Re: [OSL | CCIE_Voice] 3rd Day in raw :( SRST-unity integration problem , even CTI solution doesn't work,
Hi Vic, I really appreciate that u giving your time. I really badly stuck in this, and cannot find a solution yet. I use ccm 4.1(3)sr3c and 12.4 (5b) today's morning Ryan remote desktop to my laptop and configure translation pattern (3rd option to do SRST/unity) but surprisingly same result , 8 damn digit passed to unity (if I put this 8 digits in alternate number it would work) . I checked what u said twice no extra dial pattern , and VM profile is not working for both CTI and TP .so with 3 different solution I ran into same problem :( Me and Ryan suspect CCm version so we upgrade it with sr3c patch but same result. Hope somebody can suggest sth. Jeremy On Wed, Jan 14, 2009 at 5:52 PM, Vik Malhi vma...@ipexpert.com wrote: Do you have any # configured on your CCM with Ext 2229 other than the hunt pilot? Check in Route Plan Report. You haven't configured the CTI solution properly. If your CTI RP was 2888 the VM Mask is then Unity should see a Forwarding # of 2888. Check that out again- this CTI RP solution is a winner. Its very puzzling these damned 8 digits. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *jeremy co jeremy.coo...@gmail.com *Date: *Wed, 14 Jan 2009 05:12:19 +1100 *To: *ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Cc: *Vik Malhi vma...@ipexpert.com *Subject: *3rd Day in raw :( SRST-unity integration problem , even CTI solution doesn't work, Hi, I tried both DRNIS and CTI solution. None of them worked .in both ,8 digits passed to unity, I donnow why!!! ** RDNIS scenario : unity--HQ ---pstn-BR1 (SRST) 2001 3001 2001 call 3001 and CFNA redirect call to unity via pstn , redirecting number works fine but only 8 digits passed to unity Here is the out put of debug isdn on HQ when call forwarded to unity. HQ :499-202-2 BR1 :899-303-3XXX voice pilot number : 2229 Mar 11 20:01:40.060: ISDN Se0/0:23 Q931: RX - SETUP pd = 8 callref = 0x008E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x2181, '4992022002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '499209' Plan:ISDN, Type:National Redirecting Number i = 0xFF, '8993033001' Plan:Reserved, Type:Reserved I can see from call viewer in unity : dialed numbercalling number forwarding 93033001 4992022002 93033001 CTI scenraio : I made 2888 CTI with forward to voice mail option checked,and assign Voice mail profile with mask of to it. Then Configured SRST to forward all calls to 2888 DN. what I see in unity is again dialed numbercalling number forwarding 93033001 4992022002 93033001 I have no idea why 8 digits just passed to unity I waste lots of time to make SRST to work, but no success any help would be much appreciated. Jeremy
Re: [OSL | CCIE_Voice] QOS Marking
Hi Hany, There is a good vLecture regarding to Campus QoS on IPexpert website. Even so, your specific question is not clear for me also. My best guess is that you just have to configure voice and vlan in the interfaces, and let the qos being disable. Could be good if someone could clarify it a little more. Hany Hanna hhanna1000 at gmail.com Fri Jan 9 23:17:28 EST 2009 Previous message: [OSL | CCIE_Voice] Requested circuit/channel not available onPSTN GW ? Next message: [OSL | CCIE_Voice] Ephones still registering with gatekeeper with no-reg!! Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] What if I need to mark QOS only on routers R1, R2, R3. No marrking to be done on the switches. Should qos be just disabled on the switches? -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20090109/4e4a79b3/attachment.htm _ Organize seus contatos! O jeito mais fácil de manter a sua lista de amigos sempre em ordem! http://www.microsoft.com/windows/windowslive/events.aspx
Re: [OSL | CCIE_Voice] MOH Issue
Kamal, According to task wording below: Between Site A and Site B only g729 allowed and Site B will receive multicast MOH from router flash, no multicast traffic allowed between Ste A and SiteB - I take it as MOH from Site B router flash for Site B IP phones can actually use g711 and wouldn't bother enabling g729 for MOH. This requires placing MOH servers in G.711-only region/DP as per http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a00803f8950.shtml which I haven't mentioned in my previous email. Rgds Alex - Original Message - From: kamal yousaf To: Christian Hennrich Cc: Alex ; ccie_voice@onlinestudylist.com ; Kumar, Narinder Sent: Wednesday, January 14, 2009 5:13 AM Subject: Re: [OSL | CCIE_Voice] MOH Issue Alex, Regarding your comment, If MOH server uses 239.1.1.3 to stream G729 to remote phones, shouldn't we enable G.729 for MOH ? Only exception is using transcoder but since it can't be used for multicast,don't we have to enable G.729 ? Alex schrieb: Your mcast group IP@ in below debug is 239.1.1.3 The same group IP@ should be configured on SiteB router. No need to enable G.729 for MoH - if you ticked increment on IP address that's probably why the group IP@ got changed. Rgds Alex *From:* Kumar, Narinder mailto:narinder.ku...@uxcg.com.au *To:* ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com *Sent:* Tuesday, January 13, 2009 12:36 PM *Subject:* [OSL | CCIE_Voice] MOH Issue Quick Que on MOH CCM running multicast MOH. Between Site A and Site B only g729 allowed SiteA will receive multicast MOH . Site B will receive multicast MOH from router flash, no multicast traffic allowed between Ste A and SiteB. The way I do this question is Configure the MOH source file and tick multicast and play continuously Enable multicast on the MRG and MOH server Change the ip voice media service parameter to allow both g711 and g729 Site A works without any issue Site B Configuration: Call-manager-fallback Moh filename ( Moh file in flash) multicast moh 239.1.1.1 port 16384 route x.x.x.x MOH from site B doesn't work , what am I missing here ? *** debug ccm-manager music-on-hold all ** an 13 13:13:30.023: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:30.023: moh_process_ccb: dstadr 0.0.0.0, callid 18, port 0, codec 65535, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:30.023: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:30.079: moh_process_ccb: dstadr 142.102.65.6, callid 18, port 23552, codec 5, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:30.079: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:31.391: %ISDN-6-CONNECT: Interface Serial0/1/0:2 is now connected to 911 N/A *Jan 13 13:13:31.395: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:31.395: moh_process_ccb: dstadr 142.102.65.6, callid 18, port 23552, codec 5, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:31.399: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.119: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.139: moh_process_ccb: dstadr 239.1.1.3, callid 18, port 16384, codec 16, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:33.139: moh_process_ccb:multicast addr add_ccb *Jan 13 13:13:33.139: moh_add_ccb: ip addr 239.1.1.3 port 16384 callid 18 *Jan 13 13:13:33.139: moh_add_ccb: vmccb does not exists - creating a new one for 239.1.1.3 through IGMP *Jan 13 13:13:33.139: moh_join_group_command called for 239.1.1.3 *Jan 13 13:13:33.139: moh_join_group_command: Looking at valid idb's to configure 239.1.1.3 *Jan 13 13:13:33.139: moh_join_group_command: IGMP API on group 239.1.1.3 idb Se0/0/0.201 *Jan 13 13:13:33.139: moh_join_group_command: IGMP API on group 239.1.1.3 idb Vl102 *Jan 13 13:13:33.139: moh_create_session: called *Jan 13 13:13:33.139: moh_create_session : dstadr 239.1.1.3 does not exist - creating acontrol block *Jan
Re: [OSL | CCIE_Voice] MOH Issue
Re: [OSL | CCIE_Voice] MOH IssueVik, AFAIK if G.729 is enabled in IPVMSA and MOH server is in HQ DP (which allows only G.729 to Site B) then the following happens: - CCM instructs Site B phones to join mcast group 239.1.1.3 which is G.729 MOH -Site B router streams only G.711 MOH from flash to 239.1.1.1 http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1043334 Expected result: no MOH on Site B phones If G.729 is not enabled for IPVMSA and MOH server is in HQ DP (which allows only G.729 to Site B), then: - CCM cannot find a suitable mcast group for Site B phones to join Expected result: no MOH on Site B phones If G.729 is not enabled for IPVMSA and MOH server is in G.711-only DP (which allows G.711 to Site B), then: - CCM instructs Site B phones to join mcast group 239.1.1.1 which is G.711 ulaw MOH -Site B router streams only G.711 MOH from flash to 239.1.1.1 Expected result: MOH plays on Site B phones Do I miss something here? Rgds Alex - Original Message - From: Vik Malhi To: kamal yousaf ; Christian Hennrich Cc: ccie_voice@onlinestudylist.com ; Kumar,Narinder Sent: Wednesday, January 14, 2009 6:39 AM Subject: Re: [OSL | CCIE_Voice] MOH Issue 239.1.1.1 port 16384 is always reserved for g711u for Audio Source 1, 239.1.1.2 for g711alaw, 239.1.1.3 for g729 and 239.1.1.4 for Wideband. Since the BR1 DP is communicating with the MOH server using g729 (or should I say CCM thinks g729 is the negotiated codec whereas in realty we are spoofing from the remote site router) CallManager will increment on port # or ip address. In your case MOH---BR1 DP uses g729 and you increment on ip address. You should have G729 allowed in IP Voice Media Streaming App for this to work. Everything on CallManager should be configured as normal- without enabling g729 will cause the MOH sourced from the flash to fail. You need the commands as shown below. Call-manager-fallback Moh filename ( Moh file in flash no multicast moh 239.1.1.1 port 16384 route x.x.x.x multicast moh 239.1.1.3 port 16384 route x.x.x.x Note - you cannot change multicast ip addresses on the fly and so you must delete the first command (incorrect multicast cmd). -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- From: kamal yousaf lovingprin...@gmail.com Date: Wed, 14 Jan 2009 16:13:35 +1100 To: Christian Hennrich christian.hennr...@intact-is.com Cc: ccie_voice@onlinestudylist.com, Kumar, Narinder narinder.ku...@uxcg.com.au Subject: Re: [OSL | CCIE_Voice] MOH Issue Alex, Regarding your comment, If MOH server uses 239.1.1.3 to stream G729 to remote phones, shouldn't we enable G.729 for MOH ? Only exception is using transcoder but since it can't be used for multicast,don't we have to enable G.729 ? Alex schrieb: Your mcast group IP@ in below debug is 239.1.1.3 The same group IP@ should be configured on SiteB router. No need to enable G.729 for MoH - if you ticked increment on IP address that's probably why the group IP@ got changed. Rgds Alex *From:* Kumar, Narinder mailto:narinder.ku...@uxcg.com.au *To:* ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com *Sent:* Tuesday, January 13, 2009 12:36 PM *Subject:* [OSL | CCIE_Voice] MOH Issue Quick Que on MOH CCM running multicast MOH. Between Site A and Site B only g729 allowed SiteA will receive multicast MOH . Site B will receive multicast MOH from router flash, no multicast traffic allowed between Ste A and SiteB. The way I do this question is Configure the MOH source file and tick multicast and play continuously Enable multicast on the MRG and MOH server Change the ip voice media service parameter to allow both g711 and g729 Site A works without any issue Site B Configuration: Call-manager-fallback Moh filename ( Moh file in flash) multicast moh 239.1.1.1 port 16384 route x.x.x.x MOH from site B doesn't work , what am I missing here ? *** debug ccm-manager music-on-hold all
Re: [OSL | CCIE_Voice] Multicast MOH
can someone explain what you mean there? thanks, Ryan Trauernicht On Wed, Jan 14, 2009 at 12:39 AM, kamal yousaf lovingprin...@gmail.comwrote: I always forget g711 includes any thing g711 and below. How stupid i am. Thanks alot Vik . On Wed, Jan 14, 2009 at 5:32 PM, Vik Malhi vma...@ipexpert.com wrote: Firstly you cannot transcode a multicast stream so you are correct there. By placing the MOH server in a g711 Device Pool you are allowing all codecs that take up less bandwidth than g711 too. So that includes g729. So providing you change the IP Voice Media Streaming service params to allow G729 the MOH stream is being sent to the BR1 natively using g729. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *kamal yousaf lovingprin...@gmail.com *Date: *Wed, 14 Jan 2009 16:05:05 +1100 *To: *ccie_voice@onlinestudylist.com *Subject: *[OSL | CCIE_Voice] Multicast MOH Hi , I have MOH Sub and Pub configured to support Multicast for Br1 phones and Unicast for HQ phones (using MRGL).I placed MOH sub in G711 only device pool so that it communicates using G711 only. Now, since BR1phones/BR1 MGCP gw are in a device pool which communicates G729 to other device pools, how would my Multicast MOH get streamed to BR1 phones ? Multicast MOH server is using G711 , BR1 phone is using G729 and since there can be no transcoder invoked for Multicast MOH , how will this work ? Please help !
