Re: [OSL | CCIE_Voice] 7961 unprovisioned ucme
Hi: Yes, the sip bind command is very important, if you don't configure it the system takes one by default, (the loo0 if i remember well), but this can be dangerous, one time integrating CUE with CME (using SIP), the mailboxes where working well, but the mwi were not working, the problem was that the SIP source interface was lo0, and the CUE default gw was fas0/0.1 so CME didn't recieved the MWI call, (this can be very difficult to trobleshoot becouse no dial-peer is hitted by MWI so you can't use deb voip dialpeer inout). But in this case it wasn't the source of the problem, becouse I added the sip source interface previously and the gw didn't show any sip message nor any deb voice register message, the problem is that if you don't specify the number 1 dn x the phone reads the config ... Mar 23 13:34:30.188: TFTP: Looking for SEP0017E066C72F.cnf.xml Mar 23 13:34:30.188: TFTP: Opened system:/cme/sipphone/SEP0017E066C72F.cnf.xml, fd 7, size 3277 for process 155 Mar 23 13:34:30.204: TFTP: Finished and becouse it doesn't find the dn in the first number it start to download the config again (and it remains in a loop and in unprovisoned state) so it never sends a sip message to register so deb ccsip message and deb voice register would show anything This is what the phone needs to read in the cnf.xml file in this case to register sipLines line button=1 featureID9/featureID featureLabel4001/featureLabel proxyUSECALLMANAGER/proxy port5060/port name4001/name /line /sipLines In my case I saw this: name/name, correcting this (I do this manually copying the file to my laptop, copying back to the router and reseting the phone) solves the issue, then I replicated the problem again and I fix it changing the config (number 1 dn 1) and then upgrade profile and reset the phone, this way the phone starts exchanging sip messages with the registrar server and finally it registers. Then I checked the cnf.xml file and name4001/name was there. The good thing of this issue is that can be easly replicated to test it HTH Regards Date: Wed, 24 Mar 2010 21:13:44 -0500 Subject: Re: [OSL | CCIE_Voice] 7961 unprovisioned ucme From: ill2...@gmail.com To: gorr...@hotmail.com CC: dberlin...@gmail.com; ccie_voice@onlinestudylist.com Angel, Thanks for the update. That was a tricky one a it required good observation, where we started to think about all the other common causes instead. One thing I want to bring out, is that the sip source command is very important, and whenever I miss it, I have similar symptoms where the debug ccsip messages does not show any output. I would assume that it was two things that was needed: the sip bind, and the correction on the dn. I assume this, because the bad dn should at least show something on the debug ccsip messages. My two cents. On Wed, Mar 24, 2010 at 10:30 AM, Angel Perez gorr...@hotmail.com wrote: Thanks, the problem was that the second phone had number 2 dn 2 and it should be number 1 dn 2, I've notice that if you don't configure a number or the first dn isn't at number 1 the phone will stay at unprovisioned. After changing that and then create profile and reset at voice ragister global solves this issue Regards Date: Thu, 25 Mar 2010 03:26:14 +1300 Subject: Re: [OSL | CCIE_Voice] 7961 unprovisioned ucme From: dberlin...@gmail.com To: gorr...@hotmail.com Hi Angel Have you tried to configure the username of the problem SIP phone as the first line you have configured on the phone, i.e. username 4001 password cisco. The bind control and media you have there is for the same interface as the one you are sourcing your SIP packets - 146.102.66.254 lastly ensure your ntp time source is synced and do another create profile before retrying regards Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 6A - AAR task
Hello: It looks like if the pstn is expecting the number in other format, first check how the pstn needs the called number to be formated, then play with EPNM and AAR prefix / mask to send the number as is expected hth Date: Thu, 25 Mar 2010 00:03:26 -0400 From: kparam2...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 6A - AAR task Hi guys, I was working on lab 6A to set up AAR between the HQ and BR1. I was successful in setting the rsvp, reducing the bandwidth and forcing the call to use AAR and go over the PSTN. I was able to complete the call when i call from BR1 (1002) to HQ (5002). But when I dial from HQ(5002) to BR1(1002), I get a busy tone and the dreaded Unallocated/Unassigned isdn message on BR1 I have css-br-ld as the incoming CSS on the BR1 RTR and that css has pt-internal. 1002 is assigned pt-internal. I am not sure what I am missing here. Here are the debug outputs from HQ and BR1 RTR. on HQ Router: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=11 Mar 25 00:58:30.581: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 Mar 25 00:58:30.581: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Calling num 2123945002 Mar 25 00:58:30.585: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Called num 16178631002 Mar 25 00:58:30.585: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0097 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'HQ Phone 2' Calling Party Number i = 0x2181, '2123945002' Plan:ISDN, Type:National Called Party Number i = 0x80, '16178631002' Plan:Unknown, Type:Unknown Redirecting Number i = 0x81, '1002' Plan:Unknown, Type:Unknown HQ-RTR# Mar 25 00:58:30.621: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8097 Channel ID i = 0xA98383 Exclusive, Channel 3 Mar 25 00:58:30.657: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8 callref = 0x8097 Cause i = 0x8281 - Unallocated/unassigned number Mar 25 00:58:30.665: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8 callref = 0x0097 Mar 25 00:58:30.673: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x8097 on BR1 RTR: BR1-RTR# Mar 25 00:58:30.656: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0098 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Display i = 'HQ Phone 2' Calling Party Number i = 0x2181, '2123945002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '6178631002' Plan:ISDN, Type:National Redirecting Number i = 0x81, '1002' Plan:Unknown, Type:Unknown Mar 25 00:58:30.668: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x8098 Cause i = 0x8081 - Unallocated/unassigned number Any input will be greatly appreciated. Thanks Kalyan _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] 7961 unprovisioned ucme
Hi again: Just a side note: line button=1 name4001/name means that number 1 is associated with dn 1 (4001) = number 1 dn 1, don't mistake with label, or name commands, you can find this commands at the cnf.xml file as following: featureLabelbr2 phone 1/featureLabel displayNamebr2 phone 1 /displayName hth From: gorr...@hotmail.com To: ill2...@gmail.com Date: Thu, 25 Mar 2010 08:52:34 + CC: ccie_voice@onlinestudylist.com; dberlin...