[OSL | CCIE_Voice] Slow busy IPCCX
Anybody has any idea about why IPCCX would give slow busy signal. I can not see anything wrong. Any guidance will be appreciated.
Re: [OSL | CCIE_Voice] 0 Conf max sessions
: CISCO2811 , VID: V03 , SN: FHK103671CU NAME: VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0, DESCR: VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 PID: VWIC2-2MFT-T1/E1 , VID: V01 , SN: FOC102540TY NAME: PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 4, DESCR: PVDMII DSP SIMM with one DSP PID: PVDM2-16 , VID: V01 , SN: FOC1032054R NAME: NM-SE on Slot 1, DESCR: NM-SE PID: NM-CUE, VID: V03, SN: FOC10120B2E NAME: 40GB IDE Disc Daughter Card on Slot 1 SubSlot 0, DESCR: 40GB IDE Disc Daughter Card PID: , VID: 1.0, SN: FOC10170V2B On Mon, Jul 7, 2008 at 9:14 AM, Onur Tufekci [EMAIL PROTECTED] wrote: I actually configured all that right after I send this message but no luck. On Sun, Jul 6, 2008 at 11:23 PM, Vik Malhi [EMAIL PROTECTED] wrote: try configuring sccp ccm and sccp ccm group before you set the max sessions. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mehmet Tufekci Sent: Sunday, July 06, 2008 9:17 AM To: ccievoice1 Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] 0 Conf max sessions I have 1-3 configured under controller. I am able to see 1-8 available resources under my transcoder profile but no luck with conference. I know conference will not show up if you enable transcoder profile first. No matter what I tried I can not get it to run. On Jul 6, 2008, at 11:44 AM, ccievoice1 wrote: have you utilized all the dsp resources for your pri-group? On Sun, Jul 6, 2008 at 11:31 PM, Mehmet Tufekci [EMAIL PROTECTED] wrote: Hi All, I can not figure out why maximum session 0-0 is showing under conference profile. I did not enable transcoding profile yet. voice-card 0 dspfarm dsp services dspfarm ! ! ! interface Loopback0 ip address 172.3.102.1 255.255.255.255 ip ospf network point-to-point ! ! sccp local Loopback0 sccp ! dspfarm profile 1 transcode codec g711ulaw codec g729r8 shutdown ! dspfarm profile 2 conference codec g711ulaw codec g729r8 shutdown !
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 29, Issue 9
-- Message: 5 Date: Sun, 6 Jul 2008 21:54:32 -0500 From: Nguyen Le [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPPA Service Login To: Jonathan Charles [EMAIL PROTECTED] Cc: CCIE Voice ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 communication to the ipcc server is there. If i stop the IPPA service, I'll get a host not found error on the phones instead. going to this site connects, just gets the error. http://192.168.1.1:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp Nguyen On Sun, Jul 6, 2008 at 9:49 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Check that you have communication to the IPCC server (aka the Pub...) Jonathan On Sun, Jul 6, 2008 at 9:20 PM, Nguyen Le [EMAIL PROTECTED] wrote: On the IPPA Service. What would cause it to come up with this error. Unable to connecto to the IPPA service this is when you are trying to login via your phone. Thanks Nguyen -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080706/20d 21a5b/attachment-0001.html -- Message: 6 Date: Sun, 6 Jul 2008 21:56:51 -0500 From: Jonathan Charles [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPPA Service Login To: Nguyen Le [EMAIL PROTECTED] Cc: CCIE Voice ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_cente r/cad_enterprise/cadenterprise7_1/user/guide/cip71ug.pdf Page 20 Looks like your CTI mgr might be down... Restart it on CCM and check the IPCC services... Jonathan On Sun, Jul 6, 2008 at 9:54 PM, Nguyen Le [EMAIL PROTECTED] wrote: communication to the ipcc server is there. If i stop the IPPA service, I'll get a host not found error on the phones instead. going to this site connects, just gets the error. http://192.168.1.1:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp Nguyen On Sun, Jul 6, 2008 at 9:49 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Check that you have communication to the IPCC server (aka the Pub...) Jonathan On Sun, Jul 6, 2008 at 9:20 PM, Nguyen Le [EMAIL PROTECTED] wrote: On the IPPA Service. What would cause it to come up with this error. Unable to connecto to the IPPA service this is when you are trying to login via your phone. Thanks Nguyen -- Message: 7 Date: Sun, 6 Jul 2008 20:23:59 -0700 From: Vik Malhi [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] 0 Conf max sessions To: 'Mehmet Tufekci' [EMAIL PROTECTED],'ccievoice1' [EMAIL PROTECTED] Cc: 'OSL CCIE Voice Lab Exam' ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii try configuring sccp ccm and sccp ccm group before you set the max sessions. