[OSL | CCIE_Voice] wierd h323 gateway problem lab7
Wierd H323 gateway problem PSTN inbound call to BR2 phones does not ring phones.. calls come in and i verify using debug isdn q931 but phone does not ring, if i use csim start from gateway to call phone, it rings, so that eliminates partition and inbound CSS issue. phone can successfully make outbound calls. BR1#sh run | s dial-peer dial-peer voice 1000 voip destination-pattern 1...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad dial-peer voice 1005 voip preference 1 destination-pattern 1...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad dial-peer voice 10 pots destination-pattern 9%911 incoming called-number . no digit-strip direct-inward-dial port 0/1/1:23 dial-peer voice 15 pots translation-profile outgoing 7digitani destination-pattern 9[2-9].. port 0/1/1:23 forward-digits 7 dial-peer voice 20 pots translation-profile outgoing TEHO-BR2 destination-pattern 0114420T port 0/1/1:23 forward-digits all dial-peer voice 25 pots incoming called-number . direct-inward-dial BR1# BR1# BR1# BR1#csim start 1001 csim: called number = 1001, loop count = 1 ping count = 0 Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=1001, Peer Info Type=DIALPEER_INFO_SPEECH Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1001 Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST Feb 13 17:44:24.956: //-1//DPM/dpMatchPeers: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=1000 2: Dial-peer Tag=1005 csim err csimDisconnected recvd DISC cid(97) csim: loop = 1, failed = 1 csim: call attempted = 1, setup failed = 1, tone failed = 0 BR1# BR1# BR1# BR1# BR1# BR1#csim start 1002 csim: called number = 1002, loop count = 1 ping count = 0 Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST Feb 13 17:44:43.460: //-1//DPM/dpMatchPeers: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=1000 2: Dial-peer Tag=1005 csim err csimDisconnected recvd DISC cid(98) csim: loop = 1, failed = 1 csim: call attempted = 1, setup failed = 1, tone failed = 0 BR1# BR1# BR1# BR1# BR1# BR1#ping 10.10.210.11 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.210.11, timeout is 2 seconds: ! Success rate is 100 percent (5/5), round-trip min/avg/max = 56/56/60 ms BR1#ping 10.10.210.10 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.210.10, timeout is 2 seconds: ! Success rate is 100 percent (5/5), round-trip min/avg/max = 56/57/60 ms BR1# any thots? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] I am kiked out of my rack session and now I can not access proctorlabs.com (website timed out)
Is anyone having this problem? I can access all other website and even can ping all servers and routers of my rack. Thanks___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Q5.1
Hi, I'd say yes if the negotiated codec between the gw and ucm is g.729r8 (default codec for the dial peer), so make sure the gw is using that codec when talking to ucm or at least it's in the list for the voice class codec used by the dial peer, Thanks, On Wed, Feb 10, 2010 at 2:07 PM, vccie2010 vccie2...@gmail.com wrote: Per the PG solutions the dspfarm profile 1 transcode does not show codec g729r8 it only shows g279ar8 and g729abr8 don't we need codec G729r8 statement here since the traffic coming from UCM across GK will be G729r8 ??? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] wierd h323 gateway problem lab7
Hi Sean, Your incoming pots dial peer and voip outgoing dial peers (and perhaps your voice port) don't have any called number translation, so from your h323 gw config in ucm set significant digits to 4, also make sure the h.323 interface in your br2 gw has the h323-gateway voip interface and h323-gateway voip bind scraddr commands configured, Thanks, -. On Sat, Feb 13, 2010 at 12:35 PM, sean hurricane shurric...@gmail.comwrote: Wierd H323 gateway problem PSTN inbound call to BR2 phones does not ring phones.. calls come in and i verify using debug isdn q931 but phone does not ring, if i use csim start from gateway to call phone, it rings, so that eliminates partition and inbound CSS issue. phone can successfully make outbound calls. BR1#sh run | s dial-peer dial-peer voice 1000 voip destination-pattern 1...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.11 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad dial-peer voice 1005 voip preference 1 destination-pattern 1...$ voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.10 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad dial-peer voice 10 pots destination-pattern 9%911 incoming called-number . no digit-strip direct-inward-dial port 0/1/1:23 dial-peer voice 15 pots translation-profile outgoing 7digitani destination-pattern 9[2-9].. port 0/1/1:23 forward-digits 7 dial-peer voice 20 pots translation-profile outgoing TEHO-BR2 destination-pattern 0114420T port 0/1/1:23 forward-digits all dial-peer voice 25 pots incoming called-number . direct-inward-dial BR1# BR1# BR1# BR1#csim start 1001 csim: called number = 1001, loop count = 1 ping count = 0 Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=1001, Peer Info Type=DIALPEER_INFO_SPEECH Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1001 Feb 13 17:44:24.956: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST Feb 13 17:44:24.956: //-1//DPM/dpMatchPeers: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=1000 2: Dial-peer Tag=1005 csim err csimDisconnected recvd DISC cid(97) csim: loop = 1, failed = 1 csim: call attempted = 1, setup failed = 1, tone failed = 0 BR1# BR1# BR1# BR1# BR1# BR1#csim start 1002 csim: called number = 1002, loop count = 1 ping count = 0 Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: Calling Number=, Called Number=1002, Peer Info Type=DIALPEER_INFO_SPEECH Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1002 Feb 13 17:44:43.460: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST Feb 13 17:44:43.460: //-1//DPM/dpMatchPeers: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=1000 2: Dial-peer Tag=1005 csim err csimDisconnected recvd DISC cid(98) csim: loop = 1, failed = 1 csim: call attempted = 1, setup failed = 1, tone failed = 0 BR1# BR1# BR1# BR1# BR1# BR1#ping 10.10.210.11 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.210.11, timeout is 2 seconds: ! Success rate is 100 percent (5/5), round-trip min/avg/max = 56/56/60 ms BR1#ping 10.10.210.10 Type escape sequence to abort. Sending 5, 100-byte ICMP Echos to 10.10.210.10, timeout is 2 seconds: ! Success rate is 100 percent (5/5), round-trip min/avg/max = 56/57/60 ms BR1# any thots? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] IP Blue Phone- how to get license (S/N) back
I had my license, and by mistake I unregister the license, so VTGO phone works in demo mode now. I couldn't find the license file, is there a way to contact VTGO or by any other means to find that S/N so I can register phones again. Thanks in advance,___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] HQBR2CUE getting fast busy
HQBR2CUE getting fast busy CUE is inte to CCM, it works when the region between HQ and BR2 is setup as G711. I do have transcodes reg and in MRGL of HQ and BR2. Hq mrgl has tran config on hq router and br2 mrgl has tran config on br2 router. CUE CTI RP and Ports are in BR2 DP. Any clue pls? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Can't call out from BR1 phone in SIP SRST
Can't call out from BR1 phone in SIP SRST but I can call into it. When I dial from SIP phone in SRST the moment I press dial softkey I hangs up and if I press speaker phone and dial the first digit it hangs up. Here si the config for SIP SRST pls... voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco sip bind control source-interface FastEthernet0/0.21 bind media source-interface FastEthernet0/0.21 registrar server voice register global max-dn 1 max-pool 1 ! voice register pool 1 id network 177.2.11.0 mask 255.255.255.0 application sip.app dtmf-relay rtp-nte sip-notify call-forward b2bua busy 2220 call-forward b2bua noan 0 timeout 6 codec g711ulaw ! am i missing somehting here pls ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