@gmail.com Subject: Re: [OSL | CCIE_Voice] 7961 unprovisioned ucme Hi: Yes, the sip bind command is very important, if you don't configure it the system takes one by default, (the loo0 if i remember well), but this can be dangerous, one time integrating CUE with CME (using SIP), the mailboxes where working well, but the mwi were not working, the problem was that the SIP source interface was lo0, and the CUE default gw was fas0/0.1 so CME didn't recieved the MWI call, (this can be very difficult to trobleshoot becouse no dial-peer is hitted by MWI so you can't use deb voip dialpeer inout). But in this case it wasn't the source of the problem, becouse I added the sip source interface previously and the gw didn't show any sip message nor any deb voice register message, the problem is that if you don't specify the number 1 dn x the phone reads the config ... Mar 23 13:34:30.188: TFTP: Looking for SEP0017E066C72F.cnf.xml Mar 23 13:34:30.188: TFTP: Opened system:/cme/sipphone/SEP0017E066C72F.cnf.xml, fd 7, size 3277 for process 155 Mar 23 13:34:30.204: TFTP: Finished and becouse it doesn't find the dn in the first number it start to download the config again (and it remains in a loop and in unprovisoned state) so it never sends a sip message to register so deb ccsip message and deb voice register would show anything This is what the phone needs to read in the cnf.xml file in this case to register sipLines line button=1 featureID9/featureID featureLabel4001/featureLabel proxyUSECALLMANAGER/proxy port5060/port name4001/name /line /sipLines In my case I saw this: name/name, correcting this (I do this manually copying the file to my laptop, copying back to the router and reseting the phone) solves the issue, then I replicated the problem again and I fix it changing the config (number 1 dn 1) and then upgrade profile and reset the phone, this way the phone starts exchanging sip messages with the registrar server and finally it registers. Then I checked the cnf.xml file and name4001/name was there. The good thing of this issue is that can be easly replicated to test it HTH Regards Date: Wed, 24 Mar 2010 21:13:44 -0500 Subject: Re: [OSL | CCIE_Voice] 7961 unprovisioned ucme From: ill2...@gmail.com To: gorr...@hotmail.com CC: dberlin...@gmail.com; ccie_voice@onlinestudylist.com Angel, Thanks for the update. That was a tricky one a it required good observation, where we started to think about all the other common causes instead. One thing I want to bring out, is that the sip source command is very important, and whenever I miss it, I have similar symptoms where the debug ccsip messages does not show any output. I would assume that it was two things that was needed: the sip bind, and the correction on the dn. I assume this, because the bad dn should at least show something on the debug ccsip messages. My two cents. On Wed, Mar 24, 2010 at 10:30 AM, Angel Perez gorr...@hotmail.com wrote: Thanks, the problem was that the second phone had number 2 dn 2 and it should be number 1 dn 2, I've notice that if you don't configure a number or the first dn isn't at number 1 the phone will stay at unprovisioned. After changing that and then create profile and reset at voice ragister global solves this issue Regards Date: Thu, 25 Mar 2010 03:26:14 +1300 Subject: Re: [OSL | CCIE_Voice] 7961 unprovisioned ucme From: dberlin...@gmail.com To: gorr...@hotmail.com Hi Angel Have you tried to configure the username of the problem SIP phone as the first line you have configured on the phone, i.e. username 4001 password cisco. The bind control and media you have there is for the same interface as the one you are sourcing your SIP packets - 146.102.66.254 lastly ensure your ntp time source is synced and do another create profile before retrying regards Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information
Re: [OSL | CCIE_Voice] SIP Integration of UCM and UC
yes i did On Thu, Mar 25, 2010 at 4:22 AM, Otto Sanchez o...@ipexpert.com wrote: Hello, Did you also checked that: 1.- Sip trunk security profile has Accept Unsolicited Notification checked 2.- Some ports in UC are enabled to Send MWI Requests Thanks, On Mon, Mar 22, 2010 at 11:40 PM, Omotayo adefilabi...@gmail.com wrote: Hello Otto, I checked the Redirecting Diversion Header Delivery - Inbound and Redirecting Diversion Header Delivery - outbound Voicemail works now but MWI is not working what do i need to do to fix it thanks On Mon, Mar 22, 2010 at 10:56 AM, Omotayo adefilabi...@gmail.com wrote: Hello, That should be on the sip trunk right? I am not sure i checked that. i will confirm today and give you update Regards On Mon, Mar 22, 2010 at 2:07 AM, Otto Sanchez o...@ipexpert.comwrote: I meant for the *Out*bound direction, i.e., from ucm to uc, On Sun, Mar 21, 2010 at 5:51 PM, Otto Sanchez o...@ipexpert.comwrote: Hi, Did you take a look at this document? http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.html http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucm_sip/guide/cucintcucmsip050.htmlAlso, make sure the Redirecting Diversion Header Delivery - Inbound is checked, hth, On Fri, Mar 19, 2010 at 1:35 PM, Omotayo adefilabi...@gmail.comwrote: i have been able to get this work. i have checked all doc but no solution I still need help on this thanks On Wed, Mar 17, 2010 at 3:29 PM, Omotayo adefilabi...@gmail.comwrote: Hello, Any ideas? On Wed, Mar 17, 2010 at 9:54 AM, Omotayo adefilabi...@gmail.comwrote: Hello All, On Lab 7, after integrating the UCM with the UC using SIP. Pressing the subscriber button, i get the personal greeting message But, when pstn or a local call dials hq phone 2 or br1 phone 2, i hear Hello Cisco unity connection messaging system from a text tone phone. Any one with an idea why this i s happening NB: I deleted all the preconfigured voicemail port, huntlist, hunt group and hunt pilot on the UCM as the gude does not indicate that it is needed for the integration to wor Thanks for the anticipated response Regards ___ 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 http://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ -- Regards, Otto Sanchez CCIE #25592 (Voice) Support Engineer - IPexpert, Inc. URL: http://www.IPexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCME Octo-line shared line incoming calls
On CME i'm having issues with octo-line configuration. I am trying to meet the following requirements: 1. Shared line extension 5002 on ephone 1 and ephone 2. 2. Ephone 1 should allow only 2 incoming calls to the shared line. 3. Ephone 2 should allow only 3 incoming calls to the shared line. 4. The shared line should be limited to 4 concurrent calls. I was thinking about using busy-trigger-per-button on each phone along with huntstop channel 4 configured on the shared DN but this does not work. Any ideas on how to achieve this will be be much appreciated. Thanks, Andrew ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SIP SRST - What application to use?
All - Here's a sample section from a SIP SRST setup from the SIP SRND Admin Guide: voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 application sip.app preference 2 incoming called-number cor incoming css-internal default codec g711ulaw What the heck is this application command used for? Later on, I came across this config example: ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP SRST - What application to use?
Hello: The second example is not shown... My experience tell me that if you use application sip.app the gw won't find the app, then you will need application global service alternate Default (similar to mgcp srst) this way the gw will use h323 and call will work. A better aproach that worked for me is just delete this command application sip.app I know that this doesn't answer your question but could help Regards From: mjbe...@krollontrack.com To: ccie_voice@onlinestudylist.com Date: Thu, 25 Mar 2010 06:00:35 -0500 Subject: [OSL | CCIE_Voice] SIP SRST - What application to use? All - Here's a sample section from a SIP SRST setup from the SIP SRND Admin Guide: voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 application sip.app preference 2 incoming called-number cor incoming css-internal default codec g711ulaw What the heck is this application command used for? Later on, I came across this config example: _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP SRST - What application to use?