Re: [OSL | CCIE_Voice] BACD Voip peers
Vik, I think we are on to something. Here is what I get with the ras debug turned on: BR2#debug ras H.323 RAS Messages debugging is on BR2#h323chan_dgram_send:Sent UDP msg. Bytes sent: 83 to 172.16.101.1:1719 fd=2 .Jan 14 15:28:55.172 CET: RASLib::GW_RASSendRRQ: RRQ (seq# 3729) sent to 172.16.101.1 .Jan 14 15:28:55.180 CET: h323chan_chn_process_read_socket .Jan 14 15:28:55.180 CET: h323chan_chn_process_read_socket: fd=2 of type CONNECTED has data .Jan 14 15:28:55.180 CET: h323chan_chn_process_read_socket: h323chan accepted/connected fd=2 .Jan 14 15:28:55.180 CET: h323chan_dgram_recvdata:rcvd from [172.16.101.1:1719] on fd=2 .Jan 14 15:28:55.180 CET: RCF (seq# 3729) rcvd .Jan 14 15:28:56.404 CET: h323chan_chn_process_read_socket .Jan 14 15:28:56.404 CET: h323chan_chn_process_read_socket: fd=3 of type CONNECT_PENDING has data .Jan 14 15:28:56.404 CET: h323chan_chn_process_read_socket .Jan 14 15:28:56.404 CET: h323chan_chn_process_read_socket: fd=0 of type LISTENING has data .Jan 14 15:28:56.404 CET: h323chan_chn_process_read_socket .Jan 14 15:28:56.404 CET: h323chan_chn_process_read_socket: fd=3 of type CONNECTED has data .Jan 14 15:28:56.404 CET: h323chan_chn_process_read_socket: h323chan accepted/connected fd=3 .Jan 14 15:28:56.408 CET: h323chan_chn_process_read_socket .Jan 14 15:28:56.408 CET: h323chan_chn_process_read_socket: fd=4 of type ACCEPTED has data .Jan 14 15:28:56.408 CET: h323chan_chn_process_read_socket: h323chan accepted/connected fd=4 h323chan_dgram_send:Sent UDP msg. Bytes sent: 119 to 172.16.101.1:1719 fd=2 .Jan 14 15:28:56.412 CET: RASLib::GW_RASSendARQ: ARQ (seq# 3730) sent to 172.16.101.1 .Jan 14 15:28:56.420 CET: h323chan_chn_process_read_socket .Jan 14 15:28:56.420 CET: h323chan_chn_process_read_socket: fd=2 of type CONNECTED has data .Jan 14 15:28:56.420 CET: h323chan_chn_process_read_socket: h323chan accepted/connected fd=2 .Jan 14 15:28:56.420 CET: h323chan_dgram_recvdata:rcvd from [172.16.101.1:1719] on fd=2 .Jan 14 15:28:56.420 CET: ARJ (seq# 3730) rcvdparse_arj_nonstd: ARJ Nonstd decode succeeded, remlen = 1 .Jan 14 15:28:56.424 CET: h323chan_chn_process_read_socket .Jan 14 15:28:56.424 CET: h323chan_chn_process_read_socket: fd=3 of type CONNECTED has data .Jan 14 15:28:56.424 CET: h323chan_chn_process_read_socket: h323chan accepted/connected fd=3 .Jan 14 15:28:56.436 CET: h323chan_chn_process_read_socket .Jan 14 15:28:56.436 CET: h323chan_chn_process_read_socket: fd=4 of type ACCEPTED has data .Jan 14 15:28:56.436 CET: h323chan_chn_process_read_socket: h323chan accepted/connected fd=4 .Jan 14 15:28:56.436 CET: h323chan_chn_process_read_socket .Jan 14 15:28:56.436 CET: h323chan_chn_process_read_socket: fd=4 of type ACCEPTED has data .Jan 14 15:28:56.436 CET: h323chan_chn_process_read_socket: h323chan accepted/connected fd=4 Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0]
Re: [OSL | CCIE_Voice] BACD Voip peers
Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a timestamp:n/a
[OSL | CCIE_Voice] Sub-Pub failover HA Test
I do not have two CCM to test this scenario but is there anything to watch out for when you failover Sub(Primary) to Pub(secondary) and then back to Sub again? E.g.your whole configs might just be wiped out or remote site phones not functioning after you failover back to your Sub [image: [Eek!]] )
Re: [OSL | CCIE_Voice] BACD Voip peers
yup. Gatekeeper looks at g711ulaw as 2 (64k) call legs for a total of 128. Thanks, ryan Trauernicht On Wed, Jan 14, 2009 at 8:34 AM, Chris Parker cpar...@cparker.us wrote: Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a timestamp:n/a
Re: [OSL | CCIE_Voice] MOH Issue
Hi Alex: When you have enabled G729 in IPVMSA CCM instructs the IP Phones to join a specific multicast group, it depends on what you have configured in the MOH server, in the case you are talking about, you have configured to increment IP Address, then the IP Phones will join the multicast group 239.1.1.3, no matter the moh from the flash is G711, the IP Phones will try to join the group, and due to the fact you have configured multicast moh with that IP address and the specific port with route to the Voice VLAN Ip address and the Loopback IP address, the result will be that MOH will be streaming to the IP Phones in site B. Regards, Jose From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Alex Sent: Miércoles, Enero 14, 2009 8:59 AM To: Vik Malhi Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue Vik, AFAIK if G.729 is enabled in IPVMSA and MOH server is in HQ DP (which allows only G.729 to Site B) then the following happens: - CCM instructs Site B phones to join mcast group 239.1.1.3 which is G.729 MOH -Site B router streams only G.711 MOH from flash to 239.1.1.1 http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1043334 Expected result: no MOH on Site B phones If G.729 is not enabled for IPVMSA and MOH server is in HQ DP (which allows only G.729 to Site B), then: - CCM cannot find a suitable mcast group for Site B phones to join Expected result: no MOH on Site B phones If G.729 is not enabled for IPVMSA and MOH server is in G.711-only DP (which allows G.711 to Site B), then: - CCM instructs Site B phones to join mcast group 239.1.1.1 which is G.711 ulaw MOH -Site B router streams only G.711 MOH from flash to 239.1.1.1 Expected result: MOH plays on Site B phones Do I miss something here? Rgds Alex - Original Message - From: Vik Malhi mailto:vma...@ipexpert.com To: kamal yousaf mailto:lovingprin...@gmail.com ; Christian Hennrich mailto:christian.hennr...@intact-is.com Cc: ccie_voice@onlinestudylist.com ; Kumar,Narinder mailto:narinder.ku...@uxcg.com.au Sent: Wednesday, January 14, 2009 6:39 AM Subject: Re: [OSL | CCIE_Voice] MOH Issue 239.1.1.1 port 16384 is always reserved for g711u for Audio Source 1, 239.1.1.2 for g711alaw, 239.1.1.3 for g729 and 239.1.1.4 for Wideband. Since the BR1 DP is communicating with the MOH server using g729 (or should I say CCM thinks g729 is the negotiated codec whereas in realty we are spoofing from the remote site router) CallManager will increment on port # or ip address. In your case MOH---BR1 DP uses g729 and you increment on ip address. You should have G729 allowed in IP Voice Media Streaming App for this to work. Everything on CallManager should be configured as normal- without enabling g729 will cause the MOH sourced from the flash to fail. You need the commands as shown below. Call-manager-fallback Moh filename ( Moh file in flash no multicast moh 239.1.1.1 port 16384 route x.x.x.x multicast moh 239.1.1.3 port 16384 route x.x.x.x Note - you cannot change multicast ip addresses on the fly and so you must delete the first command (incorrect multicast cmd). -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: kamal yousaf lovingprin...@gmail.com Date: Wed, 14 Jan 2009 16:13:35 +1100 To: Christian Hennrich christian.hennr...@intact-is.com Cc: ccie_voice@onlinestudylist.com, Kumar, Narinder narinder.ku...@uxcg.com.au Subject: Re: [OSL | CCIE_Voice] MOH Issue Alex, Regarding your comment, If MOH server uses 239.1.1.3 to stream G729 to remote phones, shouldn't we enable G.729 for MOH ? Only exception is using transcoder but since it can't be used for multicast,don't we have to enable G.729 ? Alex schrieb: Your mcast group IP@ in below debug is 239.1.1.3 The same group IP@ should be configured on SiteB router. No need
Re: [OSL | CCIE_Voice] BACD Voip peers
Hi Chris: The problem is that the router that is registered to the GK sends an ARQ to the GK, and due to the fact you have configured bandwidth total with a value less than 128k the call is rejected. The question is, why, if we are using a dial peer with session target a loopback IP address, does it send an ARQ to the GK. I was reading a lot trying to find the answer, and it is not a bug, it is the normal behaviour, the recommendation for a one zone GK when yor are required to do CAC is to configure at least 128k, when you have more than one zone, you have to use the interzone command, and generally, when you have a single zone there is no CAC requirement. Maybe Vik have another point of view, but that is what I found when I saw this problem first time. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Miércoles, Enero 14, 2009 9:34 AM To: Vik Malhi Cc: Jose Gregorio Linero (jlinero); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- -- *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a timestamp:n/a
Re: [OSL | CCIE_Voice] MOH Issue
