--->GK trunk<> >>>> CUCM<--> Sip IP phone >>>> On the cme side I have a transcoder and sip ip phone configured with >>>> codec g711ulaw. >>>> On the cucm side I also have a transcoder on mrgl of gk trunk that is >>>> in a device pool configured to speak g729 only. I have unchecked on trunk >>>> "Wait for Far End H.245 Terminal Capability Set" and checked the following: >>>> - Media termination point required >>>> - Inbound Fast Start >>>> - Outbound Fast Start with G729 codec >>>> Given this facts, the call is established succesfully (transcoders are >>>> used) but supplementary services are not working from either side. >>>> >>>> Thank you >>>> >>>> ___ >>>> 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] Gatekeeper trunk
I configured that, but the result is the same. On Wed, May 23, 2012 at 6:50 PM, san r wrote: > Try the following commands in voice class or H.323 voice-service > configuration mode - h323 end > > h225 connect-passthru > emptycapability > h245 passthru tcsnonstd-passthru > > On Wed, May 23, 2012 at 5:35 PM, The Masterplan > wrote: > >> Hi, >> >> I have the following setup: >> Sip IP phone <---> CME <--->h323<> GK <--->GK trunk<> CUCM<--> >> Sip IP phone >> On the cme side I have a transcoder and sip ip phone configured with >> codec g711ulaw. >> On the cucm side I also have a transcoder on mrgl of gk trunk that is in >> a device pool configured to speak g729 only. I have unchecked on trunk >> "Wait for Far End H.245 Terminal Capability Set" and checked the following: >> - Media termination point required >> - Inbound Fast Start >> - Outbound Fast Start with G729 codec >> Given this facts, the call is established succesfully (transcoders are >> used) but supplementary services are not working from either side. >> >> Thank you >> >> ___ >> 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] Gatekeeper trunk
Hi, I have the following setup: Sip IP phone <---> CME <--->h323<> GK <--->GK trunk<> CUCM<--> Sip IP phone On the cme side I have a transcoder and sip ip phone configured with codec g711ulaw. On the cucm side I also have a transcoder on mrgl of gk trunk that is in a device pool configured to speak g729 only. I have unchecked on trunk "Wait for Far End H.245 Terminal Capability Set" and checked the following: - Media termination point required - Inbound Fast Start - Outbound Fast Start with G729 codec Given this facts, the call is established succesfully (transcoders are used) but supplementary services are not working from either side. Thank you ___ 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] Voice class codec and gatekeeper
Hi, I have 2 distinct questions: 1. In the case of gatekeeper, what is the difference between hopoff and outvia? 2. Is it a good approach to use the voice class codec below for all dial-peers in the exam (with small exceptions like dial-peer to cue or bacd where you only need g711u)? voice class codec 1 codec preference 1 g711u codec preference 2 g729r8 Thank you ___ 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] Codec question
I have the following situation: a callmanager express on a router which is also declared as a h323 gateway in a ucm. Given this facts, a codec can be forced on: voice register pool (for sip phones on cme) ephone (for sccp phones on cme) dial-peers inbound and outbound to ucm (used by phones on cme) h323 gateway (by putting the gateway in a device pool which have a certain region) And now 3 questions: If all these are set to g711, that means that the call will flow with codec g711 from cme to ucm? If only outbound dial-peer codec is forced to g729r8 and all the rest are g711, that means that cme will need a transcoder for call to ucm to succeed? If only h323 gateway codec is forced to g729r8 and all the rest are g711. that means that ucm will need a transcoder for calls to succeed? Thank you ___ 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] UnifiedFx PhoneView does not work with cme
Hi Vik, I was connected to proctor labs. I don't have now a session available, but accordingly to their site the ios on the router should have been 12.4.22T. Thank you for your support. On Mon, Mar 12, 2012 at 6:57 PM, Vik Malhi wrote: > Can you confirm the version of IOS on the BR2. > > Vik Malhi – CCIE #13890 > Managing Partner - IPexpert, Inc. > > Telephone: +1.810.326.1444 ext 420 > Fax: +1.810.454.0130 > Mailto: vma...@ipexpert.com > > > > > On Mar 12, 2012, at 6:03 AM, The Masterplan wrote: > > Hi, > > I'm interested if anyone had the following issue. I was solving workbook 2 > lab 1 and I was trying to manage branch2 phones registered in cme at branch > 2 site with PhoneView. > I pasted the lines mentioned in the pdf into the router config, but > nothingafter I tested that the group has connectivity with cme and > click add no phone appears in the interface. UCM works perfectly. I have > Windows 7 x64 and .NET framework ver. 4. > > ___ > 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] UnifiedFx PhoneView does not work with cme