So what you're saying is that SIP SRST seems to work properly even without the sip.app application specified? I haven't been able to tell a different without the application, which is what raised the question about its function. M From: Angel Perez [mailto:gorr...@hotmail.com] Sent: Thursday, March 25, 2010 6:16 AM To: Berry, Matthew J.; osl osl Subject: RE: [OSL | CCIE_Voice] SIP SRST - What application to use? Hello: The second example is not shown... My experience tell me that if you use application sip.app the gw won't find the app, then you will need application global service alternate Default (similar to mgcp srst) this way the gw will use h323 and call will work. A better aproach that worked for me is just delete this command application sip.app I know that this doesn't answer your question but could help Regards From: mjbe...@krollontrack.com To: ccie_voice@onlinestudylist.com Date: Thu, 25 Mar 2010 06:00:35 -0500 Subject: [OSL | CCIE_Voice] SIP SRST - What application to use? All - Here's a sample section from a SIP SRST setup from the SIP SRND Admin Guide: voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 application sip.app preference 2 incoming called-number cor incoming css-internal default codec g711ulaw What the heck is this application command used for? Later on, I came across this config example: Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP SRST - What application to use?
According to the SIP SRST Admin Guide (http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/sipsrst/conf iguration/guide/spsrst2.html): application application-name Selects the session-level application on the VoIP dial peer. Use the application-name argument to define a specific interactive voice response (IVR) application. Example: Router(config-register-pool)# application SIP.App I haven't played with this very much, so real-world anecdotes are welcomed. :) cheers, sd From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Berry, Matthew J. Sent: Thursday, March 25, 2010 8:39 AM To: Angel Perez; osl osl Subject: Re: [OSL | CCIE_Voice] SIP SRST - What application to use? So what you're saying is that SIP SRST seems to work properly even without the sip.app application specified? I haven't been able to tell a different without the application, which is what raised the question about its function. M From: Angel Perez [mailto:gorr...@hotmail.com] Sent: Thursday, March 25, 2010 6:16 AM To: Berry, Matthew J.; osl osl Subject: RE: [OSL | CCIE_Voice] SIP SRST - What application to use? Hello: The second example is not shown... My experience tell me that if you use application sip.app the gw won't find the app, then you will need application global service alternate Default (similar to mgcp srst) this way the gw will use h323 and call will work. A better aproach that worked for me is just delete this command application sip.app I know that this doesn't answer your question but could help Regards From: mjbe...@krollontrack.com To: ccie_voice@onlinestudylist.com Date: Thu, 25 Mar 2010 06:00:35 -0500 Subject: [OSL | CCIE_Voice] SIP SRST - What application to use? All - Here's a sample section from a SIP SRST setup from the SIP SRND Admin Guide: voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 application sip.app preference 2 incoming called-number cor incoming css-internal default codec g711ulaw What the heck is this application command used for? Later on, I came across this config example: Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 6A - AAR task
Hi Angel, Thank you for your response. I am little bit unclear with what you want me to do. I thought when you call between sites, you dial 91 + the 10 digit number. So, if I am calling from HQ (5002) to BR1(1002), I will need to dial 916178631002, correct? Also, I changed the AAR number on the BR1-Phone 2 to 6178632683 (PSTN-BR1 #) and it works fine. As you said, it looks like the BR1 RTR does not like the 10 digit format. But, BR1 RTR being a MGCP G/W, I can't use dial-peers to format the incoming called number and I thought when I look for 4 significant digits on the BR1 G/W incoming call handling, it should take care of this. I have the EPNM = 6178631XXX and for AAR mask , I tried both 6178631002 and blank. I also tried this. Called from PSTN HQ # to 6178631002. When the call comes into BR1 G/W, on the debug isdn Q931, it comes in as 8631002. Thanks Kalyan On Thu, Mar 25, 2010 at 12:03 AM, Kalyan iyer kparam2...@gmail.com wrote: Hi guys, I was working on lab 6A to set up AAR between the HQ and BR1. I was successful in setting the rsvp, reducing the bandwidth and forcing the call to use AAR and go over the PSTN. I was able to complete the call when i call from BR1 (1002) to HQ (5002). But when I dial from HQ(5002) to BR1(1002), I get a busy tone and the dreaded Unallocated/Unassigned isdn message on BR1 I have css-br-ld as the incoming CSS on the BR1 RTR and that css has pt-internal. 1002 is assigned pt-internal. I am not sure what I am missing here. Here are the debug outputs from HQ and BR1 RTR. on HQ Router: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=11 Mar 25 00:58:30.581: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 Mar 25 00:58:30.581: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Calling num 2123945002 Mar 25 00:58:30.585: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Called num 16178631002 Mar 25 00:58:30.585: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0097 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'HQ Phone 2' Calling Party Number i = 0x2181, '2123945002' Plan:ISDN, Type:National Called Party Number i = 0x80, '16178631002' Plan:Unknown, Type:Unknown Redirecting Number i = 0x81, '1002' Plan:Unknown, Type:Unknown HQ-RTR# Mar 25 00:58:30.621: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8097 Channel ID i = 0xA98383 Exclusive, Channel 3 Mar 25 00:58:30.657: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8 callref = 0x8097 Cause i = 0x8281 - Unallocated/unassigned number Mar 25 00:58:30.665: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8 callref = 0x0097 Mar 25 00:58:30.673: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x8097 on BR1 RTR: BR1-RTR# Mar 25 00:58:30.656: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0098 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Display i = 'HQ Phone 2' Calling Party Number i = 0x2181, '2123945002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '6178631002' Plan:ISDN, Type:National Redirecting Number i = 0x81, '1002' Plan:Unknown, Type:Unknown Mar 25 00:58:30.668: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x8098 Cause i = 0x8081 - Unallocated/unassigned number Any input will be greatly appreciated. Thanks Kalyan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP SRST - What application to use?