239.1.1.3 configured as base IP for MOH server, multicast moh 239.1.1.3 on SRST router and no G.729 for IPVMSA is what I was talking about. Rgds Alex - Original Message - From: kamal yousaf To: Alex Cc: ccie_voice@onlinestudylist.com Sent: Wednesday, January 14, 2009 3:10 PM Subject: Re: [OSL | CCIE_Voice] MOH Issue That means you will use 239.1.1.1 as Multicast IP on your SRST router and NOT 239.1.1.3 as your previous email suggests. Pls Correct me if i am wrong ! On Thu, Jan 15, 2009 at 12:44 AM, Alex alex.arsen...@gmail.com wrote: Kamal, According to task wording below: Between Site A and Site B only g729 allowed and Site B will receive multicast MOH from router flash, no multicast traffic allowed between Ste A and SiteB - I take it as MOH from Site B router flash for Site B IP phones can actually use g711 and wouldn't bother enabling g729 for MOH. This requires placing MOH servers in G.711-only region/DP as per http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a00803f8950.shtml which I haven't mentioned in my previous email. Rgds Alex - Original Message - From: kamal yousaf To: Christian Hennrich Cc: Alex ; ccie_voice@onlinestudylist.com ; Kumar, Narinder Sent: Wednesday, January 14, 2009 5:13 AM Subject: Re: [OSL | CCIE_Voice] MOH Issue Alex, Regarding your comment, If MOH server uses 239.1.1.3 to stream G729 to remote phones, shouldn't we enable G.729 for MOH ? Only exception is using transcoder but since it can't be used for multicast,don't we have to enable G.729 ? Alex schrieb: Your mcast group IP@ in below debug is 239.1.1.3 The same group IP@ should be configured on SiteB router. No need to enable G.729 for MoH - if you ticked increment on IP address that's probably why the group IP@ got changed. Rgds Alex *From:* Kumar, Narinder mailto:narinder.ku...@uxcg.com.au *To:* ccie_voice@onlinestudylist.com mailto:ccie_voice@onlinestudylist.com *Sent:* Tuesday, January 13, 2009 12:36 PM *Subject:* [OSL | CCIE_Voice] MOH Issue Quick Que on MOH CCM running multicast MOH. Between Site A and Site B only g729 allowed SiteA will receive multicast MOH . Site B will receive multicast MOH from router flash, no multicast traffic allowed between Ste A and SiteB. The way I do this question is Configure the MOH source file and tick multicast and play continuously Enable multicast on the MRG and MOH server Change the ip voice media service parameter to allow both g711 and g729 Site A works without any issue Site B Configuration: Call-manager-fallback Moh filename ( Moh file in flash) multicast moh 239.1.1.1 port 16384 route x.x.x.x MOH from site B doesn't work , what am I missing here ? *** debug ccm-manager music-on-hold all ** an 13 13:13:30.023: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:30.023: moh_process_ccb: dstadr 0.0.0.0, callid 18, port 0, codec 65535, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:30.023: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:30.079: moh_process_ccb: dstadr 142.102.65.6, callid 18, port 23552, codec 5, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:30.079: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:31.391: %ISDN-6-CONNECT: Interface Serial0/1/0:2 is now connected to 911 N/A *Jan 13 13:13:31.395: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:31.395: moh_process_ccb: dstadr 142.102.65.6, callid 18, port 23552, codec 5, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:31.399: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.103: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.119: moh_update_rtp: callID 17 dstCallID 18 *Jan 13 13:13:33.139: moh_process_ccb: dstadr 239.1.1.3, callid 18, port 16384, codec 16, moh_en 0, moh_addr 0.0.0.0 *Jan 13 13:13:33.139:
Re: [OSL | CCIE_Voice] BACD Voip peers
Hi Cyrus: I could not find any other solution, is the way it works, when is one zone and you have an specific CAC requirement, I would suggest to use the interzone command, however for one zone it does nothing. Regards, Jose From: Cyrus [mailto:cyrus@gmail.com] Sent: Miércoles, Enero 14, 2009 11:15 AM To: Jose Gregorio Linero (jlinero) Cc: Chris Parker; Vik Malhi; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Hey , If u configure 128 as total bandwidth your call would not going through. BACD needs at least 144 to work properly. The reason is with BACD u have one 16k call (if using G729 over wan) and one 128K call (this is caused by CME ARQ to GK) The question is if requirement be like that : -the minimum configuration lines to GK work properly - so 1 zone should be used -use cac to limit your call to sth less than 144 - run BACD on CME Is there any way to accomplish this? I couldn't find a way myself. It would be great if someone comes up with new idea to do this On Thu, Jan 15, 2009 at 2:39 AM, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: Hi Chris: The problem is that the router that is registered to the GK sends an ARQ to the GK, and due to the fact you have configured bandwidth total with a value less than 128k the call is rejected. The question is, why, if we are using a dial peer with session target a loopback IP address, does it send an ARQ to the GK. I was reading a lot trying to find the answer, and it is not a bug, it is the normal behaviour, the recommendation for a one zone GK when yor are required to do CAC is to configure at least 128k, when you have more than one zone, you have to use the interzone command, and generally, when you have a single zone there is no CAC requirement. Maybe Vik have another point of view, but that is what I found when I saw this problem first time. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Miércoles, Enero 14, 2009 9:34 AM To: Vik Malhi Cc: Jose Gregorio Linero (jlinero); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- -- *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500
Re: [OSL | CCIE_Voice] BACD Voip peers
Hey , If u configure 128 as total bandwidth your call would not going through. BACD needs at least 144 to work properly. The reason is with BACD u have one 16k call (if using G729 over wan) and one 128K call (this is caused by CME ARQ to GK) The question is if requirement be like that : -the minimum configuration lines to GK work properly - so 1 zone should be used -use cac to limit your call to sth less than 144 - run BACD on CME Is there any way to accomplish this? I couldn't find a way myself. It would be great if someone comes up with new idea to do this On Thu, Jan 15, 2009 at 2:39 AM, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: Hi Chris: The problem is that the router that is registered to the GK sends an ARQ to the GK, and due to the fact you have configured bandwidth total with a value less than 128k the call is rejected. The question is, why, if we are using a dial peer with session target a loopback IP address, does it send an ARQ to the GK. I was reading a lot trying to find the answer, and it is not a bug, it is the normal behaviour, the recommendation for a one zone GK when yor are required to do CAC is to configure at least 128k, when you have more than one zone, you have to use the interzone command, and generally, when you have a single zone there is no CAC requirement. Maybe Vik have another point of view, but that is what I found when I saw this problem first time. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Miércoles, Enero 14, 2009 9:34 AM To: Vik Malhi Cc: Jose Gregorio Linero (jlinero); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- -- *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a timestamp:n/a -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] BACD Voip peers
Thanks Jose, Strange stuff to say the least... My GK config is multizone so I will adjust my bandwidth statements accordingly. Chris Jose Gregorio Linero (jlinero) wrote: Hi Chris: The problem is that the router that is registered to the GK sends an ARQ to the GK, and due to the fact you have configured bandwidth total with a value less than 128k the call is rejected. The question is, why, if we are using a dial peer with session target a loopback IP address, does it send an ARQ to the GK. I was reading a lot trying to find the answer, and it is not a bug, it is the normal behaviour, the recommendation for a one zone GK when yor are required to do CAC is to configure at least 128k, when you have more than one zone, you have to use the interzone command, and generally, when you have a single zone there is no CAC requirement. Maybe Vik have another point of view, but that is what I found when I saw this problem first time. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Miércoles, Enero 14, 2009 9:34 AM To: Vik Malhi Cc: Jose Gregorio Linero (jlinero); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- -- *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a timestamp:n/a
[OSL | CCIE_Voice] CCIE Voice Required - Germany - Permanent
Good Afternoon. Apologies for the mass e-mail, however, I have a senior position in Germany that I am actively sourcing candidates for. I am recruiting for an International, award winning Cisco Gold Partner who is currently developing their IPT and Unified Communications division with a Senior IPT/UC Consultant who has passed their CCVP (minimum) but preferably their CCIE Voice. The team is constantly growing and we are constantly looking for IPT consultants in Germany and The Netherlands. My client is looking for a seasoned consultant who is able to lead teams of engineers through the pre sales process, manage network designs, be the on site delivery manager and act as an on site billable consultant. You must speak both German and English and be able to travel through Germany frequently. Please send your CV to me and we can speak further regarding the project then. Regards David David Clark IP Communications, Practice Manager NP Group 350 Euston Road. Regents Place. London. NW1 3AX Switchboard: +44 (0) 20 7953 Direct Line: +44 (0) 20 7953 0039 Mobile: +44 (0) 7779124559 Email: da...@npuk.com http://www.linkedin.com/in/davidclarknp http://www.linkedin.com/in/davidclarknp Please consider the environment before printing this e-mail Confidential Privileged Information may be contained in this message attachments are intended for the addressee only. If you are not the addressee or not responsible for delivery of the message to such person, you may not copy or deliver this message. In such case, destroy this message and kindly notify the sender by reply email. Internet e-mails are not secure, opinions, conclusions and other information in this message do not necessarily represent those of NP Group.