Hi, I'm interested if anyone had the following issue. I was solving workbook 2 lab 1 and I was trying to manage branch2 phones registered in cme at branch 2 site with PhoneView. I pasted the lines mentioned in the pdf into the router config, but nothingafter I tested that the group has connectivity with cme and click add no phone appears in the interface. UCM works perfectly. I have Windows 7 x64 and .NET framework ver. 4. ___ 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] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM
The same thing. The only difference is that now I don't see anymore the phone in show run. On Wed, Jan 18, 2012 at 12:34 PM, Mohd Baqari wrote: > Ok ... Try changing the provision mode to none ... Delete ephone-dn 1 and > ephone 3 Then do your testing. > > Regards, > Mohammed Al Baqari > > Sent from my iPhone > > On Jan 18, 2012, at 1:12 PM, The Masterplan > wrote: > > The phone does not appear in the output of the command: > tftp-server flash:gui/ephone_admin.html > max-ephones 14 > ephone-dn 11 dual-line > number 1002 no-reg primary > description Cisco IP Communicator > name Cisco IP Communicator > ephone-dn 12 dual-line > number 1005 no-reg primary > description IP Blue > name IP Blue > night-service bell > ephone 1 > description Cisco IP Communicator > mac-address F0DE.F173.03E2 > type CIPC > button 1:11 > ephone 2 > description IP Blue > mac-address 0050.56C0.0008 > type CIPC > button 1:12 > > After I switch to srst, the output of the command looks like this: > tftp-server flash:gui/ephone_admin.html > max-ephones 14 > ephone-dn 1 dual-line > number > description 7945 hardware > name 7945 hardware > ephone-dn 11 dual-line > number 1002 no-reg primary > description Cisco IP Communicator > name Cisco IP Communicator > ephone-dn 12 dual-line > number 1005 no-reg primary > description IP Blue > name IP Blue > night-service bell > ephone 1 > description Cisco IP Communicator > mac-address F0DE.F173.03E2 > type CIPC > button 1:11 > ephone 2 > description IP Blue > mac-address 0050.56C0.0008 > type CIPC > button 1:12 > ephone 3 > mac-address 0817.3514.5682 > button 1:1 > > > On Wed, Jan 18, 2012 at 10:25 AM, Mohd Baqari wrote: > >> Post the the output of "show run | sec ephone". Probably the old config >> of ephone-dn is saved in running config due to provision all >> >> Regards, >> Mohammed Al Baqari >> >> Sent from my iPhone >> >> On Jan 18, 2012, at 11:55 AM, The Masterplan >> wrote: >> >> Hi, >> >> I already did no create cnf-files & create cnf-files and reset the phone >> to factory defaults and nothing. I'm running srst mode auto provision all. >> See below the config: >> telephony-service >> srst mode auto-provision all >> srst dn line-mode dual >> em logout 0:0 0:0 0:0 >> max-ephones 14 >> max-dn 30 no-reg >> ip source-address 10.1.1.25 port 2000 >> system message SRST >> max-conferences 8 gain -6 >> transfer-system full-consult >> create cnf-files version-stamp 7960 Jan 18 2012 08:09:33 >> >> On Tue, Jan 17, 2012 at 7:13 PM, Ken Wyan wrote: >> >>> Did you enter no create cnf-files & create cnf-files on CME Router ? >>> >>> Which srst mode are you running? srst mode auto provision all | dn | >>> none ? >>> Better to post full telephony-service configuration here. >>> >>> (In SRST phone may be downloading previous xml configuration file from >>> the router. Delete it from flash if so) >>> >>> >>> >>> On Tue, Jan 17, 2012 at 6:53 PM, The Masterplan >> > wrote: >>> >>>> Hello, >>>> >>>> I have a problem regarding srst. The 2811 router than now is a srst was >>>> a acting as a cme in past in a demo lab and the 7960 phone was registered >>>> to it with extension . Now, the 7960 phone is registered in UCM with >>>> extension 5001 and the 2811 router is configured in telephony service srst >>>> mode. The problem is that although the old configuration of the router was >>>> erased, when it goes in srst fallback mode, the 7960 gets extension >>>> instead of 5001 and the command show telephony-service ephone shows the >>>> specified phone with message:This is an srst fallback phone. >>>> >>>> Thank you for your answer >>>> >>>> ___ >>>> 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 <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
Re: [OSL | CCIE_Voice] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM
