Re: [OSL | CCIE_Voice] Connected number display
I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx -- Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards -- Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? -- Hotmail: Trusted email with 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 -- 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] Lan QOS Scenario
FYI, I think this is too complex to answer simply: mls qos queue-set output 1 threshold 2 40 60 100 200 This line gives threshold 1 40% of the buffer space and threshold 2 60% of the buffer space, total buffer space reserved is 100% and max (threshold 3) is 200%. Reserved at 100% tells the queue to not share any of it's buffers with the other queues. Max at 200% says it can borrow up to 2 times it's buffer space from the shared pool (space available in queues 1, 3 and 4). Threshold 1 is set at 40% of the buffer size (that's calculated from the full buffer allocation of the queue), and threshold 2 is set for 60%. Remember though, these don't have to total 100%, in fact you can set threshold 1 to 300% or 1000% if you want. By setting threshold 1 to 40% you tell it to start dropping at that threshold, it has no affect on threshold 2's drop percentage. By setting threshold 1 to 40 and threshold 2 to 60, you are affectively limiting queue 2 to 60% of it's total buffers (unless you assign something to threshold 3). I would say that unless you were asked to mess with the other thresholds, leave threshold 1 at 100. I would also set threshold 3 (max) to 100% of buffers for this question. Because this potentially allows other COS to dou ble the effective size of queue 2, thus turning your 60% threshold into a 30%. But maybe I'm over thinking it. So my answer would look like this: mls qos queue-set output 1 threshold 2 100 60 100 100 sets threshold 1 to 100 percent of buffers sets threshold 2 to 60 percent of buffers (cos 4) sets threshold 3 to 100 percent of buffers Matt On Jun 8, 2010, at 10:37 PM, Pavan wrote: Looks good as farvas i can tell. Normally you would also enabl priority-queue on the interface Sent from my phone On Jun 8, 2010, at 8:20 PM, jammer jones jammerjone...@gmail.com wrote: Trying to understand this a little better. Cisco's documentation is not written in very clear english. Very frustrating trying to understand the threshold values as well as the shape versus share bandwidth values. QOS. Cos 5 for queue 1 queue 2 queue 3 queue 4 0 similar to lab 2 . Queue one has the 25% of the bandwidth. other bandwidth is shared as 30 40 30. If the queue 2 is saturated by 60% then the cos 4 has to be dropped. Here is what I think it is. Can someone please correct me if i am wrong and provide any positive feedback. ! mls qos srr-queue output cos-map queue 2 threshold 2 4 !maps cos 4 to queue 2 and threshold 2 mls qos srr-queue output cos-map queue 4 threshold 1 0 !maps cos 0 to queue 4 mls qos queue-set output 1 threshold 2 40 60 100 200 ! when queue 2 threshold 2 exceeds 60% cos packets with cos 4 will be dropped mls qos ! ! interface GigabitEthernet1/0/1 description Office_912_lab_a switchport access vlan 48 switchport mode access switchport voice vlan 51 srr-queue bandwidth share 1 30 40 30 ! sets queues 2 - 4 to 30 40 30 srr-queue bandwidth shape 4 0 0 0 ! sets queue 1 to 25% of the link mls qos trust cos spanning-tree portfast ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
Hi Kobel, Is there any document for how to configure the CUP Publish trunk method? I could understand from the posts that we still need to create a SIP trunk, configure it in the CUP Publish service parameter field, and assign each user to a line appearence Anyway, if the CUCM gateway is not configured, then how to tell the CUP to listen to publish information sent by CUCM on the trunk? It is really strange how Cisco left this undocumented!!! Thank you in advance Regards, _ 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] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
Hi, I'm not aware of any document describing this explicitly. This is the only document I know: http://docwiki.cisco.com/wiki/Cisco_Unified_Presence%2C_Release_7.x_--_How_to_Configure_the_SIP_Trunk_on_Cisco_Unified_Communications_Manager#How_to_Configure_the_SIP_Trunk_on_Cisco_Unified_Communications_Manager For me it looks strange, like it's not edited very well (in fact all steps for PUBLISH method are there, just not very clearly described and the headings seem to be incorrect) This is how I did it two times: * usual CUPS initialization wizard * create SIP trunk on CUCM with CUPS IP@ (but no other settings required) * in CUPS presence settings select this trunk (the PUBLISH checkbox is checked by default AFAIK) - this should also change CCM service parameter (PUBLISH trunk to CUPS) * create users and associate them with line appearances * configure IPPM or CUPC I didin't configure SIP trunk security profile, SUBSCRIBE CSS in CUCM, nor presence gateway in CUPS. I did it two times just to make sure that it works. I hope I'm not missing anything. After following those steps, I'm able to see presence in IPPM an CUPS - if you lift the handset on user's line, you see Busy status in IPPM/CUPC. In CCM traces there are also PUBLISH messages visible (sent from CUCM to CUPS). It would be great if somebody could confirm this independently. regards kobel 2010/6/20 Mouhammad Nasser engnasse...