Re: [OSL | CCIE_Voice] MOH Issue
The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM ³thinks² that the MOH server back in HQ is active and the stream is g729. But I guess that¹s the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM ³thinks² is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Alex alex.arsen...@gmail.com Date: Wed, 14 Jan 2009 13:58:36 - To: Vik Malhi vma...@ipexpert.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue Vik, AFAIK if G.729 is enabled in IPVMSA and MOH server is in HQ DP (which allows only G.729 to Site B) then the following happens: - CCM instructs Site B phones to join mcast group 239.1.1.3 which is G.729 MOH -Site B router streams only G.711 MOH from flash to 239.1.1.1 http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guid e09186a00802d1c31.html#wp1043334 Expected result: no MOH on Site B phones If G.729 is not enabled for IPVMSA and MOH server is in HQ DP (which allows only G.729 to Site B), then: - CCM cannot find a suitable mcast group for Site B phones to join Expected result: no MOH on Site B phones If G.729 is not enabled for IPVMSA and MOH server is in G.711-only DP (which allows G.711 to Site B), then: - CCM instructs Site B phones to join mcast group 239.1.1.1 which is G.711 ulaw MOH -Site B router streams only G.711 MOH from flash to 239.1.1.1 Expected result: MOH plays on Site B phones Do I miss something here? Rgds Alex - Original Message - From: Vik Malhi mailto:vma...@ipexpert.com To: kamal yousaf mailto:lovingprin...@gmail.com ; Christian Hennrich mailto:christian.hennr...@intact-is.com Cc: ccie_voice@onlinestudylist.com ; Kumar,Narinder mailto:narinder.ku...@uxcg.com.au Sent: Wednesday, January 14, 2009 6:39 AM Subject: Re: [OSL | CCIE_Voice] MOH Issue 239.1.1.1 port 16384 is always reserved for g711u for Audio Source 1, 239.1.1.2 for g711alaw, 239.1.1.3 for g729 and 239.1.1.4 for Wideband. Since the BR1 DP is communicating with the MOH server using g729 (or should I say CCM ³thinks² g729 is the negotiated codec whereas in realty we are spoofing from the remote site router) CallManager will increment on port # or ip address. In your case MOH---BR1 DP uses g729 and you increment on ip address. You should have G729 allowed in IP Voice Media Streaming App for this to work. Everything on CallManager should be configured as normal- without enabling g729 will cause the MOH sourced from the flash to fail. You need the commands as shown below. Call-manager-fallback Moh filename ( Moh file in flash no multicast moh 239.1.1.1 port 16384 route x.x.x.x multicast moh 239.1.1.3 port 16384 route x.x.x.x Note you cannot change multicast ip addresses on the fly and so you must delete the first command (incorrect multicast cmd). -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: kamal yousaf lovingprin...@gmail.com Date: Wed, 14 Jan 2009 16:13:35 +1100 To: Christian Hennrich christian.hennr...@intact-is.com Cc: ccie_voice@onlinestudylist.com, Kumar, Narinder narinder.ku...@uxcg.com.au Subject: Re: [OSL | CCIE_Voice] MOH Issue Alex, Regarding your comment, If MOH server uses 239.1.1.3 to stream G729 to remote phones, shouldn't we enable G.729 for MOH ? Only exception is using transcoder but
Re: [OSL | CCIE_Voice] BACD Voip peers
Ok- Jose got it right and this is what he was about to say (right?) What is the difference between these two dial-peers on a gateway registered to GK? Dial-peer voice 1 voip destination-pattern 1... session target ras ... AND Dial-peer voice 2 voip destination-pattern 1... session target ipv4:Loopback0 IP Address ... When peer 1 is used an ARQ is sent to GK- it contains a bandwdidth Request and also a called # that needs resolving. When peer 2 is used an ARQ is also sent to the GK since this is also a VOIP call on a gateway to a gatkeeper- it contains just a bandwidth request though since we already did the resolution on the gateway (Lo0). With the B-ACD call we all know that we are sending the call to himself- the Loopback interface- as a workaround to invoke service aa on the inbound voip dial-peer. However the gateway/gk is master/slave and no VOIP call can take place without GK's authorization. The gateway doesn't cross-reference locally configured ip addresses for every session target. So even though the call is local GK still needs to accept the bandwidth request. And this is where bandwidth total ... 128 would hurt you since the B-ACD call will always use g711ulaw and the INTRA-bandwidth required is going to be 128kbps. Bandwidth total and B-ACD are dangerous for this reason. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Chris Parker cpar...@cparker.us Date: Wed, 14 Jan 2009 09:34:13 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Jose Gregorio Linero (jlinero) jlin...@cisco.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4 (67) [50/0/4.0] tx:0/0/0ms None noise:0dBm acom:0dBm long duration call detected:n long dur callduration :n/a timestamp:n/a128F : 71 125594800ms.54 +-1 +30 pid:3501 Originate 3500 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long dur callduration :n/a
[OSL | CCIE_Voice] For MLP, is TS required?
I understand that when configuring LLQ w/ FR, TS is required. If we configured FRF.12, TS is also required. Several questions; 1. If we configure MLP alone, is TS required? I was under the assumption that it's not required. Going by Volume 3 L5 Q43, the solutions doesn't have TS configured when asked to configure fragmentation using MLP w/ LLQ. However, when I configure MLP on my router w/ out TS, I get the following error message- Jan 14 16:46:39.543: %FR-3-MLPOFR_ERROR: MLPoFR not configured properly on Link Virtual-Access2 Bundle Virtual-Access3 :Frame Relay traffic shaping must be enabled 2. If we configure MLP w/ LLQ, is TS required? Vik, can you provide your insight?
Re: [OSL | CCIE_Voice] MOH Issue
Hi Vik, My response/querry in between lines to your response please...I hv have Q's??? --- On Wed, 1/14/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] MOH Issue To: Alex alex.arsen...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: Wednesday, January 14, 2009, 11:18 PM The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1 .with this I beleive we will only have G711 MOH stream only ??? Are we supposed to set on SRST router to use 239.1.1.3then the stream will be G729 or still G711 And what should be set on IPVMA Service Parameter just G711 or both OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. ---with this tte strea mwill be G729 ... And what should be set on IPVMA Service Parameter just G711 or both The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM “thinks” that the MOH server back in HQ is active and the stream is g729. But I guess that’s the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM “thinks” is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Alex alex.arsen...@gmail.com Date: Wed, 14 Jan 2009 13:58:36 - To: Vik Malhi vma...@ipexpert.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue Vik, AFAIK if G.729 is enabled in IPVMSA and MOH server is in HQ DP (which allows only G.729 to Site B) then the following happens: - CCM instructs Site B phones to join mcast group 239.1.1.3 which is G.729 MOH -Site B router streams only G.711 MOH from flash to 239.1.1.1 http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1043334 Expected result: no MOH on Site B phones If G.729 is not enabled for IPVMSA and MOH server is in HQ DP (which allows only G.729 to Site B), then: - CCM cannot find a suitable mcast group for Site B phones to join Expected result: no MOH on Site B phones If G.729 is not enabled for IPVMSA and MOH server is in G.711-only DP (which allows G.711 to Site B), then: - CCM instructs Site B phones to join mcast group 239.1.1.1 which is G.711 ulaw MOH -Site B router streams only G.711 MOH from flash to 239.1.1.1 Expected result: MOH plays on Site B phones Do I miss something here? Rgds Alex - Original Message - From: Vik Malhi mailto:vma...@ipexpert.com To: kamal yousaf mailto:lovingprin...@gmail.com ; Christian Hennrich mailto:christian.hennr...@intact-is.com Cc: ccie_voice@onlinestudylist.com ; Kumar,Narinder mailto:narinder.ku...@uxcg.com.au Sent: Wednesday, January 14, 2009 6:39 AM Subject: Re: [OSL | CCIE_Voice] MOH Issue 239.1.1.1 port 16384 is always reserved for g711u for Audio Source 1, 239.1.1.2 for g711alaw, 239.1.1.3 for g729 and 239.1.1.4 for Wideband. Since the BR1 DP is communicating with the MOH server using g729 (or should I say CCM “thinks” g729 is the negotiated codec whereas in realty we are spoofing from the remote site router) CallManager will increment on port # or ip address. In your case MOH---BR1 DP uses g729 and you increment on ip address. You should have G729 allowed in IP Voice Media Streaming App for this to work. Everything on CallManager should be configured as normal- without enabling g729 will cause the MOH sourced from the flash to fail. You need the commands as shown below. Call-manager-fallback Moh filename ( Moh file in flash no multicast moh 239.1.1.1 port 16384 route x.x.x.x multicast moh 239.1.1.3 port 16384 route x.x.x.x Note – you cannot change multicast ip addresses on the fly and so you must delete the first command (incorrect multicast cmd). -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The
[OSL | CCIE_Voice] MLP layer 2 overhead
Can anyone tell for certain if MLP with FR is 13 bytes for overhead on layer 2 or is it 13 (MLP) + 4 (FR)? Page 33 on SRND for QOS only said 13 bytes for MLP (PPP). It doesnt say it includes FR. Vik can you comment on that? You WAN video I believe said it does, but just wanting to make sure. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] MOH Issue
If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.comwrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM ³thinks² that the MOH server back in HQ is active and the stream is g729. But I guess that¹s the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM ³thinks² is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] marking on routers
So Vik, as Ryan pinted out ...do we still need to mark the traffic on router using class-map/policy-map etc eben though we are marking on dial-peers please --- On Sat, 1/10/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] marking on routers To: Ryan Trauernicht ryanstudyvo...@gmail.com, Jose Gregorio Linero (jlinero) jlin...@cisco.com Cc: ccie_voice@onlinestudylist.com, Majdi Harb majdi.h...@gmail.com Date: Saturday, January 10, 2009, 8:43 AM Correct. Unless there is another reason why you need to turn QoS on the switch (e.g. Policer or Tx queues cos mapping) then you can simply keep QoS turned off and the original marking will be preserved. And you would have to modify the Enterprise params, Service Params, potentially CTI and IPVMSA params. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Fri, 9 Jan 2009 16:46:17 -0600 To: Jose Gregorio Linero (jlinero) jlin...@cisco.com Cc: ccie_voice@onlinestudylist.com, Majdi Harb majdi.h...@gmail.com Subject: Re: [OSL | CCIE_Voice] marking on routers I guess thinking about this alittle more... if you do not have qos turned on at the switch level (aka trusting everything). Do you even need to mark on the ingress of the ports or mark at all for that matter. CM is marking packets correctly. If you put the ip qos dscp cs3 sign command on the dial-peers that will mark packets correctly for the sip and h323 traffic. mgcp ip qos dscp cs3 sign for mgcp. Why would you really need to mark traffic? On Fri, Jan 9, 2009 at 3:16 PM, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: Yes Ryan, that is wright From: Ryan Trauernicht [mailto:ryanstudyvo...@gmail.com] Sent: Viernes, Enero 09, 2009 3:40 PM To: Jose Gregorio Linero (jlinero) Cc: Majdi Harb; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] marking on routers Why would you need both directions. If you mark them on the ingress of the fast ethernet on each location you dont need the other direction. On Fri, Jan 9, 2009 at 12:42 PM, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: Hi Majdi: Actually you have to do it in both directions for both TCP and UDP. Regards, Jose From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Majdi Harb Sent: Viernes, Enero 09, 2009 1:14 PM To: ccie_vo...@onlinestudylist.com Subject: [OSL | CCIE_Voice] marking on routers Hi can someone please correct me in the following, i want to mark sccp, h323, mgcp and sip traffic to cs3 on sites routers, i've done the following on HQ router: ip access-list extended CONTROL permit tcp any range 2000 2002 any permit udp any eq 2427 any permit tcp any eq 2428 any permit tcp any any eq 1720 permit tcp any any range 11000 11999 permit udp any any eq 1719 permit udp any any eq 1718 permit tdp any any eq 5060 permit udp any any eq 5060 class-map match-any SIGNAL match access-group name CONTROL policy-map IPPHONE class SIGNAL set ip dscp cs3 int f0/0.101 service-policy input IPPHONE i'm not sure if i'm using the right direction in the above matches, what if i have SIP FXS on HQ router, is (permit tdp any any eq 5060 and permit udp any any eq 5060) correct or it should be (permit tdp any eq 5060 any and permit udp any eq 5060 any) please correct me... or am i really off on this ..? Regards, majdi
Re: [OSL | CCIE_Voice] MLP layer 2 overhead
I guess on top of that if you do MLP with LFI is that the 13 bytes or is just MLP 13bytes of overhead. If you add in LFI how much layer 2 overhead does that add? On Wed, Jan 14, 2009 at 12:13 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Can anyone tell for certain if MLP with FR is 13 bytes for overhead on layer 2 or is it 13 (MLP) + 4 (FR)? Page 33 on SRND for QOS only said 13 bytes for MLP (PPP). It doesnt say it includes FR. Vik can you comment on that? You WAN video I believe said it does, but just wanting to make sure. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] MOH Issue
Can you post the output of debug ccm-m music all. Check that the MOH is being active using debug ephone moh. Dead air is better than tone. CCM thinks everything is working so the problem is lying in the spoofing part. I don¹t think it is anything to do with your MOH file- have you tried it with the music-on-hold.au that is provided? -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Wed, 14 Jan 2009 12:15:47 -0600 To: Antonio McCarver amccar...@cciequest.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.com wrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM ³thinks² that the MOH server back in HQ is active and the stream is g729. But I guess that¹s the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM ³thinks² is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] BACD Voip peers
Yeah. 128 will be enough for calls made locally from the CME phones though. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Cyrus cyrus@gmail.com Date: Thu, 15 Jan 2009 03:15:07 +1100 To: Jose Gregorio Linero (jlinero) jlin...@cisco.com Cc: Chris Parker cpar...@cparker.us, Vik Malhi vma...@ipexpert.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Hey , If u configure 128 as total bandwidth your call would not going through. BACD needs at least 144 to work properly. The reason is with BACD u have one 16k call (if using G729 over wan) and one 128K call (this is caused by CME ARQ to GK) The question is if requirement be like that : -the minimum configuration lines to GK work properly - so 1 zone should be used -use cac to limit your call to sth less than 144 - run BACD on CME Is there any way to accomplish this? I couldn't find a way myself. It would be great if someone comes up with new idea to do this On Thu, Jan 15, 2009 at 2:39 AM, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: Hi Chris: The problem is that the router that is registered to the GK sends an ARQ to the GK, and due to the fact you have configured bandwidth total with a value less than 128k the call is rejected. The question is, why, if we are using a dial peer with session target a loopback IP address, does it send an ARQ to the GK. I was reading a lot trying to find the answer, and it is not a bug, it is the normal behaviour, the recommendation for a one zone GK when yor are required to do CAC is to configure at least 128k, when you have more than one zone, you have to use the interzone command, and generally, when you have a single zone there is no CAC requirement. Maybe Vik have another point of view, but that is what I found when I saw this problem first time. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Miércoles, Enero 14, 2009 9:34 AM To: Vik Malhi Cc: Jose Gregorio Linero (jlinero); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- -- *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67
Re: [OSL | CCIE_Voice] For MLP, is TS required?