Yes, in my lab everything looks find without the command From: mjbe...@krollontrack.com To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Date: Thu, 25 Mar 2010 07:39:23 -0500 Subject: RE: [OSL | CCIE_Voice] SIP SRST - What application to use? So what you’re saying is that SIP SRST seems to work properly even without the sip.app application specified? I haven’t been able to tell a different without the application, which is what raised the question about its function. M From: Angel Perez [mailto:gorr...@hotmail.com] Sent: Thursday, March 25, 2010 6:16 AM To: Berry, Matthew J.; osl osl Subject: RE: [OSL | CCIE_Voice] SIP SRST - What application to use? Hello: The second example is not shown... My experience tell me that if you use application sip.app the gw won't find the app, then you will need application global service alternate Default (similar to mgcp srst) this way the gw will use h323 and call will work. A better aproach that worked for me is just delete this command application sip.app I know that this doesn't answer your question but could help Regards From: mjbe...@krollontrack.com To: ccie_voice@onlinestudylist.com Date: Thu, 25 Mar 2010 06:00:35 -0500 Subject: [OSL | CCIE_Voice] SIP SRST - What application to use? All - Here's a sample section from a SIP SRST setup from the SIP SRND Admin Guide: voice register pool 1 id network 10.10.201.0 mask 255.255.255.0 application sip.app preference 2 incoming called-number cor incoming css-internal default codec g711ulaw What the heck is this application command used for? Later on, I came across this config example: Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 6A - AAR task
Hi: This is your scenario: hq --(1)-- pstn --(2)-- br1, Let say for example hq call br1 via AAR, 5002 call 1002, if there isn't enough bandwith and AAR is enable, 5002 will take 1002 EPNM 6178631XXX, plus AAR prefix 91 - 916178631002 - this should match a router pattern in partition that phone AAR css should see (9 should be stripped) this route pattern should point to PSTN, if the call is droped as it try to ingress at pstn (1) the the problem is that the the number 16178631002 is not a matchin an outgoing dial-peer to BR1, if the call fails at it egress from the pstn (2) then the problem is that the called number you are getting from pstn is not matching a number at CCM (probably a css problem, significant digits, etc are incorrect). For example if significant digits is set to 4 and css of gw can see 5002 the called number should be set to 5002 and match the dn Same apply in the opposite direction I don't know the detail of your ccm (EPNM, AAR prefix, css, etc) neither the pstn details, so first isolate the problem between (1) and (2) and then just check the pstn dial-peer and the called number you get at (1) and also the called number you get at (2) and the br1 gw config related to called number hth Date: Thu, 25 Mar 2010 09:54:05 -0400 From: kparam2...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab 6A - AAR task Hi Angel, Thank you for your response. I am little bit unclear with what you want me to do. I thought when you call between sites, you dial 91 + the 10 digit number. So, if I am calling from HQ (5002) to BR1(1002), I will need to dial 916178631002, correct? Also, I changed the AAR number on the BR1-Phone 2 to 6178632683 (PSTN-BR1 #) and it works fine. As you said, it looks like the BR1 RTR does not like the 10 digit format. But, BR1 RTR being a MGCP G/W, I can't use dial-peers to format the incoming called number and I thought when I look for 4 significant digits on the BR1 G/W incoming call handling, it should take care of this. I have the EPNM = 6178631XXX and for AAR mask , I tried both 6178631002 and blank. I also tried this. Called from PSTN HQ # to 6178631002. When the call comes into BR1 G/W, on the debug isdn Q931, it comes in as 8631002. Thanks Kalyan On Thu, Mar 25, 2010 at 12:03 AM, Kalyan iyer kparam2...@gmail.com wrote: Hi guys, I was working on lab 6A to set up AAR between the HQ and BR1. I was successful in setting the rsvp, reducing the bandwidth and forcing the call to use AAR and go over the PSTN. I was able to complete the call when i call from BR1 (1002) to HQ (5002). But when I dial from HQ(5002) to BR1(1002), I get a busy tone and the dreaded Unallocated/Unassigned isdn message on BR1 I have css-br-ld as the incoming CSS on the BR1 RTR and that css has pt-internal. 1002 is assigned pt-internal. I am not sure what I am missing here. Here are the debug outputs from HQ and BR1 RTR. on HQ Router: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=11 Mar 25 00:58:30.581: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 Mar 25 00:58:30.581: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Calling num 2123945002 Mar 25 00:58:30.585: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Called num 16178631002 Mar 25 00:58:30.585: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0097 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'HQ Phone 2' Calling Party Number i = 0x2181, '2123945002' Plan:ISDN, Type:National Called Party Number i = 0x80, '16178631002' Plan:Unknown, Type:Unknown Redirecting Number i = 0x81, '1002' Plan:Unknown, Type:Unknown HQ-RTR# Mar 25 00:58:30.621: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8097 Channel ID i = 0xA98383 Exclusive, Channel 3 Mar 25 00:58:30.657: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8 callref = 0x8097 Cause i = 0x8281 - Unallocated/unassigned number Mar 25 00:58:30.665: ISDN Se0/0/0:23 Q931: TX - RELEASE pd = 8 callref = 0x0097 Mar 25 00:58:30.673: ISDN Se0/0/0:23 Q931: RX - RELEASE_COMP pd = 8 callref = 0x8097 on BR1 RTR: BR1-RTR# Mar 25 00:58:30.656: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0098 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Display i = 'HQ Phone 2' Calling Party Number i = 0x2181,
Re: [OSL | CCIE_Voice] UCCX script question
No Randall of course i have no problem to share it with the community. i will do so 2010/3/24 Randall Saborio ill2...@gmail.com Lucky Cristobal. Are you concerned about your intellectual property rights, or will you share it with all of us? :) We won't get mad if you want to keep it private, but just suggesting. On Wed, Mar 24, 2010 at 12:03 PM, Cristobal Priego cristobalpri...@gmail.com wrote: thank you very much 2010/3/24 Tanner Ezell tanner.ez...@gmail.com Eh? I've attached a document I developed which explains everything you need to get information from the script to the CAD software. On Wed, Mar 24, 2010 at 12:29 PM, Cristobal Priego cristobalpri...@gmail.com wrote: Hello Tanner, thanks for your reply. I have a question, I'm trying to do my customized Get call Contact Info Step with the Set Enterprise call info step , none of the options that i created on my enterprise data configuration shows up i know I'm not understanding something properly. Can I do a Get call Contact Info Step and push some personalized (customized) Strings to the Agent desktop ? am I on the right track ? thanks 2010/3/23 Tanner Ezell tanner.ez...@gmail.com Add a new variable to the work flow Use the set enterprise call info step to pass the variable along. On Tue, Mar 23, 2010 at 6:26 PM, Cristobal Priego cristobalpri...@gmail.com wrote: Hello, I'd like to get some advice on this. I need to create a script that will get some variables from the customer. and I'd like those variables to be displayed on the Agent Desktop. I've been looking on Enterprise Data Format and on the Desktop Administrator I created a workflow and I Modified a few fields for Enterprise Data and Added a new layout list. Where i'm having problems is at the time to put all of this together. could you please help me which steps do i need to use thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Cheers, Tanner Ezell -- Cheers, Tanner Ezell ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX script question
The document was sent to the list, but apparently has yet to be approved (apparently no one pays attention?). The document can be found here: http://tannerezell.com/media/UCCX%20Custom%20Reports%20-%20Scripting%20-%20Get%20Session%20ID.pdf As an aside, you may want to consider the IP of the person who created the document before you start passing it around :) Cheers On Thu, Mar 25, 2010 at 11:26 AM, Cristobal Priego cristobalpri...@gmail.com wrote: No Randall of course i have no problem to share it with the community. i will do so 2010/3/24 Randall Saborio ill2...@gmail.com Lucky Cristobal. Are you concerned about your intellectual property rights, or will you share it with all of us? :) We won't get mad if you want to keep it private, but just suggesting. On Wed, Mar 24, 2010 at 12:03 PM, Cristobal Priego cristobalpri...@gmail.com wrote: thank you very much 2010/3/24 Tanner Ezell tanner.ez...@gmail.com Eh? I've attached a document I developed which explains everything you need to get information from the script to the CAD software. On Wed, Mar 24, 2010 at 12:29 PM, Cristobal Priego cristobalpri...@gmail.com wrote: Hello Tanner, thanks for your reply. I have a question, I'm trying to do my customized Get call Contact Info Step with the Set Enterprise call info step , none of the options that i created on my enterprise data configuration shows up i know I'm not understanding something properly. Can I do a Get call Contact Info Step and push some personalized (customized) Strings to the Agent desktop ? am I on the right track ? thanks 2010/3/23 Tanner Ezell tanner.ez...@gmail.com Add a new variable to the work flow Use the set enterprise call info step to pass the variable along. On Tue, Mar 23, 2010 at 6:26 PM, Cristobal Priego cristobalpri...@gmail.com wrote: Hello, I'd like to get some advice on this. I need to create a script that will get some variables from the customer. and I'd like those variables to be displayed on the Agent Desktop. I've been looking on Enterprise Data Format and on the Desktop Administrator I created a workflow and I Modified a few fields for Enterprise Data and Added a new layout list. Where i'm having problems is at the time to put all of this together. could you please help me which steps do i need to use thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Cheers, Tanner Ezell -- Cheers, Tanner Ezell ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Cheers, Tanner Ezell ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] vouchers