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mehmet Tufekci Sent: Sunday, July 06, 2008 9:17 AM To: ccievoice1 Cc: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] 0 Conf max sessions I have 1-3 configured under controller. I am able to see 1-8 available resources under my transcoder profile but no luck with conference. I know conference will not show up if you enable transcoder profile first. No matter what I tried I can not get it to run. On Jul 6, 2008, at 11:44 AM, ccievoice1 wrote: have you utilized all the dsp resources for your pri-group? On Sun, Jul 6, 2008 at 11:31 PM, Mehmet Tufekci [EMAIL PROTECTED] wrote: Hi All, I can not figure out why maximum session 0-0 is showing under conference profile. I did not enable transcoding profile yet. voice-card 0 dspfarm dsp services dspfarm ! ! ! interface Loopback0 ip address 172.3.102.1 255.255.255.255 ip ospf network point-to-point ! ! sccp local Loopback0 sccp ! dspfarm profile 1 transcode codec g711ulaw codec g729r8 shutdown ! dspfarm profile 2 conference codec g711ulaw codec g729r8 shutdown ! -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080706/051 f6946/attachment-0001.html -- Message: 8 Date: Sun, 6 Jul 2008 21:25:22 -0600 From: Derrick Shumake [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPPA Service Login Cc: CCIE
Re: [OSL | CCIE_Voice] GDM Log-in
Hi Frog, I understand what is going on and I am sure what is provided is a valid solution. There are two different scenarios going on and that is making us run around circles. One is to create an ephone-dn with GDM number and assign it to the Phone so MWI light will come on this will not allow you to assign GDM's number as E164 and if you try to go to voicemail thru this ephone-dn then it will ask you for ID. Another one (described below) is to create an ephone-dn and access to GDM thru that extension by using E164 but this will not help with the MWI. Does this make sense now? Cheers, Onur. On Jul 7, 2008, at 7:35 PM, FrogOnDSCP46EF wrote: Hi Onue, It sounds like your situation is the same as Jack Skellington in The nightmare before Christmas ( I was shocked to see that 3D version is available now) I think your quesiton is already answered and is a valid one :) -frog - strip- Onur. Yes, there is no need to create separate voice mail box for 2nd line, just use the same mailbox to access the GDM, the phone belongs to one person only. There is no way that you could access GDM directly by pressing 9 without logging in to the mailbox first. What u need to do is put 2nd lines DN as E.164 under first lines DN settings in CUE. After this when u take line 2 and press message key then it will ask for password only and then once u logged in you can press 9 to access GDM. ---strip -- Smile, you'll save someone else's day! Frog Message: 2 Date: Mon, 7 Jul 2008 12:22:38 -0400 From: Onur Tufekci [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] GDM Log-in To: Juan [EMAIL PROTECTED] Cc: o Ninja [EMAIL PROTECTED], ccie_voice@onlinestudylist.com, [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8 Hi Vik, Mark, This subject is getting frustrated (I sure am) since no one is able to figure out what exactly is going on. Is it possible for you guys to shed some light on this? Best Regards, Onur. - Onur. Yes, there is no need to create separate voice mail box for 2nd line, just use the same mailbox to access the GDM, the phone belongs to one person only. There is no way that you could access GDM directly by pressing 9 without logging in to the mailbox first. What u need to do is put 2nd lines DN as E.164 under first lines DN settings in CUE. After this when u take line 2 and press message key then it will ask for password only and then once u logged in you can press 9 to access GDM.