TS is required in both scenarios is the answer to your question. And it would be VERY unlikely that you would be asked to configure LFI without LLQ (in my humble opinion). -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: wafers44 wafer...@gmail.com Date: Wed, 14 Jan 2009 12:04:00 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com Subject: For MLP, is TS required? Jan 14 16:46:39.543: %FR-3-MLPOFR_ERROR: MLPoFR not configured properly on Link Virtual-Access2 Bundle Virtual-Access3 :Frame Relay traffic shaping must be enabled
Re: [OSL | CCIE_Voice] For MLP, is TS required?
Thanks for the clarification. On Wed, Jan 14, 2009 at 12:49 PM, Vik Malhi vma...@ipexpert.com wrote: TS is required in both scenarios is the answer to your question. And it would be VERY unlikely that you would be asked to configure LFI without LLQ (in my humble opinion). -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: wafers44 wafer...@gmail.com Date: Wed, 14 Jan 2009 12:04:00 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com Subject: For MLP, is TS required? Jan 14 16:46:39.543: %FR-3-MLPOFR_ERROR: MLPoFR not configured properly on Link Virtual-Access2 Bundle Virtual-Access3 :Frame Relay traffic shaping must be enabled
Re: [OSL | CCIE_Voice] BACD Voip peers
Vic, Loopback solution is used as workaround to force CME to kick in the Xcoder.As we can see with G711 ,there is no need for loopback solution. But BACD work properly ,we need at least 144k BW on GK, please correct me if I'm wrong 128+16 for incoming calls from WAN On Thu, Jan 15, 2009 at 4:56 AM, Vik Malhi vma...@ipexpert.com wrote: Ok- Jose got it right and this is what he was about to say (right?) What is the difference between these two dial-peers on a gateway registered to GK? Dial-peer voice 1 voip destination-pattern 1... session target ras ... AND Dial-peer voice 2 voip destination-pattern 1... session target ipv4:Loopback0 IP Address ... When peer 1 is used an ARQ is sent to GK- it contains a bandwdidth Request and also a called # that needs resolving. When peer 2 is used an ARQ is also sent to the GK since this is also a VOIP call on a gateway to a gatkeeper- it contains just a bandwidth request though since we already did the resolution on the gateway (Lo0). With the B-ACD call we all know that we are sending the call to himself- the Loopback interface- as a workaround to invoke service aa on the inbound voip dial-peer. However the gateway/gk is master/slave and no VOIP call can take place without GK's authorization. The gateway doesn't cross-reference locally configured ip addresses for every session target. So even though the call is local GK still needs to accept the bandwidth request. And this is where bandwidth total ... 128 would hurt you since the B-ACD call will always use g711ulaw and the INTRA-bandwidth required is going to be 128kbps. Bandwidth total and B-ACD are dangerous for this reason. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Chris Parker cpar...@cparker.us Date: Wed, 14 Jan 2009 09:34:13 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Jose Gregorio Linero (jlinero) jlin...@cisco.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Voip peers I have had problems getting BACD to dial using voip from the phones on CME. I can dial into the BACD fine from the PSTN, but not from my IP phones. Here is my config: voice service voip allow-connections h323 to h323 dial-peer voice 3500 pots service aa incoming called-number 3500 port 0/2/0:23 ! dial-peer voice 3501 voip destination-pattern 3500 session target ipv4:172.16.101.3 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3502 voip service aa incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Eveytime I call the number I get no circuit 128A : 67 12560ms.53 +-1 +3280 pid:20004 Answer 3004 dur 00:00:00 tx:0/0 rx:0/0 22 (no circuit (34)) Telephony 50/0/4
Re: [OSL | CCIE_Voice] MOH Issue
home lab that I pulled the sample audio from the MOH folder. I set it to G711only and change the IP address to 239.1.1.1 and all is well. On Wed, Jan 14, 2009 at 12:44 PM, Vik Malhi vma...@ipexpert.com wrote: Can you post the output of debug ccm-m music all. Check that the MOH is being active using debug ephone moh. Dead air is better than tone. CCM thinks everything is working so the problem is lying in the spoofing part. I don't think it is anything to do with your MOH file- have you tried it with the music-on-hold.au that is provided? -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Ryan Trauernicht ryanstudyvo...@gmail.com *Date: *Wed, 14 Jan 2009 12:15:47 -0600 *To: *Antonio McCarver amccar...@cciequest.com *Cc: *ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] MOH Issue If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.com wrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM thinks that the MOH server back in HQ is active and the stream is g729. But I guess that's the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM thinks is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
[OSL | CCIE_Voice] anyway to remove sdspfarm config without doing no telephony ?
Hi, every time I want to change sdspfarm config , I have to do no telephony and put everything back again. Is there any way to remove sdspfarm tag command, in case I want to change Mac address ? Jeremy
Re: [OSL | CCIE_Voice] MOH Issue
My fault monday mistake! I had it based on port based and not IP based. All working now. On Wed, Jan 14, 2009 at 1:16 PM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: home lab that I pulled the sample audio from the MOH folder. I set it to G711only and change the IP address to 239.1.1.1 and all is well. On Wed, Jan 14, 2009 at 12:44 PM, Vik Malhi vma...@ipexpert.com wrote: Can you post the output of debug ccm-m music all. Check that the MOH is being active using debug ephone moh. Dead air is better than tone. CCM thinks everything is working so the problem is lying in the spoofing part. I don't think it is anything to do with your MOH file- have you tried it with the music-on-hold.au that is provided? -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Ryan Trauernicht ryanstudyvo...@gmail.com *Date: *Wed, 14 Jan 2009 12:15:47 -0600 *To: *Antonio McCarver amccar...@cciequest.com *Cc: *ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] MOH Issue If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.com wrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM thinks that the MOH server back in HQ is active and the stream is g729. But I guess that's the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM thinks is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] Layer 2 overhead
FR = 4 bytes FRF.12 = 8 bytes Agreed. For MLPoFR (w/ or w/out LFI - but in our case we would only be using MLP for LFI) I've been using 4B (FR) + 13B (MLP). Also, in all the IPExpert solution guides for Volume 3 atleast they've been using 4B+13B for MLPoFR On Wed, Jan 14, 2009 at 1:50 PM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] For MLP, is TS required?
Hi: If you are configuring MLPoFR you have to have TS. Regards, Jose From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of wafers44 Sent: Miércoles, Enero 14, 2009 1:04 PM To: ccie_voice@onlinestudylist.com; Vik Malhi Subject: [OSL | CCIE_Voice] For MLP, is TS required? I understand that when configuring LLQ w/ FR, TS is required. If we configured FRF.12, TS is also required. Several questions; 1. If we configure MLP alone, is TS required? I was under the assumption that it's not required. Going by Volume 3 L5 Q43, the solutions doesn't have TS configured when asked to configure fragmentation using MLP w/ LLQ. However, when I configure MLP on my router w/ out TS, I get the following error message- Jan 14 16:46:39.543: %FR-3-MLPOFR_ERROR: MLPoFR not configured properly on Link Virtual-Access2 Bundle Virtual-Access3 :Frame Relay traffic shaping must be enabled 2. If we configure MLP w/ LLQ, is TS required? Vik, can you provide your insight?