Hi List, I've discussed this with PL team and taken their permission before posting this. I've around 20 odd vouchers available at minimal price. Those are left overs after passing my lab. If anyone interested pl PM me. All vouchers are valid for V3. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Gatekeeper registration questions
All, Can anyone explain the core of how a gateway registration is recorded with the sh gatek endp command. Normally, as I understand it, a full H323 gw will register to a gk as type H323-GW, whereas the UCM will register as VOIP-GW. However, I'm running into a situation where no matter what I do my BR2 endpoint is registering with the HQ gk as VOIP-GW. I found reference elsewhere to a 323gw will register as voip-gw if it is configured as a CUBE. In this case though it isn't configured as a CUBE. I've put the pertinent VERY basic commands below. What I'm really looking for is some clarification on the different types and why they register differently if possible. Thanks for any help, Paul (#16842 RS/Sec) HQ-RTR#sh run | s gatekeeper gatekeeper zone local HQ ipexpert.com 10.10.110.1 no shutdown HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 10.10.110.3 1720 10.10.110.3 65184 HQVOIP-GW E164-ID: 3102 H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.1.200.20 48779 10.1.200.20 32824 HQVOIP-GW H323-ID: HQgk_1 Voice Capacity Max.= Avail.= Current.= 0 10.1.200.21 37078 10.1.200.21 32849 HQVOIP-GW H323-ID: HQgk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registratio BR2-RTRen BR2-RTR#sh run int Loop0 Building configuration... Current configuration : 163 bytes ! interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip h323-id BR2-RTR end BR2-RTR#sh gateway H.323 ITU-T Version: 4.0 H323 Stack Version: 0.1 H.323 service is up Gateway BR2-RTR is registered to Gatekeeper HQ Alias list (CLI configured) E164-ID 3102 H323-ID BR2-RTR Alias list (last RCF) E164-ID 3102 H323-ID BR2-RTR H323 resource thresholding is Disabled BR2-RTR# BR2-RTR#sh run | s voice service voice service voip ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 6A - AAR task
Hi Angel, I totally follow your logic and explanation. When calling from HQ to BR1, I see the call on the BR1 RTR (from the previous debugs that were attached). I checked and rechecked on the CallManager to make sure on the BR1 G/W that I have the significant digits set to 4, CSS set css-br1-ld on the G/W which includes pt-internal (1002). Thats why I was asking if I missed something. I infact redid the lab twice and got the same problem. I am wondering if anyone else had the same problem and what they did to fix this? Thanks Kalyan On Thu, Mar 25, 2010 at 10:51 AM, Angel Perez gorr...@hotmail.com wrote: Hi: This is your scenario: hq --(1)-- pstn --(2)-- br1, Let say for example hq call br1 via AAR, 5002 call 1002, if there isn't enough bandwith and AAR is enable, 5002 will take 1002 EPNM 6178631XXX, plus AAR prefix 91 - 916178631002 - this should match a router pattern in partition that phone AAR css should see (9 should be stripped) this route pattern should point to PSTN, if the call is droped as it try to ingress at pstn (1) the the problem is that the the number 16178631002 is not a matchin an outgoing dial-peer to BR1, if the call fails at it egress from the pstn (2) then the problem is that the called number you are getting from pstn is not matching a number at CCM (probably a css problem, significant digits, etc are incorrect). For example if significant digits is set to 4 and css of gw can see 5002 the called number should be set to 5002 and match the dn Same apply in the opposite direction I don't know the detail of your ccm (EPNM, AAR prefix, css, etc) neither the pstn details, so first isolate the problem between (1) and (2) and then just check the pstn dial-peer and the called number you get at (1) and also the called number you get at (2) and the br1 gw config related to called number hth -- Date: Thu, 25 Mar 2010 09:54:05 -0400 From: kparam2...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab 6A - AAR task Hi Angel, Thank you for your response. I am little bit unclear with what you want me to do. I thought when you call between sites, you dial 91 + the 10 digit number. So, if I am calling from HQ (5002) to BR1(1002), I will need to dial 916178631002, correct? Also, I changed the AAR number on the BR1-Phone 2 to 6178632683 (PSTN-BR1 #) and it works fine. As you said, it looks like the BR1 RTR does not like the 10 digit format. But, BR1 RTR being a MGCP G/W, I can't use dial-peers to format the incoming called number and I thought when I look for 4 significant digits on the BR1 G/W incoming call handling, it should take care of this. I have the EPNM = 6178631XXX and for AAR mask , I tried both 6178631002 and blank. I also tried this. Called from PSTN HQ # to 6178631002. When the call comes into BR1 G/W, on the debug isdn Q931, it comes in as 8631002. Thanks Kalyan On Thu, Mar 25, 2010 at 12:03 AM, Kalyan iyer kparam2...@gmail.comwrote: Hi guys, I was working on lab 6A to set up AAR between the HQ and BR1. I was successful in setting the rsvp, reducing the bandwidth and forcing the call to use AAR and go over the PSTN. I was able to complete the call when i call from BR1 (1002) to HQ (5002). But when I dial from HQ(5002) to BR1(1002), I get a busy tone and the dreaded Unallocated/Unassigned isdn message on BR1 I have css-br-ld as the incoming CSS on the BR1 RTR and that css has pt-internal. 1002 is assigned pt-internal. I am not sure what I am missing here. Here are the debug outputs from HQ and BR1 RTR. on HQ Router: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=11 Mar 25 00:58:30.581: ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13 Mar 25 00:58:30.581: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Calling num 2123945002 Mar 25 00:58:30.585: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x0 0x0, Called num 16178631002 Mar 25 00:58:30.585: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x0097 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'HQ Phone 2' Calling Party Number i = 0x2181, '2123945002' Plan:ISDN, Type:National Called Party Number i = 0x80, '16178631002' Plan:Unknown, Type:Unknown Redirecting Number i = 0x81, '1002' Plan:Unknown, Type:Unknown HQ-RTR# Mar 25 00:58:30.621: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8097 Channel ID i = 0xA98383 Exclusive, Channel 3 Mar 25 00:58:30.657: ISDN Se0/0/0:23 Q931: RX - DISCONNECT pd = 8
Re: [OSL | CCIE_Voice] vouchers
Kapil, We in no way condone soliciting on this list. This is strictly a CCIE study list and all posts should only contain topics that pertain to CCIE Study. Consider this you first warning, a second offense will result in expulsion from this list. --Started: 23 Mar 2010 9:05:33- Kapil Atrish: actually I want to offer them on the voice alias, if someone interested in buying Drew LePla: We do not allow soliciting on OSL. Kapil Atrish: I thought of checking with you before blindly posting it Drew LePla: Yeah they frown on that just strictly a support forum for CCIE topics. Regards, Drew LePla - COMP TIA A+, CCNA - IPexpert Lead Technical Support Engineer Mailto: mailto:dle...@ipexpert.com dle...@ipexpert.com Telephone: +1.810.326.1444, ext. 204 Live Assistance, Please visit: http://www.ipexpert.com/chat www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Classroom and Self-Study Cisco CCNA (RS, Voice Security), CCNP, CCVP, CCSP and CCIE (RS, Voice, Security Service Provider) Certification Training with locations throughout the United States, Europe and Australia. Be sure to check out our online communities at http://www.ipexpert.com/communities www.ipexpert.com/communities and our public website at http://www.ipexpert.com www.ipexpert.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kapil atrish Sent: Thursday, March 25, 2010 12:36 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] vouchers Hi List, I've discussed this with PL team and taken their permission before posting this. I've around 20 odd vouchers available at minimal price. Those are left overs after passing my lab. If anyone interested pl PM me. All vouchers are valid for V3. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX script question
Hello Tanner,
I have a question. I know I'm doing something wrong. when the call is sent
to the agent. on the Data Field part. I see ANI, DNIS, Layout (default)
what I'm trying to do is, based on the called number i will make a
comparison in a loop, when i find a match in the script based on the called
number, i want to set a variable with a name. then i would like the agent to
be able to see ANI, DNIS and the customized name on the Data Field.