@hotmail.com Hi Kobel, Is there any document for how to configure the CUP Publish trunk method? I could understand from the posts that we still need to create a SIP trunk, configure it in the CUP Publish service parameter field, and assign each user to a line appearence Anyway, if the CUCM gateway is not configured, then how to tell the CUP to listen to publish information sent by CUCM on the trunk? It is really strange how Cisco left this undocumented!!! Thank you in advance Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPS presence integration with CUCM - PUBLISH vs. SUBSCRIBE
I missed one question of yours - CUCM PUBLISH trunk is also configured in CUPS (presence - settings). There you can check the checkbox to enable PUBLISH method and select on of the SIP trunk on CUCM which should be used for this purpose. This also changes the CCM service paramter for CCM via AXL. This is how CUPS knows that it should listen to PUBLISH messages. In traces I was able to see PUBLISH messages being sent by CUCM and answered with 200 OK (CSeq: PUBLISH) by CUPS. 2010/6/20 Mouhammad Nasser engnasse...@hotmail.com Hi Kobel, Is there any document for how to configure the CUP Publish trunk method? I could understand from the posts that we still need to create a SIP trunk, configure it in the CUP Publish service parameter field, and assign each user to a line appearence Anyway, if the CUCM gateway is not configured, then how to tell the CUP to listen to publish information sent by CUCM on the trunk? It is really strange how Cisco left this undocumented!!! Thank you in advance Regards, -- Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. 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] Connected number display
Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. >From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Connected number display
Angel, Yes you are correct to play with the display we set calling party or called party transformation. The problem in this scenario is that you have to change the number depending on the gateway it is egressing ... the one I mentioned may do the trick (not sure)I will have to test this on my lab which seriously I don't want to do for a week at least lol. Ash Ashar Siddiqui wrote: Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I canimagine to make this work is withto different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XXDDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XXDDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option,br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ 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] Connected number display
ok try this way and i am assuming u r calling from Br1 first call should go thru BR1-GW if fails then go to BR2-GW and both are MGCP also assuming number your are dialling 972 xxx and make sure under service parameters these parameter should be false Stop Routing on Out of Bandwidth FlagFalse Stop Routing on Unallocated Number Flag False Stop Routing on User Busy Flag False Route Pattern : 9.[2-9]xx Route Partition : BR1-PT Gatway/Route list : BR1-RED-LOC-BR1-BR2-RL Add BR1-GW Calling party transformation mask on xxx (7x Digits) subscriber, Isdn Called party transformation mask PreDot subscriber, Isdn Add BR2-GW Calling party transformation mask on xx (10x Digits) National, Isdn Called party transformation mask PreDot 972 (city code) National, Isdn try it and let me know if u have any question On Sun, Jun 20, 2010 at 6:49 AM, Ashar Siddiqui siddas...@gmail.com wrote: Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx -- Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards -- Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? -- Hotmail: Trusted email with 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 -- 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Connected number display