Re: [OSL | CCIE_Voice] Layer 2 overhead
Ok good.. FRF.12 w/ FR = 8 FR = 4 FRF.12 = 4 sorry for the confusion. On Wed, Jan 14, 2009 at 1:58 PM, wafers44 wafer...@gmail.com wrote: FR = 4 bytes FRF.12 = 8 bytes Agreed. For MLPoFR (w/ or w/out LFI - but in our case we would only be using MLP for LFI) I've been using 4B (FR) + 13B (MLP). Also, in all the IPExpert solution guides for Volume 3 atleast they've been using 4B+13B for MLPoFR On Wed, Jan 14, 2009 at 1:50 PM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] MOH Issue
Hi Ryan: No it does not, it could be G711. Regards, Jose From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ryan Trauernicht Sent: Miércoles, Enero 14, 2009 1:16 PM To: Antonio McCarver Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.com wrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM ³thinks² that the MOH server back in HQ is active and the stream is g729. But I guess that¹s the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM ³thinks² is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] Layer 2 overhead
MLPoFR is actually either 10 or 11 bytes (my memory is failing me). In the SRND it states MLP is 13 bytes. The 13 bytes I can only imagine is a conservative estimate or is MLPoATM. It certainly is very conservative for MLPoFR. I would clarify with the proctor- I would not use 13 + 4 = 17 bytes. Page 33 of the QoS SRND talks about these values and I would treat the 13 bytes listed for MLP as being appropriate for MLPoFR. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Wed, 14 Jan 2009 13:50:10 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Layer 2 overhead Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Layer 2 overhead
So Vik when it says don't use FRF.12 , that is we got to use MLP LFI in that case how much the payload be 20+4+13 =17 or 20+13 = 33 Kindly let us know... --- On Thu, 1/15/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 15, 2009, 1:42 AM MLPoFR is actually either 10 or 11 bytes (my memory is failing me). In the SRND it states MLP is 13 bytes. The 13 bytes I can only imagine is a conservative estimate or is MLPoATM. It certainly is very conservative for MLPoFR. I would clarify with the proctor- I would not use 13 + 4 = 17 bytes. Page 33 of the QoS SRND talks about these values and I would treat the 13 bytes listed for MLP as being appropriate for MLPoFR. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Wed, 14 Jan 2009 13:50:10 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Layer 2 overhead Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Layer 2 overhead
sorry need to add RTP/UDP/Header 40 too to these vlaues..so shall it be MLP with LFI = 20+40+4+13 =77 or 20+40-+13 =73 --- On Thu, 1/15/09, anil batra anil...@yahoo.com wrote: From: anil batra anil...@yahoo.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com Date: Thursday, January 15, 2009, 1:52 AM So Vik when it says don't use FRF.12 , that is we got to use MLP LFI in that case how much the payload be 20+4+13 =17 or 20+13 = 33 Kindly let us know... --- On Thu, 1/15/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 15, 2009, 1:42 AM MLPoFR is actually either 10 or 11 bytes (my memory is failing me). In the SRND it states MLP is 13 bytes. The 13 bytes I can only imagine is a conservative estimate or is MLPoATM. It certainly is very conservative for MLPoFR. I would clarify with the proctor- I would not use 13 + 4 = 17 bytes. Page 33 of the QoS SRND talks about these values and I would treat the 13 bytes listed for MLP as being appropriate for MLPoFR. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Wed, 14 Jan 2009 13:50:10 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Layer 2 overhead Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Layer 2 overhead
It is 17 for sure. Regards, Shadab CCIE# 22893 (Voice) Technology Solutions Network ~Sent from my NOKIA E61i~ -Original Message- From: anil batra [mailto:anil...@yahoo.com] Sent: Thursday, January 15, 2009 04:23 AM China Standard Time To: Ryan Trauernicht; ccie_voice@onlinestudylist.com; Vik Malhi Subject:Re: [OSL | CCIE_Voice] Layer 2 overhead So Vik when it says don't use FRF.12 , that is we got to use MLP LFI in that case how much the payload be 20+4+13 =17 or 20+13 = 33 Kindly let us know... --- On Thu, 1/15/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 15, 2009, 1:42 AM MLPoFR is actually either 10 or 11 bytes (my memory is failing me). In the SRND it states MLP is 13 bytes. The 13 bytes I can only imagine is a conservative estimate or is MLPoATM. It certainly is very conservative for MLPoFR. I would clarify with the proctor- I would not use 13 + 4 = 17 bytes. Page 33 of the QoS SRND talks about these values and I would treat the 13 bytes listed for MLP as being appropriate for MLPoFR. -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Wed, 14 Jan 2009 13:50:10 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Layer 2 overhead Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Layer 2 overhead
And yes, its 77 w/o compression 39 with compression Regards, Shadab CCIE# 22893 (Voice) Technology Solutions Network ~Sent from my NOKIA E61i~ -Original Message- From: anil batra [mailto:anil...@yahoo.com] Sent: Thursday, January 15, 2009 04:26 AM China Standard Time To: Ryan Trauernicht; ccie_voice@onlinestudylist.com; Vik Malhi Subject:Re: [OSL | CCIE_Voice] Layer 2 overhead sorry need to add RTP/UDP/Header 40 too to these vlaues..so shall it be MLP with LFI = 20+40+4+13 =77 or 20+40-+13 =73 --- On Thu, 1/15/09, anil batra anil...@yahoo.com wrote: From: anil batra anil...@yahoo.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com Date: Thursday, January 15, 2009, 1:52 AM So Vik when it says don't use FRF.12 , that is we got to use MLP LFI in that case how much the payload be 20+4+13 =17 or 20+13 = 33 Kindly let us know... --- On Thu, 1/15/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 15, 2009, 1:42 AM MLPoFR is actually either 10 or 11 bytes (my memory is failing me). In the SRND it states MLP is 13 bytes. The 13 bytes I can only imagine is a conservative estimate or is MLPoATM. It certainly is very conservative for MLPoFR. I would clarify with the proctor- I would not use 13 + 4 = 17 bytes. Page 33 of the QoS SRND talks about these values and I would treat the 13 bytes listed for MLP as being appropriate for MLPoFR. -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Wed, 14 Jan 2009 13:50:10 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Layer 2 overhead Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] MOH Issue
Could you post your sho run on the router that you are sourcing the MOH from? If you are getting dead air, that means your CCM is setup correctly and the issue is pointing to the local MOH configuration. Do you have at least 1 ephone defined? Another note, if you had it working with g711 and just changed the moh / multicast information to reflect g729 you may have to delete the entire call-manager fallback configuration and paste it back in with the g729 information. I have run into this in the past. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jose Gregorio Linero (jlinero) Sent: Wednesday, January 14, 2009 3:07 PM To: Ryan Trauernicht; Antonio McCarver Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue Hi Ryan: No it does not, it could be G711. Regards, Jose From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ryan Trauernicht Sent: Miércoles, Enero 14, 2009 1:16 PM To: Antonio McCarver Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Issue If i set my MOH server to G729 for the remote branch and put a G711 file on the flash with the following commands: moh .wav multicast moh 239.1.1.3 port 16384 route X.X.X.X X.X.X.X I get dead air is that b/c the file type loaded on the flash needs to be g729? On Wed, Jan 14, 2009 at 12:11 PM, Antonio McCarver amccar...@cciequest.com wrote: Hello group, I am at the very beginning stages of my lab prep so please forgive me if this is one of those come on newbie, you should've known that questions. I have read and re-read the MOH section in the CallManager Fundamentals book, and in the CUCM 7.x SRND and I don't see where either went into detail about the different mcast addresses 239.1.1.1, .2, or .3. My question is, where can I look to read up on them and this issue? Amp Quoting Vik Malhi vma...@ipexpert.com: The two solutions work- either you place your MOH server in a g711-always DP and your should set the SRST router to use 239.1.1.1. OR...IF you did but the MOH server in a DP that uses g729 to site B (for whatever reason) then you should set the SRST router to use 239.1.1.3. The MOH file on the flash will be sent out using the same IP Address CCM is telling the phone/gateway to listen. The phone on hold is receiving RTP packets and the payload type will be g711u- however CCM ³thinks² that the MOH server back in HQ is active and the stream is g729. But I guess that¹s the whole idea of spoofing- CCM is not aware of what is going on. The codec CCM ³thinks² is being used and the actual codec are different- but that will not affect the end result. Also- while we are on the topic of sourcing music from the flash- you all should be putting in the command: no mgcp timer receive-rtcp (in the case of an MGCP gateway) -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications.
Re: [OSL | CCIE_Voice] Layer 2 overhead
Vik any reason why the IPExperts lab do 13+4 for MLPoFR? On Wed, Jan 14, 2009 at 2:35 PM, Shadab Abbasi (moabbasi) moabb...@cisco.com wrote: And yes, its 77 w/o compression 39 with compression Regards, Shadab CCIE# 22893 (Voice) Technology Solutions Network ~Sent from my NOKIA E61i~ -Original Message- From: anil batra [mailto:anil...@yahoo.com anil...@yahoo.com] Sent: Thursday, January 15, 2009 04:26 AM China Standard Time To: Ryan Trauernicht; ccie_voice@onlinestudylist.com; Vik Malhi Subject:Re: [OSL | CCIE_Voice] Layer 2 overhead sorry need to add RTP/UDP/Header 40 too to these vlaues..so shall it be MLP with LFI = 20+40+4+13 =77 or 20+40-+13 =73 --- On Thu, 1/15/09, anil batra anil...@yahoo.com wrote: From: anil batra anil...@yahoo.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com Date: Thursday, January 15, 2009, 1:52 AM So Vik when it says don't use FRF.12 , that is we got to use MLP LFI in that case how much the payload be 20+4+13 =17 or 20+13 = 33 Kindly let us know... --- On Thu, 1/15/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Ryan Trauernicht ryanstudyvo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Thursday, January 15, 2009, 1:42 AM MLPoFR is actually either 10 or 11 bytes (my memory is failing me). In the SRND it states MLP is 13 bytes. The 13 bytes I can only imagine is a conservative estimate or is MLPoATM. It certainly is very conservative for MLPoFR. I would clarify with the proctor- I would not use 13 + 4 = 17 bytes. Page 33 of the QoS SRND talks about these values and I would treat the 13 bytes listed for MLP as being appropriate for MLPoFR. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Ryan Trauernicht ryanstudyvo...@gmail.com Date: Wed, 14 Jan 2009 13:50:10 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Layer 2 overhead Reading the SRND and a few other books on the WAN QOS... looks like FR layer 2 is not included in the page 33 statements. Vik can you confirm these are correct for layer 2 byte sizes. MLP in the SRST states it is 13 bytes for layer 2. That actually includes LFI MLP without LFI FR = 10 bytes MLP with LFI and without FR = 13 bytes MLP with LFI FR = 17 bytes FR = 4 bytes FRF.12 = 8 bytes anyone agree or disagree? Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] BACD Voip peers
Vik, Makes complete sense. What if the voip peers used the SIP protocol? In that case I guess the GK would not be involved? Chris Vik Malhi wrote: Ok- Jose got it right and this is what he was about to say (right?) What is the difference between these two dial-peers on a gateway registered to GK? Dial-peer voice 1 voip destination-pattern 1... session target ras ... AND Dial-peer voice 2 voip destination-pattern 1... session target ipv4:Loopback0 IP Address ... When peer 1 is used an ARQ is sent to GK- it contains a bandwdidth Request and also a called # that needs resolving. When peer 2 is used an ARQ is also sent to the GK since this is also a VOIP call on a gateway to a gatkeeper- it contains just a bandwidth request though since we already did the resolution on the gateway (Lo0). With the B-ACD call we all know that we are sending the call to himself- the Loopback interface- as a workaround to invoke service aa on the inbound voip dial-peer. However the gateway/gk is master/slave and no VOIP call can take place without GK's authorization. The gateway doesn't cross-reference locally configured ip addresses for every session target. So even though the call is local GK still needs to accept the bandwidth request. And this is where bandwidth total ... 128 would hurt you since the B-ACD call will always use g711ulaw and the INTRA-bandwidth required is going to be 128kbps. Bandwidth total and B-ACD are dangerous for this reason.