i'd like to do something like this
Called Number = 2003
if (DNIS == 2001 )
{
Customer = Safeway
}
else if (DNIS== 2002)
{
Customer = Raleys
}
else if (DNIS == 2003)
{
Customer = SafeMart
}
then I'd like the agent to see on the Data field when the phone is ringing
on the agent
ANI = 408-123-4567
DNIS = 2003
Customer = SafeMart
I don't know or I can't get the last part to work
I have enhanced license on my uccx version 5.0.2.
on the desktop administrator under Enterprise Data Configuration
I created my field list:
0 customer
and layout list:
Customer : ANI, DNIS, Customer, user.layout
i went to workflow groups agents key accounts work flow,
on the event Ringing Action I only have these options: Run Macro, Call
Control, Launch External App, Agent state, Utility Action
and i don't know what to do
how do i get this to work ?
thanks
2010/3/25 Tanner Ezell tanner.ez...@gmail.com
The document was sent to the list, but apparently has yet to be
approved (apparently no one pays attention?).
The document can be found here:
http://tannerezell.com/media/UCCX%20Custom%20Reports%20-%20Scripting%20-%20Get%20Session%20ID.pdf
As an aside, you may want to consider the IP of the person who created
the document before you start passing it around :)
Cheers
On Thu, Mar 25, 2010 at 11:26 AM, Cristobal Priego
cristobalpri...@gmail.com wrote:
No Randall of course i have no problem to share it with the community. i
will do so
2010/3/24 Randall Saborio ill2...@gmail.com
Lucky Cristobal.
Are you concerned about your intellectual property rights, or will you
share it with all of us? :)
We won't get mad if you want to keep it private, but just suggesting.
On Wed, Mar 24, 2010 at 12:03 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
thank you very much
2010/3/24 Tanner Ezell tanner.ez...@gmail.com
Eh?
I've attached a document I developed which explains everything you
need to get information from the script to the CAD software.
On Wed, Mar 24, 2010 at 12:29 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
thanks for your reply.
I have a question, I'm trying to do my customized Get call Contact
Info
Step
with the Set Enterprise call info step , none of the options that i
created
on my enterprise data configuration shows up
i know I'm not understanding something properly. Can I do a Get
call
Contact
Info Step and push some personalized (customized) Strings to the
Agent
desktop ?
am I on the right track ?
thanks
2010/3/23 Tanner Ezell tanner.ez...@gmail.com
Add a new variable to the work flow
Use the set enterprise call info step to pass the variable along.
On Tue, Mar 23, 2010 at 6:26 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello,
I'd like to get some advice on this.
I need to create a script that will get some variables from the
customer.
and I'd like those variables to be displayed on the Agent
Desktop.
I've been looking on Enterprise Data Format and on the Desktop
Administrator
I created a workflow and I Modified a few fields for Enterprise
Data
and
Added a new layout list.
Where i'm having problems is at the time to put all of this
together.
could
you please help me
which steps do i need to use
thanks
___
For more information regarding industry leading CCIE Lab
training,
please
visit www.ipexpert.com
--
Cheers,
Tanner Ezell
--
Cheers,
Tanner Ezell
___
For more information regarding industry leading CCIE Lab training,
please
visit www.ipexpert.com
--
Cheers,
Tanner Ezell
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX script question
I'd have to check, but I don't believe the ECC fields are populated in
CAD while the phone is ringing. Only after.
Aside from that, you must use the Set Enterprise Info step, make sure
the field is added to the default layout of the agent and you'll be
set.
---
I just re-read what you posted, and it sounds like you created another
layout list, if that is correct you must specify it with the Set
Enterprise Info step, or add it to the default layout.
On Thu, Mar 25, 2010 at 2:16 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
I have a question. I know I'm doing something wrong. when the call is sent
to the agent. on the Data Field part. I see ANI, DNIS, Layout (default)
what I'm trying to do is, based on the called number i will make a
comparison in a loop, when i find a match in the script based on the called
number, i want to set a variable with a name. then i would like the agent to
be able to see ANI, DNIS and the customized name on the Data Field.
i'd like to do something like this
Called Number = 2003
if (DNIS == 2001 )
{
Customer = Safeway
}
else if (DNIS== 2002)
{
Customer = Raleys
}
else if (DNIS == 2003)
{
Customer = SafeMart
}
then I'd like the agent to see on the Data field when the phone is ringing
on the agent
ANI = 408-123-4567
DNIS = 2003
Customer = SafeMart
I don't know or I can't get the last part to work
I have enhanced license on my uccx version 5.0.2.
on the desktop administrator under Enterprise Data Configuration
I created my field list:
0 customer
and layout list:
Customer : ANI, DNIS, Customer, user.layout
i went to workflow groups agents key accounts work flow,
on the event Ringing Action I only have these options: Run Macro, Call
Control, Launch External App, Agent state, Utility Action
and i don't know what to do
how do i get this to work ?
thanks
2010/3/25 Tanner Ezell tanner.ez...@gmail.com
The document was sent to the list, but apparently has yet to be
approved (apparently no one pays attention?).