@Shadow He is not talking about changing ANI. Ash Shadow of Voice wrote: ok try this way and i am assuming u r calling from Br1 first call should go thru BR1-GW if fails then go to BR2-GW and both are MGCP also assuming number your are dialling 972 xxx and make sure under service parameters these parameter should be false Stop Routing on Out of Bandwidth Flag False Stop Routing on Unallocated Number Flag False Stop Routing on User Busy Flag False Route Pattern : 9.[2-9]xx Route Partition : BR1-PT Gatway/Route list : BR1-RED-LOC-BR1-BR2-RL Add BR1-GW Calling party transformation mask on xxx (7x Digits) subscriber, Isdn Called party transformation mask PreDot subscriber, Isdn Add BR2-GW Calling party transformation mask on xx (10x Digits) National, Isdn Called party transformation mask PreDot 972 (city code) National, Isdn try it and let me know if u have any question On Sun, Jun 20, 2010 at 6:49 AM, Ashar Siddiqui siddas...@gmail.com wrote: Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. >From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ 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] Connected number display
Hi: Did you test both rp alone first to make sure it working correctly? Did you shutdown controller at br1 before testing backup path? thx Date: Sun, 20 Jun 2010 11:49:27 +0100 From: siddas...@gmail.com To: voip.ccieci...@gmail.com CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. _ 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] Connected number display
ok sorry for that i didn't get the question i thought he as problem with ANI i'll try to figure it out from previous mails thx Ash On Sun, Jun 20, 2010 at 7:22 AM, Angel Perez gorr...@hotmail.com wrote: Hi: Did you test both rp alone first to make sure it working correctly? Did you shutdown controller at br1 before testing backup path? thx -- Date: Sun, 20 Jun 2010 11:49:27 +0100 From: siddas...@gmail.com To: voip.ccieci...@gmail.com CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx -- Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards -- Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? -- Hotmail: Trusted email with 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 -- Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 -- Hotmail: Free, trusted and rich email service. Get it now.https://signup.live.com/signup.aspx?id=60969 ___ 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
[OSL | CCIE_Voice] Problem Connection between HQ and BR1 (vol2 Lab8)
Hi, I've a connection issue between HQ and BR1, I can't bring the interface dlci 201 up. below my configuration: BR1: interface Serial0/0/1:0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ! router ospf 1 router-id 10.10.101.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! HQ: interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! Also I tried to revert configuration on routers and do everything from start, but still have problem between HQ and BR1. After reconfigure everything connection BR2 and HQ works great. Any idea Thank you Naoufal ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Problem Connection between HQ and BR1 (vol2 Lab8)
Is this on your on kit or one of the PL racks ? What is the status of the frame relay PVCs ? sh frame-relay pvc sh frame-relay pvc 101 etc and the lmi to the frame switch sh frame-relay lmi check that the status messages are being sent and received thus LMI Statistics for interface Serial1/0 (Frame Relay DTE) LMI TYPE = ANSI Invalid Unnumbered info 0 Invalid Prot Disc 0 Invalid dummy Call Ref 0 Invalid Msg Type 0 Invalid Status Message 0 Invalid Lock Shift 0 Invalid Information ID 0 Invalid Report IE Len 0 Invalid Report Request 0 Invalid Keep IE Len 0 Num Status Enq. Sent 18 Num Status msgs Rcvd 18 Num Update Status Rcvd 0 Num Status Timeouts 0 Last Full Status Req 00:00:06 Last Full Status Rcvd 00:00:06 Regards Graham On 20 Jun 2010, at 14:23, naoufal.kerboute wrote: Hi, I've a connection issue between HQ and BR1, I can't bring the interface dlci 201 up. below my configuration: BR1: interface Serial0/0/1:0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ! router ospf 1 router-id 10.10.101.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! HQ: interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! Also I tried to revert configuration on routers and do everything from start, but still have problem between HQ and BR1. After reconfigure everything connection BR2 and HQ works great. Any idea Thank you Naoufal ___ 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
[OSL | CCIE_Voice] RE : Problem Connection be tween HQ and BR1 (vol2 Lab8)