[OSL | CCIE_Voice] NTP CM
Any reason why CM will not keep its NTP clock. I have a local router with the following commands: ntp master 3 ntp source loopback0 (IP address is 192.168.187.1) I have edited the c:\WINNT\System32\Drivers\Etc\ntp.config My file looks like: server 192.168.187.1 # Set Local Clock to Authoritive Time Source fudge 192.168.187.1 stratum 5 # Resets Stratum from default 3 to 5 driftfile C:\WINNT\system32\drivers\etc\ntp.drift # path for drift file I stopped the Network Time Protocol service and I run ntpdate.exe 192.168.187.1 command from the cmd. That sets the clock to sync to NTP router just fine. After I reboot CM it goes back to GMT it looks like. I see it trying to sync but it never does. I have waited over 15mins and nothing. Thanks, Ryan Trauernicht
[OSL | CCIE_Voice] Feeling lost
I'm just starting out in the lab, and already feeling lost. 1. WHERE does the IP Blue client come from? If it has to be purchasedfrom WHERE? 2. I see some posts about connecting a VPN up to run hardware that's local to me up to the Proctor Labs rack, and they mention documentation from Proctor Labs on doing this. Again, WHERE is this documentation? Some guidance definately appreciated. Cliff
Re: [OSL | CCIE_Voice] Feeling lost
Hi Cliff, Finally there are some questions on here that I can answer! ;) 1. IPBlue = http://www.ipblue.com/download.asp?product=vtgo 2. IPExpert Tech FAQs = http://ipexpert.ccieblog.com/2008/11/01/proctor-labs-voice-faq/ http://proctorlabs.com/forum/ for FAQ's on how to connect via EZVPN... Also, when you schedule an online lab and have the rack available there will be comprehensive instructions/config/VPN profile made available before you connect! Hope this helps! Rob
Re: [OSL | CCIE_Voice] Feeling lost
Hmmm. Can't find the setup for a hardware VPN to support physical phones on my end. The IP Expert Tech FAQ's hint at it, but provide no guidance. ProtorLabs forums are completely hosed up and unusable for some reason. Just a bunch of server debug coming to the browser. - Original Message - From: rob To: Cliff McGlamry Cc: Vik Malhi ; ccie_voice@onlinestudylist.com Sent: Wednesday, January 14, 2009 6:35 PM Subject: Re: [OSL | CCIE_Voice] Feeling lost Hi Cliff, Finally there are some questions on here that I can answer! ;) 1. IPBlue = http://www.ipblue.com/download.asp?product=vtgo 2. IPExpert Tech FAQs = http://ipexpert.ccieblog.com/2008/11/01/proctor-labs-voice-faq/ http://proctorlabs.com/forum/ for FAQ's on how to connect via EZVPN... Also, when you schedule an online lab and have the rack available there will be comprehensive instructions/config/VPN profile made available before you connect! Hope this helps! Rob
Re: [OSL | CCIE_Voice] Feeling lost
I'm also having problems with the proctorlabs forums at the moment! Wait until they're back online and then check the FAQ section for the Voice racks... There's a thread on there with an example EZVPN configuration that you can modify depending on which rack you're assigned..
Re: [OSL | CCIE_Voice] BACD Voip peers
The Loopback solution is used for calls from the WAN invoking the AA and also CME phones calling the AA- (144/128). -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Cyrus cyrus@gmail.com Date: Thu, 15 Jan 2009 06:13:45 +1100 To: Vik Malhi vma...@ipexpert.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vic, Loopback solution is used as workaround to force CME to kick in the Xcoder.As we can see with G711 ,there is no need for loopback solution. But BACD work properly ,we need at least 144k BW on GK, please correct me if I'm wrong 128+16 for incoming calls from WAN On Thu, Jan 15, 2009 at 4:56 AM, Vik Malhi vma...@ipexpert.com wrote: Ok- Jose got it right and this is what he was about to say (right?) What is the difference between these two dial-peers on a gateway registered to GK? Dial-peer voice 1 voip destination-pattern 1... session target ras ... AND Dial-peer voice 2 voip destination-pattern 1... session target ipv4:Loopback0 IP Address ... When peer 1 is used an ARQ is sent to GK- it contains a bandwdidth Request and also a called # that needs resolving. When peer 2 is used an ARQ is also sent to the GK since this is also a VOIP call on a gateway to a gatkeeper- it contains just a bandwidth request though since we already did the resolution on the gateway (Lo0). With the B-ACD call we all know that we are sending the call to himself- the Loopback interface- as a workaround to invoke service aa on the inbound voip dial-peer. However the gateway/gk is master/slave and no VOIP call can take place without GK's authorization. The gateway doesn't cross-reference locally configured ip addresses for every session target. So even though the call is local GK still needs to accept the bandwidth request. And this is where bandwidth total ... 128 would hurt you since the B-ACD call will always use g711ulaw and the INTRA-bandwidth required is going to be 128kbps. Bandwidth total and B-ACD are dangerous for this reason. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Chris Parker cpar...@cparker.us Date: Wed, 14 Jan 2009 09:34:13 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Jose Gregorio Linero (jlinero) jlin...@cisco.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to bring a very complicated problem with using the bandwidth total command inside gatekeeper and how it impacts B-ACD. Chris- please make the call to the B-ACD AA from a CME phone and paste the output of debug ras (assuming the router is registered to a gatekeeper). -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: _vma...@ipexpert.com _ Join our free online support and peer group communities: _http://www.IPexpert.com/communities _IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. *From: *Jose Gregorio Linero (jlinero) jlin...@cisco.com *Date: *Tue, 13 Jan 2009 22:03:59 -0500 *To: *Chris Parker cpar...@cparker.us, ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Hi Chris: Is this router registered to a gatekeeper?. Regards, Jose -Original Message- From: Chris Parker [mailto:cpar...@cparker.us] Sent: Tuesday, January 13, 2009 09:56 PM Eastern Standard Time To:
Re: [OSL | CCIE_Voice] BACD Voip peers
Vic, Loppback solution is *only* used as a workaround for invoking Xcoder .Because calling from WAN to CME and then hitting AA would not force CME to involve Xcoder. So as a workaround we direct traffic to the loopback thus CME and Xcoder and then AA will come to play. Here is the proof. GK with setup of G711 . voice translation-rule 800 rule 1 /6667878/ // voice translation-profile INCOMING_GK translate called 1 dial-peer voice 800 voip service aa translation-profile incoming INCOMING_GK incoming called-number 6667878 dtmf-relay h245-alphanumeric codec g711u no vad sh gatekeeper call LocalCallIDAge(secs) BW 138-14 13 128(Kbps) Endpt(s): Alias E.164Addr src EP: GK_TRUNK_12001 CallSignalAddr Port RASSignalAddr Port 140.0.0.1 3345 140.0.0.1 3245 Endpt(s): Alias E.164Addr dst EP: CME6667878 CallSignalAddr Port RASSignalAddr Port 140.0.3.2541720 140.0.3.25449270 Opbviously ARQ does not comes into play adn we need just 128 for G711 . This way Service aa kicking in and work properly with out loopback solution. Cyrus On Thu, Jan 15, 2009 at 11:09 AM, Vik Malhi vma...@ipexpert.com wrote: The Loopback solution is used for calls from the WAN invoking the AA and also CME phones calling the AA- (144/128). -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: *vma...@ipexpert.com * Join our free online support and peer group communities: *http://www.IPexpert.com/communities *IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- *From: *Cyrus cyrus@gmail.com *Date: *Thu, 15 Jan 2009 06:13:45 +1100 *To: *Vik Malhi vma...@ipexpert.com *Cc: *ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] BACD Voip peers Vic, Loopback solution is used as workaround to force CME to kick in the Xcoder.As we can see with G711 ,there is no need for loopback solution. But BACD work properly ,we need at least 144k BW on GK, please correct me if I'm wrong 128+16 for incoming calls from WAN On Thu, Jan 15, 2009 at 4:56 AM, Vik Malhi vma...@ipexpert.com wrote: Ok- Jose got it right and this is what he was about to say (right?) What is the difference between these two dial-peers on a gateway registered to GK? Dial-peer voice 1 voip destination-pattern 1... session target ras ... AND Dial-peer voice 2 voip destination-pattern 1... session target ipv4:Loopback0 IP Address ... When peer 1 is used an ARQ is sent to GK- it contains a bandwdidth Request and also a called # that needs resolving. When peer 2 is used an ARQ is also sent to the GK since this is also a VOIP call on a gateway to a gatkeeper- it contains just a bandwidth request though since we already did the resolution on the gateway (Lo0). With the B-ACD call we all know that we are sending the call to himself- the Loopback interface- as a workaround to invoke service aa on the inbound voip dial-peer. However the gateway/gk is master/slave and no VOIP call can take place without GK's authorization. The gateway doesn't cross-reference locally configured ip addresses for every session target. So even though the call is local GK still needs to accept the bandwidth request. And this is where bandwidth total ... 128 would hurt you since the B-ACD call will always use g711ulaw and the INTRA-bandwidth required is going to be 128kbps. Bandwidth total and B-ACD are dangerous for this reason. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Chris Parker cpar...@cparker.us Date: Wed, 14 Jan 2009 09:34:13 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Jose Gregorio Linero (jlinero) jlin...@cisco.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vik, When I type no gateway and try the call again it goes through. So I must be running into this issue. I do have bandwidth total configured on my GK as well. It is set to 96. I'm guessing if I bump it up to 128 to allow a g711 call it'll work? Chris Vik Malhi wrote: Jose is about to
Re: [OSL | CCIE_Voice] BACD Voip peers