Re: [OSL | CCIE_Voice] GDM Log-in
I just learned this you can assign GDM number as you E164 number to your main line. On Jul 6, 2008, at 10:16 AM, o Ninja wrote: I know, but the case is that I dont want to spare a button on my phones just to know that GDM has messages, I wanted to receive these messages in lines I have configured previously. CC: ccie_voice@onlinestudylist.com From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] GDM Log-in Date: Sun, 6 Jul 2008 10:13:41 -0400 Users can see the voicemail light (if you use ephone-dn for the GDM) but from the discussion with others and reading little bit it does not seem possible to reach the VMs in GDM directly. No matter what you do you need to go thru your own mailbox and dial 9 to listen messages. On Jul 6, 2008, at 8:02 AM, o Ninja wrote: Hi Mehmet, I am trying to do that also, I want to leave a message to the GDM mailbox and then the members of this GDM receive to receive the messages. This is a simple solution but looks like CUE is not able to do it. Conheça já o Windows Live Spaces, o site de relacionamentos do Messenger! Crie já o seu! Conheça já o Windows Live Spaces, o site de relacionamentos do Messenger! Crie já o seu!
Re: [OSL | CCIE_Voice] 0 Conf max sessions
I have 1-3 configured under controller. I am able to see 1-8 available resources under my transcoder profile but no luck with conference. I know conference will not show up if you enable transcoder profile first. No matter what I tried I can not get it to run. On Jul 6, 2008, at 11:44 AM, ccievoice1 wrote: have you utilized all the dsp resources for your pri-group? On Sun, Jul 6, 2008 at 11:31 PM, Mehmet Tufekci [EMAIL PROTECTED] wrote: Hi All, I can not figure out why maximum session 0-0 is showing under conference profile. I did not enable transcoding profile yet. voice-card 0 dspfarm dsp services dspfarm ! ! ! interface Loopback0 ip address 172.3.102.1 255.255.255.255 ip ospf network point-to-point ! ! sccp local Loopback0 sccp ! dspfarm profile 1 transcode codec g711ulaw codec g729r8 shutdown ! dspfarm profile 2 conference codec g711ulaw codec g729r8 shutdown !
[OSL | CCIE_Voice] GDM Log-in
Hi All, Am I wasting my time trying to figure out a way to get GDM log-in to work directly, without pressing 9? I am trying to use the shared line and go thru there but no luck. Also I tried couple of other ways like forwarding VMs to distribution list so users in the list will receive the VMs in their mailboxes. Regards, Onur.
Re: [OSL | CCIE_Voice] VMware Fusion IP Communicator
Thank you, I got it to work. I was using the VPN client on the MAC and nothing on the VMs. Now I run the VPN client on the VMs and it works. Cheers, Onur. On Jun 28, 2008, at 10:33 AM, Matthew Bynum wrote: Try configuring the NIC to be Bridged rather than NAT. -- Matthew Bynum On Sat, Jun 28, 2008 at 9:02 AM, Mehmet Tufekci [EMAIL PROTECTED] wrote: Can we use IP communicator with Fusion? It starts up normally but never registers with callmanager.
Re: [OSL | CCIE_Voice] Annunciator to external callers
You can change the wav file in the TFTP folder in CCM. You need to record a new one. On Jun 28, 2008, at 2:42 PM, Ricardo Arevalo wrote: Hi, I need help on this: How can i do to get external callers to get an error mesagge (annunciator) when calling to internal unassigned number instead of just a fast busy? thnks //r.a.