This is on PL racks, Now it's resolved by proctor, but I want to know whats is the problem. Proctor engineer told me to post the issue in the rack then they told me it's resolved, but what is the problem because the configuration seems OK. I'm comfused :s Message d'origine De: Graham Hopkins [mailto:ghopk...@wolf-rock.co.uk] Date: dim. 6/20/2010 1:50 À: naoufal.kerboute Cc: ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] Problem Connection between HQ and BR1 (vol2 Lab8) Is this on your on kit or one of the PL racks ? What is the status of the frame relay PVCs ? sh frame-relay pvc sh frame-relay pvc 101 etc and the lmi to the frame switch sh frame-relay lmi check that the status messages are being sent and received thus LMI Statistics for interface Serial1/0 (Frame Relay DTE) LMI TYPE = ANSI Invalid Unnumbered info 0 Invalid Prot Disc 0 Invalid dummy Call Ref 0 Invalid Msg Type 0 Invalid Status Message 0 Invalid Lock Shift 0 Invalid Information ID 0 Invalid Report IE Len 0 Invalid Report Request 0 Invalid Keep IE Len 0 Num Status Enq. Sent 18 Num Status msgs Rcvd 18 Num Update Status Rcvd 0 Num Status Timeouts 0 Last Full Status Req 00:00:06 Last Full Status Rcvd 00:00:06 Regards Graham On 20 Jun 2010, at 14:23, naoufal.kerboute wrote: Hi, I've a connection issue between HQ and BR1, I can't bring the interface dlci 201 up. below my configuration: BR1: interface Serial0/0/1:0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ! router ospf 1 router-id 10.10.101.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! HQ: interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! Also I tried to revert configuration on routers and do everything from start, but still have problem between HQ and BR1. After reconfigure everything connection BR2 and HQ works great. Any idea Thank you Naoufal ___ 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
[OSL | CCIE_Voice] RE : Problem Connection be tween HQ and BR1 (vol2 Lab8)
The PL engineer told me that The cable was severed so a waste of time Thank you guys Message d'origine De: Graham Hopkins [mailto:ghopk...@wolf-rock.co.uk] Date: dim. 6/20/2010 1:50 À: naoufal.kerboute Cc: ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] Problem Connection between HQ and BR1 (vol2 Lab8) Is this on your on kit or one of the PL racks ? What is the status of the frame relay PVCs ? sh frame-relay pvc sh frame-relay pvc 101 etc and the lmi to the frame switch sh frame-relay lmi check that the status messages are being sent and received thus LMI Statistics for interface Serial1/0 (Frame Relay DTE) LMI TYPE = ANSI Invalid Unnumbered info 0 Invalid Prot Disc 0 Invalid dummy Call Ref 0 Invalid Msg Type 0 Invalid Status Message 0 Invalid Lock Shift 0 Invalid Information ID 0 Invalid Report IE Len 0 Invalid Report Request 0 Invalid Keep IE Len 0 Num Status Enq. Sent 18 Num Status msgs Rcvd 18 Num Update Status Rcvd 0 Num Status Timeouts 0 Last Full Status Req 00:00:06 Last Full Status Rcvd 00:00:06 Regards Graham On 20 Jun 2010, at 14:23, naoufal.kerboute wrote: Hi, I've a connection issue between HQ and BR1, I can't bring the interface dlci 201 up. below my configuration: BR1: interface Serial0/0/1:0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ! router ospf 1 router-id 10.10.101.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! HQ: interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! Also I tried to revert configuration on routers and do everything from start, but still have problem between HQ and BR1. After reconfigure everything connection BR2 and HQ works great. Any idea Thank you Naoufal ___ 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] RE : Problem Connection between HQ and BR1 (vol2 Lab8)
Your Config looks fine to me... Maybe they found an issue in their FR switch? Great troubleshooting notes posted in this thread, though. Would put you on very good footing to take the issue to the proctor in the actual lab. On Jun 20, 2010 9:54 AM, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: This is on PL racks, Now it's resolved by proctor, but I want to know whats is the problem. Proctor engineer told me to post the issue in the rack then they told me it's resolved, but what is the problem because the configuration seems OK. I'm comfused :s Message d'origine De: Graham Hopkins [mailto:ghopk...@wolf-rock.co.ukghopk...@wolf-rock.co.uk ] Date: dim. 6/20/2010 1:50 À: naoufal.kerboute Cc: ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] Problem Connection between HQ and BR1 (vol2 Lab8) Is this on your on kit or one of the PL racks ? What is the status of the frame relay PVCs ? sh frame-relay pvc sh frame-relay pvc 101 etc and the lmi to the frame switch sh frame-relay lmi check that the status messages are being sent and received thus LMI Statistics for interface Serial1/0 (Frame Relay DTE) LMI TYPE = ANSI Invalid Unnumbered info 0 Invalid Prot Disc 0 Invalid dummy Call Ref 0 Invalid Msg Type 0 Invalid Status Message 0 Invalid Lock Shift 0 Invalid Information ID 0 Invalid Report IE Len 0 Invalid Report Request 0 Invalid Keep IE Len 0 Num Status Enq. Sent 18 Num Status msgs Rcvd 18 Num Update Status Rcvd 0 Num Status Timeouts 0 Last Full Status Req 00:00:06 Last Full Status Rcvd 00:00:06 Regards Graham On 20 Jun 2010, at 14:23, naoufal.kerboute wrote: Hi, I've a connection issue between HQ and BR1, I can't bring the interface dlci 201 up. below my configuration: BR1: interface Serial0/0/1:0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.2 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ! router ospf 1 router-id 10.10.101.