The document can be found here:
http://tannerezell.com/media/UCCX%20Custom%20Reports%20-%20Scripting%20-%20Get%20Session%20ID.pdf
As an aside, you may want to consider the IP of the person who created
the document before you start passing it around :)
Cheers
On Thu, Mar 25, 2010 at 11:26 AM, Cristobal Priego
cristobalpri...@gmail.com wrote:
No Randall of course i have no problem to share it with the community. i
will do so
2010/3/24 Randall Saborio ill2...@gmail.com
Lucky Cristobal.
Are you concerned about your intellectual property rights, or will you
share it with all of us? :)
We won't get mad if you want to keep it private, but just suggesting.
On Wed, Mar 24, 2010 at 12:03 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
thank you very much
2010/3/24 Tanner Ezell tanner.ez...@gmail.com
Eh?
I've attached a document I developed which explains everything you
need to get information from the script to the CAD software.
On Wed, Mar 24, 2010 at 12:29 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
thanks for your reply.
I have a question, I'm trying to do my customized Get call
Contact
Info
Step
with the Set Enterprise call info step , none of the options that
i
created
on my enterprise data configuration shows up
i know I'm not understanding something properly. Can I do a Get
call
Contact
Info Step and push some personalized (customized) Strings to the
Agent
desktop ?
am I on the right track ?
thanks
2010/3/23 Tanner Ezell tanner.ez...@gmail.com
Add a new variable to the work flow
Use the set enterprise call info step to pass the variable along.
On Tue, Mar 23, 2010 at 6:26 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello,
I'd like to get some advice on this.
I need to create a script that will get some variables from the
customer.
and I'd like those variables to be displayed on the Agent
Desktop.
I've been looking on Enterprise Data Format and on the Desktop
Administrator
I created a workflow and I Modified a few fields for Enterprise
Data
and
Added a new layout list.
Where i'm having problems is at the time to put all of this
together.
could
you please help me
which steps do i need to use
thanks
___
For more information regarding industry leading CCIE Lab
training,
please
visit www.ipexpert.com
--
Cheers,
Tanner Ezell
--
Cheers,
Tanner Ezell
___
For more information regarding industry leading CCIE Lab training,
Re: [OSL | CCIE_Voice] UCCX script question
They are populated while ringing and they are only called ecc's when
using cad with ucce not uccx. I think tanner hit the nail on the head
though with the layout and set enterprise data steps.
Chad Stachowicz
415-794-8770
Please excuse any mispellings as this message was from my mobile device
On Mar 25, 2010, at 9:22 AM, Tanner Ezell tanner.ez...@gmail.com
wrote:
I'd have to check, but I don't believe the ECC fields are populated in
CAD while the phone is ringing. Only after.
Aside from that, you must use the Set Enterprise Info step, make sure
the field is added to the default layout of the agent and you'll be
set.
---
I just re-read what you posted, and it sounds like you created another
layout list, if that is correct you must specify it with the Set
Enterprise Info step, or add it to the default layout.
On Thu, Mar 25, 2010 at 2:16 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
I have a question. I know I'm doing something wrong. when the call
is sent
to the agent. on the Data Field part. I see ANI, DNIS, Layout
(default)
what I'm trying to do is, based on the called number i will make a
comparison in a loop, when i find a match in the script based on
the called
number, i want to set a variable with a name. then i would like the
agent to
be able to see ANI, DNIS and the customized name on the Data Field.
i'd like to do something like this
Called Number = 2003
if (DNIS == 2001 )
{
Customer = Safeway
}
else if (DNIS== 2002)
{
Customer = Raleys
}
else if (DNIS == 2003)
{
Customer = SafeMart
}
then I'd like the agent to see on the Data field when the phone is
ringing
on the agent
ANI = 408-123-4567
DNIS = 2003
Customer = SafeMart
I don't know or I can't get the last part to work
I have enhanced license on my uccx version 5.0.2.
on the desktop administrator under Enterprise Data Configuration
I created my field list:
0 customer
and layout list:
Customer : ANI, DNIS, Customer, user.layout
i went to workflow groups agents key accounts work flow,
on the event Ringing Action I only have these options: Run Macro,
Call
Control, Launch External App, Agent state, Utility Action
and i don't know what to do
how do i get this to work ?
thanks
2010/3/25 Tanner Ezell tanner.ez...@gmail.com
The document was sent to the list, but apparently has yet to be
approved (apparently no one pays attention?).
The document can be found here:
http://tannerezell.com/media/UCCX%20Custom%20Reports%20-%20Scripting%20-%20Get%20Session%20ID.pdf
As an aside, you may want to consider the IP of the person who
created
the document before you start passing it around :)
Cheers
On Thu, Mar 25, 2010 at 11:26 AM, Cristobal Priego
cristobalpri...@gmail.com wrote:
No Randall of course i have no problem to share it with the
community. i
will do so
2010/3/24 Randall Saborio ill2...@gmail.com
Lucky Cristobal.
Are you concerned about your intellectual property rights, or
will you
share it with all of us? :)
We won't get mad if you want to keep it private, but just
suggesting.
On Wed, Mar 24, 2010 at 12:03 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
thank you very much
2010/3/24 Tanner Ezell tanner.ez...@gmail.com
Eh?
I've attached a document I developed which explains everything
you
need to get information from the script to the CAD software.
On Wed, Mar 24, 2010 at 12:29 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
thanks for your reply.
I have a question, I'm trying to do my customized Get call
Contact
Info
Step
with the Set Enterprise call info step , none of the options
that
i
created
on my enterprise data configuration shows up
i know I'm not understanding something properly. Can I do a Get
call
Contact
Info Step and push some personalized (customized) Strings to
the
Agent
desktop ?
am I on the right track ?
thanks
2010/3/23 Tanner Ezell tanner.ez...@gmail.com
Add a new variable to the work flow
Use the set enterprise call info step to pass the variable
along.
On Tue, Mar 23, 2010 at 6:26 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello,
I'd like to get some advice on this.
I need to create a script that will get some variables from
the
customer.
and I'd like those variables to be displayed on the Agent
Desktop.
I've been looking on Enterprise Data Format and on the
Desktop
Administrator
I created a workflow and I Modified a few fields for
Enterprise
Data
and
Added a new layout list.
Where i'm having problems is at the time to put all of this
together.
could
you please help me
which steps do i need to use
thanks
___
For more information regarding industry leading CCIE Lab
training,
please
visit www.ipexpert.com
--
Cheers,
Tanner Ezell
--
Cheers,
Re: [OSL | CCIE_Voice] UCCX script question
I was able to get it to work
thank you very much for all your help
2010/3/25 Chad Stachowicz chadstachow...@gmail.com
They are populated while ringing and they are only called ecc's when using
cad with ucce not uccx. I think tanner hit the nail on the head though with
the layout and set enterprise data steps.
Chad Stachowicz
415-794-8770
Please excuse any mispellings as this message was from my mobile device
On Mar 25, 2010, at 9:22 AM, Tanner Ezell tanner.ez...@gmail.com wrote:
I'd have to check, but I don't believe the ECC fields are populated in
CAD while the phone is ringing. Only after.