The phone does not appear in the output of the command: tftp-server flash:gui/ephone_admin.html max-ephones 14 ephone-dn 11 dual-line number 1002 no-reg primary description Cisco IP Communicator name Cisco IP Communicator ephone-dn 12 dual-line number 1005 no-reg primary description IP Blue name IP Blue night-service bell ephone 1 description Cisco IP Communicator mac-address F0DE.F173.03E2 type CIPC button 1:11 ephone 2 description IP Blue mac-address 0050.56C0.0008 type CIPC button 1:12 After I switch to srst, the output of the command looks like this: tftp-server flash:gui/ephone_admin.html max-ephones 14 ephone-dn 1 dual-line number description 7945 hardware name 7945 hardware ephone-dn 11 dual-line number 1002 no-reg primary description Cisco IP Communicator name Cisco IP Communicator ephone-dn 12 dual-line number 1005 no-reg primary description IP Blue name IP Blue night-service bell ephone 1 description Cisco IP Communicator mac-address F0DE.F173.03E2 type CIPC button 1:11 ephone 2 description IP Blue mac-address 0050.56C0.0008 type CIPC button 1:12 ephone 3 mac-address 0817.3514.5682 button 1:1 On Wed, Jan 18, 2012 at 10:25 AM, Mohd Baqari wrote: > Post the the output of "show run | sec ephone". Probably the old config of > ephone-dn is saved in running config due to provision all > > Regards, > Mohammed Al Baqari > > Sent from my iPhone > > On Jan 18, 2012, at 11:55 AM, The Masterplan > wrote: > > Hi, > > I already did no create cnf-files & create cnf-files and reset the phone > to factory defaults and nothing. I'm running srst mode auto provision all. > See below the config: > telephony-service > srst mode auto-provision all > srst dn line-mode dual > em logout 0:0 0:0 0:0 > max-ephones 14 > max-dn 30 no-reg > ip source-address 10.1.1.25 port 2000 > system message SRST > max-conferences 8 gain -6 > transfer-system full-consult > create cnf-files version-stamp 7960 Jan 18 2012 08:09:33 > > On Tue, Jan 17, 2012 at 7:13 PM, Ken Wyan wrote: > >> Did you enter no create cnf-files & create cnf-files on CME Router ? >> >> Which srst mode are you running? srst mode auto provision all | dn | none >> ? >> Better to post full telephony-service configuration here. >> >> (In SRST phone may be downloading previous xml configuration file from >> the router. Delete it from flash if so) >> >> >> >> On Tue, Jan 17, 2012 at 6:53 PM, The Masterplan >> wrote: >> >>> Hello, >>> >>> I have a problem regarding srst. The 2811 router than now is a srst was >>> a acting as a cme in past in a demo lab and the 7960 phone was registered >>> to it with extension . Now, the 7960 phone is registered in UCM with >>> extension 5001 and the 2811 router is configured in telephony service srst >>> mode. The problem is that although the old configuration of the router was >>> erased, when it goes in srst fallback mode, the 7960 gets extension >>> instead of 5001 and the command show telephony-service ephone shows the >>> specified phone with message:This is an srst fallback phone. >>> >>> Thank you for your answer >>> >>> ___ >>> 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 <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
Re: [OSL | CCIE_Voice] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM
Hi, I already did no create cnf-files & create cnf-files and reset the phone to factory defaults and nothing. I'm running srst mode auto provision all. See below the config: telephony-service srst mode auto-provision all srst dn line-mode dual em logout 0:0 0:0 0:0 max-ephones 14 max-dn 30 no-reg ip source-address 10.1.1.25 port 2000 system message SRST max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jan 18 2012 08:09:33 On Tue, Jan 17, 2012 at 7:13 PM, Ken Wyan wrote: > Did you enter no create cnf-files & create cnf-files on CME Router ? > > Which srst mode are you running? srst mode auto provision all | dn | none ? > Better to post full telephony-service configuration here. > > (In SRST phone may be downloading previous xml configuration file from the > router. Delete it from flash if so) > > > > On Tue, Jan 17, 2012 at 6:53 PM, The Masterplan > wrote: > >> Hello, >> >> I have a problem regarding srst. The 2811 router than now is a srst was a >> acting as a cme in past in a demo lab and the 7960 phone was registered to >> it with extension . Now, the 7960 phone is registered in UCM with >> extension 5001 and the 2811 router is configured in telephony service srst >> mode. The problem is that although the old configuration of the router was >> erased, when it goes in srst fallback mode, the 7960 gets extension >> instead of 5001 and the command show telephony-service ephone shows the >> specified phone with message:This is an srst fallback phone. >> >> Thank you for your answer >> >> ___ >> 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 <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
[OSL | CCIE_Voice] SRST problem: Phones in srst mode have a different configuration than they were registered to CUCM
Hello, I have a problem regarding srst. The 2811 router than now is a srst was a acting as a cme in past in a demo lab and the 7960 phone was registered to it with extension . Now, the 7960 phone is registered in UCM with extension 5001 and the 2811 router is configured in telephony service srst mode. The problem is that although the old configuration of the router was erased, when it goes in srst fallback mode, the 7960 gets extension instead of 5001 and the command show telephony-service ephone shows the specified phone with message:This is an srst fallback phone. Thank you for your answer ___ 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