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! HQ: interface Serial0/0/1:0 no ip address encapsulation frame-relay frame-relay lmi-type ansi ! interface Serial0/0/1:0.1 point-to-point ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ! interface Serial0/0/1:0.2 point-to-point ip address 10.10.112.1 255.255.255.0 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 202 ! router ospf 1 router-id 10.10.100.1 log-adjacency-changes network 10.10.0.0 0.0.255.255 area 0 ! Also I tried to revert configuration on routers and do everything from start, but still have problem between HQ and BR1. After reconfigure everything connection BR2 and HQ works great. Any idea Thank you Naoufal ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] AAR and ANI formatting
Hello, I'm playing with AAR and VM. When out of bandwidth condition occurs, AAR correctly kicks in for different types of calls (between HQ and BR1, direct calls to VM from BR1, for incoming PSTN calls to BR1 forwarded to voicemail in HQ). It seems that the configuration is ok. But I've an issue with ANI format sent to PSTN when AAR is used. All ANI/DNIS manipulation is done on BR1 gateway via Calling/Called Party Transformation Rules. When I make a call from BR1 to VM in HQ via PSTN (explicitly, using 9.12123945600), the ANI is formatted correctly (6178631xxx/subscriber). But when I press the messages button in BR1, I can see following output from debug isdn q931 on BR1 router (outgoing SETUP): Calling Party Number i = 0x0081, '1002' Plan:Unknown, Type:Unknown Called Party Number i = 0xA0, '12123945600' Plan:Unknown, Type:National Redirecting Number i = 0x81, '5600' Plan:Unknown, Type:Unknown Surprisingly, in the CUC port monitor I can see completely different information - please compare with attached screenshot. I needed to configure alternative extension in CUC to correctly recognize the caller as CUC subscriber. It seems that AAR can handle such call correctly, but it doesn't respect the ANI transformation rules on the GW. Have you also observed this behaviour? Is there any workaround? regards kobel attachment: ScreenShot190.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] RE : AAR and ANI formattin g
Try to do apply a translation-rule on the dial peer routing call to UC using ur internal extension. Message d'origine De: ccie_voice-boun...@onlinestudylist.com de la part de kobel Date: dim. 6/20/2010 3:01 À: ccie_voice@onlinestudylist.com Objet : [OSL | CCIE_Voice] AAR and ANI formatting Hello, I'm playing with AAR and VM. When out of bandwidth condition occurs, AAR correctly kicks in for different types of calls (between HQ and BR1, direct calls to VM from BR1, for incoming PSTN calls to BR1 forwarded to voicemail in HQ). It seems that the configuration is ok. But I've an issue with ANI format sent to PSTN when AAR is used. All ANI/DNIS manipulation is done on BR1 gateway via Calling/Called Party Transformation Rules. When I make a call from BR1 to VM in HQ via PSTN (explicitly, using 9.12123945600), the ANI is formatted correctly (6178631xxx/subscriber). But when I press the messages button in BR1, I can see following output from debug isdn q931 on BR1 router (outgoing SETUP): Calling Party Number i = 0x0081, '1002' Plan:Unknown, Type:Unknown Called Party Number i = 0xA0, '12123945600' Plan:Unknown, Type:National Redirecting Number i = 0x81, '5600' Plan:Unknown, Type:Unknown Surprisingly, in the CUC port monitor I can see completely different information - please compare with attached screenshot. I needed to configure alternative extension in CUC to correctly recognize the caller as CUC subscriber. It seems that AAR can handle such call correctly, but it doesn't respect the ANI transformation rules on the GW. Have you also observed this behaviour? Is there any workaround? regards kobel ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RE : AAR and ANI formatting
Thanks, but I forgot to mention - all the gateways are MGCP. I need to solve this on CUCM. On Sun, Jun 20, 2010 at 9:47 PM, naoufal.kerboute naoufal.kerbo...@cbi.mawrote: Try to do apply a translation-rule on the dial peer routing call to UC using ur internal extension. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access