Loopback solution is required for invoking the xcoder for calls from the WAN. However here is the bit I think you might be missing. The loopback solution is also required for Cme phones calling the AA and it is for this latter call I was originally focused on. Vik Malhi - CCIE#13890 Senior Technical Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join IPexpert's Free CCIE Peer Groups Study Communities at www.IPexpert.com/communities On Jan 14, 2009, at 5:49 PM, Cyrus cyrus@gmail.com wrote: Vic, Loppback solution is only used as a workaround for invoking Xcoder .Because calling from WAN to CME and then hitting AA would not force CME to involve Xcoder. So as a workaround we direct traffic to the loopback thus CME and Xcoder and then AA will come to play. Here is the proof. GK with setup of G711 . voice translation-rule 800 rule 1 /6667878/ // voice translation-profile INCOMING_GK translate called 1 dial-peer voice 800 voip service aa translation-profile incoming INCOMING_GK incoming called-number 6667878 dtmf-relay h245-alphanumeric codec g711u no vad sh gatekeeper call LocalCallIDAge(secs) BW 138-14 13 128(Kbps) Endpt(s): Alias E.164Addr src EP: GK_TRUNK_12001 CallSignalAddr Port RASSignalAddr Port 140.0.0.1 3345 140.0.0.1 3245 Endpt(s): Alias E.164Addr dst EP: CME6667878 CallSignalAddr Port RASSignalAddr Port 140.0.3.2541720 140.0.3.25449270 Opbviously ARQ does not comes into play adn we need just 128 for G711 . This way Service aa kicking in and work properly with out loopback solution. Cyrus On Thu, Jan 15, 2009 at 11:09 AM, Vik Malhi vma...@ipexpert.com wrote: The Loopback solution is used for calls from the WAN invoking the AA and also CME phones calling the AA- (144/128). -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video- On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Cyrus cyrus@gmail.com Date: Thu, 15 Jan 2009 06:13:45 +1100 To: Vik Malhi vma...@ipexpert.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Voip peers Vic, Loopback solution is used as workaround to force CME to kick in the Xcoder.As we can see with G711 ,there is no need for loopback solution. But BACD work properly ,we need at least 144k BW on GK, please correct me if I'm wrong 128+16 for incoming calls from WAN On Thu, Jan 15, 2009 at 4:56 AM, Vik Malhi vma...@ipexpert.com wrote: Ok- Jose got it right and this is what he was about to say (right?) What is the difference between these two dial-peers on a gateway registered to GK? Dial-peer voice 1 voip destination-pattern 1... session target ras ... AND Dial-peer voice 2 voip destination-pattern 1... session target ipv4:Loopback0 IP Address ... When peer 1 is used an ARQ is sent to GK- it contains a bandwdidth Request and also a called # that needs resolving. When peer 2 is used an ARQ is also sent to the GK since this is also a VOIP call on a gateway to a gatkeeper- it contains just a bandwidth request though since we already did the resolution on the gateway (Lo0). With the B-ACD call we all know that we are sending the call to himself- the Loopback interface- as a workaround to invoke service aa on the inbound voip dial-peer. However the gateway/gk is master/slave and no VOIP call can take place without GK's authorization. The gateway doesn't cross- reference locally configured ip addresses for every session target. So even though the call is local GK still needs to accept the bandwidth request. And this is where bandwidth total ... 128 would hurt you since the B-ACD call will always use g711ulaw and the INTRA-bandwidth required is going to be 128kbps. Bandwidth total and B-ACD are dangerous for this reason. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video- On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Chris Parker cpar...@cparker.us Date: Wed,
[OSL | CCIE_Voice] All the files you need are on the router flash
I've seen this statment and it has been applied to both the Proctor Labs and the Cisco labs. I'm curious about something. 1. When installing things like CME, the TAR files often contain a read me file that has details on implementation. Is that read me file available? Is the TAR archive available where it can be extracted out? 2. Is there a TFTP server on the laptop provided in the actual exam? Does it have something like WinRAR that can rip open a TAR file? Not trying to get into NDA territoryjust trying to understand what some of the statements mean. I saw something today that indicates the command prompt isn't available on the laptop (bummerI use that a lot). Any other known surprises like this anyone would care to share? Cliff
Re: [OSL | CCIE_Voice] NTP CM
Hi, After editing the ntp.conf file please fo to CMD as C:\Prog file\cisco\xntp ntpdate -b IPADD of NTP pls try this and let me know On Thu, Jan 15, 2009 at 4:33 AM, Ryan Trauernicht ryanstudyvo...@gmail.comwrote: Any reason why CM will not keep its NTP clock. I have a local router with the following commands: ntp master 3 ntp source loopback0 (IP address is 192.168.187.1) I have edited the c:\WINNT\System32\Drivers\Etc\ntp.config My file looks like: server 192.168.187.1 # Set Local Clock to Authoritive Time Source fudge 192.168.187.1 stratum 5 # Resets Stratum from default 3 to 5 driftfile C:\WINNT\system32\drivers\etc\ntp.drift # path for drift file I stopped the Network Time Protocol service and I run ntpdate.exe 192.168.187.1 command from the cmd. That sets the clock to sync to NTP router just fine. After I reboot CM it goes back to GMT it looks like. I see it trying to sync but it never does. I have waited over 15mins and nothing. Thanks, Ryan Trauernicht
[OSL | CCIE_Voice] UNIVERCD
Does anyone know what is going on with the Cisco UNIVERCD? I understand that we will have access to this during the actual lab, but many of the major sections related to voice are broken links. If this happens in the lab exam, are you just screwed or what? Cliff
Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question!
Hi Vik, The 4082032220 is CTI route point in CCM, the CTI route point solution works only if I am disabling isdn outgoing ie redirectin-number under serial interface 0/2/0:23, If I am enbaling isdn outgoing ie redirect-number, then the CTI solution is not working. Could please let me know the above solution is right or I am missing some thing. --- On Wed, 14/1/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! To: mmailb...@yahoo.com, jeremy co jeremy.coo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Wednesday, 14 January, 2009, 12:34 PM In your lab what is 4082032220? It should be a RP with a VM Prof Mask = 3001 Call Fwd to VM. -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Balamurugan Singaram mmailb...@yahoo.com Reply-To: mmailb...@yahoo.com Date: Mon, 12 Jan 2009 21:33:19 -0800 (PST) To: jeremy co jeremy.coo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! Hi Vik, For SRST voice mail follow the CTI route point solution it, but till I am facing the redirect number problem. THE CTI debug is paste below, could you please let me know your suggestion please Best is solution to upgrade the IOS in home lab ? HQ# *Jan 11 04:14:13.417: ISDN Se0/3/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Facility i = 0x9F8B0100A10F020101020100800748512D32303031 Protocol Profile = Networking Extensions 0xA10F020101020100800748512D32303031 Component = Invoke component Invoke Id = 1 Operation = CallingName Name presentation allowed Name = HQ-2001 Progress Ind i = 0x8083 - Origination address is non-ISDN Display i = 'HQ-2001' Calling Party Number i = 0x0081, '2001' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '19723033001' Plan:Unknown, Type:Unknown *Jan 11 04:14:13.457: ISDN Se0/3/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x 800F Channel ID i = 0xA98383 Exclusive, Channel 3 *Jan 11 04:14:13.517: ISDN Se0/3/0:23 Q931: RX - ALERTING pd = 8 callref = 0x8 00F Progress Ind i = 0x8188 - In-band info or appropriate now available *Jan 11 04:14:18.549: ISDN Se0/3/0:23 Q931: RX - SETUP pd = 8 callref = 0x0181 Bearer Capability i = 0x9090A2 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Facility i = 0x9F8B0100A10F020101020100800748512D32303031 Protocol Profile = Networking Extensions 0xA10F020101020100800748512D32303031 Component = Invoke component Invoke Id = 1 Operation = CallingName Name presentation allowed Name = HQ-2001 Progress Ind i = 0x8083 - Origination address is non-ISDN Display i = 'HQ-2001' Calling Party Number i = 0x0081, '2001' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '4082032220' Plan:ISDN, Type:National Redirecting Number i = 0x7FE0FF, '3001' Plan:Reserved, Type:Reserved *Jan 11 04:14:18.577: ISDN Se0/3/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x --- On Tue, 13/1/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! To: jeremy co jeremy.coo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, 13 January, 2009, 12:30 AM I don’t get any RDNIS so you are doing much better than me. I think this the RDNIS with SRST has bugs that are fixed in 12.4(7). What IOS are you using? -- Vik Malhi – CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: jeremy co jeremy.coo...@gmail.com Date: Mon, 12 Jan 2009 21:51:57 +1100 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] unity and SRST wired
Re: [OSL | CCIE_Voice] NTP CM
That did not help Digging alittle bit further into this I see that my CM is actually pulling clock from my ESX box. Not sure why that is happening. Anyone else running ESX for their Call Manager having the same issue? On Wed, Jan 14, 2009 at 10:41 PM, karuna durai karu...@gmail.com wrote: Hi, After editing the ntp.conf file please fo to CMD as C:\Prog file\cisco\xntp ntpdate -b IPADD of NTP pls try this and let me know On Thu, Jan 15, 2009 at 4:33 AM, Ryan Trauernicht ryanstudyvo...@gmail.com wrote: Any reason why CM will not keep its NTP clock. I have a local router with the following commands: ntp master 3 ntp source loopback0 (IP address is 192.168.187.1) I have edited the c:\WINNT\System32\Drivers\Etc\ntp.config My file looks like: server 192.168.187.1 # Set Local Clock to Authoritive Time Source fudge 192.168.187.1 stratum 5 # Resets Stratum from default 3 to 5 driftfile C:\WINNT\system32\drivers\etc\ntp.drift # path for drift file I stopped the Network Time Protocol service and I run ntpdate.exe 192.168.187.1 command from the cmd. That sets the clock to sync to NTP router just fine. After I reboot CM it goes back to GMT it looks like. I see it trying to sync but it never does. I have waited over 15mins and nothing. Thanks, Ryan Trauernicht
Re: [OSL | CCIE_Voice] Layer 2 overhead
This is getting confusing. I read in IEexpert (or Internetwork Expert, I forgot) that for MLPoFR(g729), without compression: 27.6 with compression: 12.4 Let me see if I can still access that article... Message: 2 Date: Wed, 14 Jan 2009 15:09:01 -0600 From: Ryan Trauernicht ryanstudyvo...@gmail.com Subject: Re: [OSL | CCIE_Voice] Layer 2 overhead To: Shadab Abbasi (moabbasi) moabb...@cisco.com Cc: ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com, anil...@yahoo.com Message-ID: 8198026e0901141309v7fdc242fvaa686123dd7a4...@mail.gmail.com Content-Type: text/plain; charset=windows-1252 Vik any reason why the IPExperts lab do 13+4 for MLPoFR? On Wed, Jan 14, 2009 at 2:35 PM, Shadab Abbasi (moabbasi) moabb...@cisco.com wrote: And yes, its 77 w/o compression 39 with compression Regards, Shadab CCIE# 22893 (Voice) Technology Solutions Network ~Sent from my NOKIA E61i~ -Original Message- From: anil batra [mailto:anil...@yahoo.com anil...@yahoo.com] Sent: Thursday, January 15, 2009 04:26 AM China Standard Time To: Ryan Trauernicht; ccie_voice@onlinestudylist.com; Vik Malhi Subject:Re: [OSL | CCIE_Voice] Layer 2 overhead sorry need to add RTP/UDP/Header 40 too to these vlaues..so shall it be MLP with LFI = 20+40+4+13 =77 or 20+40-+13 =73