Aside from that, you must use the Set Enterprise Info step, make sure
the field is added to the default layout of the agent and you'll be
set.
---
I just re-read what you posted, and it sounds like you created another
layout list, if that is correct you must specify it with the Set
Enterprise Info step, or add it to the default layout.
On Thu, Mar 25, 2010 at 2:16 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
I have a question. I know I'm doing something wrong. when the call is
sent
to the agent. on the Data Field part. I see ANI, DNIS, Layout (default)
what I'm trying to do is, based on the called number i will make a
comparison in a loop, when i find a match in the script based on the
called
number, i want to set a variable with a name. then i would like the agent
to
be able to see ANI, DNIS and the customized name on the Data Field.
i'd like to do something like this
Called Number = 2003
if (DNIS == 2001 )
{
Customer = Safeway
}
else if (DNIS== 2002)
{
Customer = Raleys
}
else if (DNIS == 2003)
{
Customer = SafeMart
}
then I'd like the agent to see on the Data field when the phone is
ringing
on the agent
ANI = 408-123-4567
DNIS = 2003
Customer = SafeMart
I don't know or I can't get the last part to work
I have enhanced license on my uccx version 5.0.2.
on the desktop administrator under Enterprise Data Configuration
I created my field list:
0 customer
and layout list:
Customer : ANI, DNIS, Customer, user.layout
i went to workflow groups agents key accounts work flow,
on the event Ringing Action I only have these options: Run Macro, Call
Control, Launch External App, Agent state, Utility Action
and i don't know what to do
how do i get this to work ?
thanks
2010/3/25 Tanner Ezell tanner.ez...@gmail.com
The document was sent to the list, but apparently has yet to be
approved (apparently no one pays attention?).
The document can be found here:
http://tannerezell.com/media/UCCX%20Custom%20Reports%20-%20Scripting%20-%20Get%20Session%20ID.pdf
As an aside, you may want to consider the IP of the person who created
the document before you start passing it around :)
Cheers
On Thu, Mar 25, 2010 at 11:26 AM, Cristobal Priego
cristobalpri...@gmail.com wrote:
No Randall of course i have no problem to share it with the community.
i
will do so
2010/3/24 Randall Saborio ill2...@gmail.com
Lucky Cristobal.
Are you concerned about your intellectual property rights, or will you
share it with all of us? :)
We won't get mad if you want to keep it private, but just suggesting.
On Wed, Mar 24, 2010 at 12:03 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
thank you very much
2010/3/24 Tanner Ezell tanner.ez...@gmail.com
Eh?
I've attached a document I developed which explains everything you
need to get information from the script to the CAD software.
On Wed, Mar 24, 2010 at 12:29 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello Tanner,
thanks for your reply.
I have a question, I'm trying to do my customized Get call
Contact
Info
Step
with the Set Enterprise call info step , none of the options that
i
created
on my enterprise data configuration shows up
i know I'm not understanding something properly. Can I do a Get
call
Contact
Info Step and push some personalized (customized) Strings to the
Agent
desktop ?
am I on the right track ?
thanks
2010/3/23 Tanner Ezell tanner.ez...@gmail.com
Add a new variable to the work flow
Use the set enterprise call info step to pass the variable along.
On Tue, Mar 23, 2010 at 6:26 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
Hello,
I'd like to get some advice on this.
I need to create a script that will get some variables from the
customer.
and I'd like those variables to be displayed on the Agent
Desktop.
I've been looking on Enterprise Data Format and on the Desktop
Administrator
I created a workflow and I Modified a few fields for Enterprise
Data
and
Added a new layout list.
Where i'm having problems is at the time to put all of this
together.
could
you please help me
which steps do i need to use
thanks
___
For more information regarding industry leading CCIE Lab
training,
please
Re: [OSL | CCIE_Voice] Gatekeeper registration questions
Paul, I haven't played around with forcing the output Type to either VOIP-GW or H323-GW much - but are you sure that your attempts is not backwards? In other words - I had the site C GW configured as a CUBE/with allow-connections configured and my gatekeeper output looked like: GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 60462 GKH323-GW E164-ID: 3001 E164-ID: 3002 E164-ID: 3999 H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 10.10.210.101720 10.10.210.1032885 GKVOIP-GW H323-ID: GK-Trunk_1 Voice Capacity Max.= Avail.= Current.= 0 10.10.210.111720 10.10.210.1132786 GKVOIP-GW H323-ID: GK-Trunk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 I removed the allow connection statements and my output changed to - this would seem to indicate the device will register as a VOIP-GW when it is not acting as a ip to ip GW: HQ-RTR#show gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 51687 GKVOIP-GW E164-ID: 3001 E164-ID: 3002 E164-ID: 3999 H323-ID: CUCME Voice Capacity Max.= Avail.= Current.= 0 10.10.210.101720 10.10.210.1032885 GKVOIP-GW H323-ID: GK-Trunk_1 Voice Capacity Max.= Avail.= Current.= 0 10.10.210.111720 10.10.210.1132786 GKVOIP-GW H323-ID: GK-Trunk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 3 Date: Thu, 25 Mar 2010 12:57:37 -0400 From: pa...@marshallcomm.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Gatekeeper registration questions All, Can anyone explain the core of how a gateway registration is recorded with the “sh gatek endp” command. Normally, as I understand it, a full H323 gw will register to a gk as type “H323-GW”, whereas the UCM will register as “VOIP-GW”. However, I’m running into a situation where no matter what I do my BR2 endpoint is registering with the HQ gk as VOIP-GW. I found reference elsewhere to a 323gw will register as voip-gw if it is configured as a CUBE. In this case though it isn’t configured as a CUBE. I’ve put the pertinent VERY basic commands below. What I’m really looking for is some clarification on the different types and why they register differently if possible. Thanks for any help, Paul (#16842 RS/Sec) HQ-RTR#sh run | s gatekeeper gatekeeper zone local HQ ipexpert.com 10.10.110.1 no shutdown HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.3 1720 10.10.110.3 65184 HQ VOIP-GW E164-ID: 3102 H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.1.200.20 48779 10.1.200.20 32824 HQ VOIP-GW H323-ID: HQgk_1 Voice Capacity Max.= Avail.= Current.= 0 10.1.200.21 37078 10.1.200.21 32849 HQ VOIP-GW H323-ID: HQgk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registratio BR2-RTRen BR2-RTR#sh run int Loop0 Building configuration... Current configuration : 163 bytes ! interface Loopback0 ip address 10.10.110.3 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip h323-id BR2-RTR end BR2-RTR#sh gateway H.323 ITU-T Version: 4.0 H323 Stack Version: 0.1 H.323 service is up Gateway BR2-RTR is registered to Gatekeeper HQ Alias list (CLI configured) E164-ID 3102 H323-ID BR2-RTR Alias list (last RCF) E164-ID 3102 H323-ID BR2-RTR H323 resource thresholding is Disabled BR2-RTR# BR2-RTR#sh run | s voice service voice service voip _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID27925::T:WLMTAGL:ON:WL:en-US:WM_HMP:032010_3___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