Yes it does show 5002 is calling. From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] Sent: 18 June 2010 22:24 To: Jones, Brett; 'bkvalent...@gmail.com'; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access From your remote destination, when you make a call into to 617 863 1002, does it show as coming from extension 5002? - Reply message - From: Jones, Brett brett.jo...@redstone.co.uk Date: Fri, Jun 18, 2010 12:08 pm Subject: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access To: 'bkvalent...@gmail.com' 'bkvalent...@gmail.com', ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi, I have found that if I entre 2123942123 as my remote destination number it prompts me for my pin. I have check there are no “Calling Transformation Rules” applied to the gateway. It seems as if the calling number is not being passed to the UCM. Thanks From: Jones, Brett Sent: 18 June 2010 16:36 To: 'bkvalent...@gmail.com'; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access Hi Brain, I have set my mobile number to be 2123942123 and on the debug I can see the same number coming into the gateway but still no joy. Any other ideas? Thanks Brett From: bkvalent...@gmail.com [mailto:bkvalent...@gmail.com] Sent: 18 June 2010 00:33 To: Jones, Brett; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access Yes. On your remote destination, don't put any leading digits. Don't put the 9 trunk access code. Leave it the same as your mobile ani. Use application dial rules to prefix the trunk access code. Mobile connect uses application dial rules to xform the redirecting number. MVA doesn't like it when the ANI of the caller is shorter than the configured remote destination. Brian - Reply message - From: Jones, Brett brett.jo...@redstone.co.uk Date: Thu, Jun 17, 2010 7:22 pm Subject: [OSL | CCIE_Voice] Vol 1 Lab 5C Q12 Mobile Voice Access To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi, I have configure mobile voice access as describe in the video walk through, however when I dial in from my mobile or any other number the IVR asks for me to enter my remote destination number followed by the pound key and not my pin number. When I enter 12345# (which is the pin number) it tell me that it's not a recognised remote destination number. I have set the service parameter to partial match and even changed the matched digits to 7. Anyone see this before? Thanks DISCLAIMER: This correspondence may contain information which is confidential or proprietary or both. Any dissemination, distribution, copying or use of this communication without prior permission of the sender is strictly prohibited. If you are not the intended recipient you may not disclose, copy or use this information. If you have received this message in error, please contact the sender to discuss its return or destruction. The contents, comments and views contained or expressed within this correspondence do not necessarily reflect those of Redstone, its subsidiaries, affiliates, associates or sister companies and are not intended to create legal relations with the recipient. Redstone may monitor email traffic data and also the content of email for the purposes of security and staff training. If you would like to know more about Redstone Converged Solutions, visit us on the web at www.redstoneconverged.co.uk or contact our Head Office on 0845 20 1. Redstone Converged Solutions Limited Registered in England Wales with Company Number: 02027207 Registered Office: Kirtlington Business Centre, Slade Farm, Kirtlington, Kidlington, Oxfordshire, OX5 3JA Click herehttps://www.mailcontrol.com/sr/3BDuQqHkNS3TndxI!oX7UsqBPMJYdVQNfx!p7HNYDaJwLgMpMG8nI9CV!dDo3HBPyNfl1u4k+dgWVaQUGz3XVw== to report this email as spam. DISCLAIMER: This correspondence may contain information which is confidential or proprietary or both. Any dissemination, distribution, copying or use of this communication without prior permission of the sender is strictly prohibited. If you are not the intended recipient you may not disclose, copy or use this information. If youXR DISCLAIMER: This correspondence may contain information which is confidential or proprietary or both. Any dissemination, distribution, copying or use of this communication without prior permission of the sender is strictly prohibited. If you are not the intended recipient you may not disclose, copy or use this information. If you have received this message in error, please contact the sender to discuss its return or destruction. The contents, comments and views contained or expressed within this correspondence do not necessarily reflect those of Redstone, its subsidiaries, affiliates,
Re: [OSL | CCIE_Voice] SRST cor to block International calls
Thank You Nasser Ashar for your expert answers It worked ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] San Jose
Anyone in SJC this week for the OWLE? Shoot me an email if you want to grab a drink tonight. - Sent from my Blackberry ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Gatekeeper CUBE - Pulling My Hair Out!