Re: [OSL | CCIE_Voice] MWI ON-OFF
I got it all configured already but no luck. P8-BR2-RTR#show run Building configuration... Current configuration : 6232 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname P8-BR2-RTR ! boot-start-marker boot-end-marker ! logging buffered 4096 debugging ! no aaa new-model ! resource policy ! memory-size iomem 30 network-clock-participate wic 0 no network-clock-participate wic 1 network-clock-select 1 E1 0/0/0 ip subnet-zero ! ! ip cef no ip dhcp use vrf connected ip dhcp excluded-address 10.8.202.1 10.8.202.39 ip dhcp excluded-address 10.8.202.51 10.8.202.254 ! ip dhcp pool voice network 10.8.202.0 255.255.255.0 option 150 ip 10.8.202.1 default-router 10.8.202.1 ! ! no ip domain lookup ! isdn switch-type primary-ni ! voice-card 0 dspfarm dsp services dspfarm ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 sip ! ! ! ! ! ! ! ! ! ! voice translation-rule 1 rule 1 /^331328\(3...$\)/ /\1/ ! voice translation-rule 2 rule 1 /^3...$/ /331328\0/ ! ! voice translation-profile pstn-in translate called 1 ! voice translation-profile pstn-out translate calling 2 ! ! ! ! ! ! controller E1 0/0/0 pri-group timeslots 1-3,16 ! controller E1 0/0/1 ! controller T1 0/1/0 framing esf linecode b8zs channel-group 0 timeslots 1-24 ! ! ! ! ! interface Loopback0 ip address 172.8.102.1 255.255.255.255 ip ospf network point-to-point ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.280 encapsulation dot1Q 280 ip address 10.8.202.1 255.255.255.0 no snmp trap link-status h323-gateway voip interface h323-gateway voip id UCME ipaddr 172.8.100.1 1719 h323-gateway voip h323-id UCME h323-gateway voip tech-prefix 2# h323-gateway voip bind srcaddr 10.8.202.1 ! interface Service-Engine0/0 ip unnumbered FastEthernet0/0.280 service-module ip address 10.8.202.2 255.255.255.0 service-module ip default-gateway 10.8.202.1 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0:15 no ip address isdn switch-type primary-ni isdn incoming-voice voice isdn outgoing display-ie no cdp enable ! interface Serial0/1/0:0 no ip address encapsulation frame-relay IETF frame-relay lmi-type ansi ! interface Serial0/1/0:0.1 point-to-point ip address 162.8.102.2 255.255.255.0 frame-relay interface-dlci 102 ! router ospf 1 log-adjacency-changes network 10.8.102.0 0.0.0.255 area 0 network 10.8.202.0 0.0.0.255 area 0 network 162.8.102.0 0.0.0.255 area 0 network 172.8.102.0 0.0.0.255 area 0 network 192.8.102.0 0.0.0.255 area 0 ! ip classless ip route 10.8.202.2 255.255.255.255 Service-Engine0/0 ! ! ip http server ip http authentication local no ip http secure-server ip http path flash: ! ! ! ! tftp-server flash:P00307010100.bin tftp-server flash:P00305000400.bin tftp-server flash:P00303020214.bin tftp-server flash:P00403020214.bin tftp-server flash:P00305000600.sbn tftp-server flash:P00307020200.bin tftp-server flash:P00307020200.loads tftp-server flash:P00307020200.sb2 tftp-server flash:P00307020200.sbn tftp-server flash:P00307020400.bin tftp-server flash:P00307020400.sbn tftp-server flash:P00307020400.sb2 tftp-server flash:P00307020400.loads ! control-plane ! ! ! voice-port 0/0/0:15 translation-profile incoming pstn-in translation-profile outgoing pstn-out ! ! ! sccp local FastEthernet0/0.280 sccp ccm 10.8.202.1 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register XCODER ! dspfarm profile 2 transcode codec g711ulaw codec g729r8 maximum sessions 2 associate application SCCP ! dspfarm profile 1 conference codec g711ulaw codec g729r8 associate application SCCP shutdown ! ! dial-peer voice 1 pots direct-inward-dial port 0/0/0:15 ! dial-peer voice 2 voip incoming called-number 2#T dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad ! dial-peer voice 3600 voip destination-pattern 3600 session protocol sipv2 session target ipv4:10.8.202.2 incoming called-number 399[89] dtmf-relay sip-notify codec g711ulaw ip qos dscp cs3 signaling no vad ! dial-peer voice 1000 voip destination-pattern [12]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad ! num-exp 2#3... 3... gateway timer receive-rtp 1200 ! sip-ua ! ! ! ! gatekeeper shutdown ! ! telephony-service load 7910 P00403020214 load 7960-7940 P00307020400 max-ephones 3 max-dn 10 ip source-address 10.8.202.1 port 2000 auto assign 1 to 3 sdspfarm units 4 sdspfarm transcode sessions 2 sdspfarm tag 1 CONF sdspfarm tag 2 XCODER create cnf-files version-stamp Jan 01 2002 00:00:00 voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T moh music-on-hold.au web admin system name cisco password cisco dn-webedit time-webedit transfer-system