Hey everyone...I have NO IDEA what is causing my issue and I was hoping for your assistance. I am currently working on Volume 2, Lab 1, Task 4.2 with no success. The goal is to dial 3XXX from HQ or BR1 and route the call from CUCM--GK--CUBE--BR2-RTR. I am getting the *Viazone gateway selection failed for zone VIA* error message. I have included the relevant configuration below. Any help is appreciated! ip domain name proctorlabs.com 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 interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR sccp local FastEthernet0/0.20 sccp ccm 10.10.200.3 identifier 1 priority 1 version 7.0 sccp sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate profile 1 register HQ-XCODER dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP dial-peer voice 3000 voip description GET CALL FROM GATEKEEPER incoming called-number 3... dial-peer voice 3001 voip description SEND CALL BACK TO GATEKEEPER destination-pattern 3...$ session target ras codec g711ulaw gatekeeper zone local UCM proctorlabs.com zone local VIA proctorlabs.com zone local UCME proctorlabs.com outvia VIA zone prefix UCM 1... zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown telephony-service sdspfarm units 5 sdspfarm transcode sessions 4 sdspfarm tag 1 HQ-XCODER max-ephones 1 max-dn 1 ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jun 20 2010 23:20:02 ! HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.2 1720 10.10.110.2 65228 VIA H323-GW H323-ID: BR1-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.110.3 1720 10.10.110.3 57209 UCME H323-GW H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.200.3 1720 10.10.200.3 51074 VIA H323-GW H323-ID: HQ-RTR Voice Capacity Max.= Avail.= Current.= 0 192.168.1.1036728 192.168.1.1033279 UCM VOIP-GW H323-ID: gk-trunk_1 Voice Capacity Max.= Avail.= Current.= 0 192.168.1.1135438 192.168.1.1132790 UCM VOIP-GW H323-ID: gk-trunk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 5 HQ-RTR#debug gatek main 10 HQ-RTR# Jun 21 00:15:32.991: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 21 00:15:32.995: ////GK/gk_rassrv_arq: arqp=0x47D96338,crv=0xB0, answerCall=0 Jun 21 00:15:32.995: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/gk_dns_query: No Name servers Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_get_addrinfo: (1#3001) Matched tech-prefix 1# Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_get_addrinfo: (1#3001) Matched zone prefix 3 and remainder 001 Jun 21 00:15:32.995: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4A68299C Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jun 21 0 HQ-RTR#0:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4B9C9910 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=3 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone and z_outvianamep=VIA Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (VIA) specified. Pick an IP-IP gateway in that viazone. Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: zonep: 0x4A682E7C, tpp: 0x4A62E180, current_endpt: 0 Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: Gateway selection will start at the top of the linked list. use_count=0, current_endpt=0 Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: qelemp=0x0, loop_count=0 Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: Could not find an
Re: [OSL | CCIE_Voice] Gatekeeper CUBE - Pulling My Hair Out!