Re: [OSL | CCIE_Voice] SRST voicemail
Do you have direct-inward-dial in your dial-peer? On Jun 25, 2008, at 6:32 PM, Gregory Jost (grjost) wrote: I’m getting secondary dial-tone on forwards to voicemail in SRST mode from PSTN. I’m able to call internally and it routes to the appropriate mailbox in SRST mode, but calls from PSTN are routing back to the SRST router and giving secondary dial-tone. Any ideas? I’m sure it’s something with my PSTN-WAN router. The weird thing is that that direct calls work, but if there’s an RDNIS sent it loops back to the originating gateway. call-manager-fallback transfer-pattern .T voicemail 912122211600 call-forward pattern .T call-forward busy 91212221101. call-forward noan 91212221101. timeout 3 Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922
Re: [OSL | CCIE_Voice] BACD Problem
If you are using g729 from the gateway then the answer might be transcoder can not be invoked for BACD. On Jun 24, 2008, at 9:38 PM, Nguyen Le wrote: Try unchecking wait for h.245 terminal capabilities in your GK trunk configuration On Tue, Jun 24, 2008 at 6:41 PM, Jose Linero Welcker [EMAIL PROTECTED] wrote: Hi: I am testing the B-ACD TCL in BR2, the connection between BR2 (CME) and the CCM is trough a gatekeeper and when I called to the pilot number of the script is not working. The local calls from the IP Phones registered to the CME are working and the script is ok, the calls coming from the PSTN to the BACD are working too. The configuration I have specifically to BACD is: application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param queue-manager-debugs 1 param aa-hunt2 4210 param number-of-hunt-grps 1 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 1 param handoff-string aa param dial-by-extension-option 1 paramspace english language en param max-time-vm-retry 2 param aa-pilot 4500 paramspace english location flash: param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param voice-mail 4600 param max-time-call-retry 700 param service-name queue dial-peer voice 10 voip destination-pattern 4500 session target ipv4:172.1.102.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 11 voip service aa incoming called-number 4500 dtmf-relay h245-alphanumeric codec g711ulaw The dial peer to receive the calls from CCM is: dial-peer voice 5 voip translation-profile incoming DNIS destination-pattern [23]... session target ras incoming called-number . tech-prefix 1# dtmf-relay h245-alphanumeric I have configured the transcodec: BR2-RTR-2821#sh sccp SCCP Admin State: UP Gateway IP Address: 142.101.66.1, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 142.101.66.1, Port Number: 2000 Priority: N/A, Version: 3.1, Identifier: 1 Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 142.101.66.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 and the BR2 router is also been configured as IPIPGW: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! Doing a debug voice application I have this error: BR2-RTR-2821# Jun 24 23:32:09.308: //-1//AFW_:/C_ServiceSession_Event_Handler: Jun 24 23:32:09.308: //-1//AFW_:/AFW_Session_New: Jun 24 23:32:09.308: //146//AFW_:/C_PackageSession_NewCall: Session module listened by TclModule_45F87050_0_99922156 Jun 24 23:32:09.308: //146//AFW_:/Open_SetupIndication: Calling #(3001), Called #(852#4500), peer_tag(5) Jun 24 23:32:09.308: //-1//AFW_:/C_PackageSession_GetSigPeer: Jun 24 23:32:19.396: //146//AFW_:/AnyState_Disconnected: Jun 24 23:32:19.396: //146//AFW_:/Session_Close: lastFailureCause 47 Jun 24 23:32:19.396: //146//AFW_:/AFW_M_Session_Terminate: Jun 24 23:32:19.396: //146//AFW_:/AFW_M_Session_Terminate: lastFailureCause 47 Looking for the meaning of this error is: Last Disconnect Cause is 2F , Last Disconnect Text is no resource (47), I am stuck with this problem, any idea of what is the cause? Regards, Jose Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it!