I have felt your pain ...try this... ! gatekeeper shut no shut ! Hopefully it works. Regards, Mike On Sun, Jun 20, 2010 at 8:21 PM, CCIE VOICE ccievoiced...@gmail.com wrote: Hey everyone...I have NO IDEA what is causing my issue and I was hoping for your assistance. I am currently working on Volume 2, Lab 1, Task 4.2 with no success. The goal is to dial 3XXX from HQ or BR1 and route the call from CUCM--GK--CUBE--BR2-RTR. I am getting the *Viazone gateway selection failed for zone VIA* error message. I have included the relevant configuration below. Any help is appreciated! ip domain name proctorlabs.com 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 interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR sccp local FastEthernet0/0.20 sccp ccm 10.10.200.3 identifier 1 priority 1 version 7.0 sccp sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate profile 1 register HQ-XCODER dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP dial-peer voice 3000 voip description GET CALL FROM GATEKEEPER incoming called-number 3... dial-peer voice 3001 voip description SEND CALL BACK TO GATEKEEPER destination-pattern 3...$ session target ras codec g711ulaw gatekeeper zone local UCM proctorlabs.com zone local VIA proctorlabs.com zone local UCME proctorlabs.com outvia VIA zone prefix UCM 1... zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown telephony-service sdspfarm units 5 sdspfarm transcode sessions 4 sdspfarm tag 1 HQ-XCODER max-ephones 1 max-dn 1 ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jun 20 2010 23:20:02 ! HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.2 1720 10.10.110.2 65228 VIA H323-GW H323-ID: BR1-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.110.3 1720 10.10.110.3 57209 UCME H323-GW H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.200.3 1720 10.10.200.3 51074 VIA H323-GW H323-ID: HQ-RTR Voice Capacity Max.= Avail.= Current.= 0 192.168.1.1036728 192.168.1.1033279 UCM VOIP-GW H323-ID: gk-trunk_1 Voice Capacity Max.= Avail.= Current.= 0 192.168.1.1135438 192.168.1.1132790 UCM VOIP-GW H323-ID: gk-trunk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 5 HQ-RTR#debug gatek main 10 HQ-RTR# Jun 21 00:15:32.991: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 21 00:15:32.995: ////GK/gk_rassrv_arq: arqp=0x47D96338,crv=0xB0, answerCall=0 Jun 21 00:15:32.995: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/gk_dns_query: No Name servers Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_get_addrinfo: (1#3001) Matched tech-prefix 1# Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_get_addrinfo: (1#3001) Matched zone prefix 3 and remainder 001 Jun 21 00:15:32.995: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4A68299C Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jun 21 0 HQ-RTR#0:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4B9C9910 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=3 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone and z_outvianamep=VIA Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (VIA) specified. Pick an IP-IP gateway in that viazone. Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: zonep: 0x4A682E7C, tpp: 0x4A62E180, current_endpt: 0 Jun 21 00:15:32.995:
Re: [OSL | CCIE_Voice] Gatekeeper CUBE - Pulling My Hair Out!
I believe you need a zone prefix for zone VIA. Have you tried to put that on? On Mon, Jun 21, 2010 at 12:21 PM, CCIE VOICE ccievoiced...@gmail.comwrote: Hey everyone...I have NO IDEA what is causing my issue and I was hoping for your assistance. I am currently working on Volume 2, Lab 1, Task 4.2 with no success. The goal is to dial 3XXX from HQ or BR1 and route the call from CUCM--GK--CUBE--BR2-RTR. I am getting the *Viazone gateway selection failed for zone VIA* error message. I have included the relevant configuration below. Any help is appreciated! ip domain name proctorlabs.com 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 interface Loopback0 ip address 10.10.110.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR sccp local FastEthernet0/0.20 sccp ccm 10.10.200.3 identifier 1 priority 1 version 7.0 sccp sccp ccm group 1 bind interface FastEthernet0/0.20 associate ccm 1 priority 1 associate profile 1 register HQ-XCODER dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP dial-peer voice 3000 voip description GET CALL FROM GATEKEEPER incoming called-number 3... dial-peer voice 3001 voip description SEND CALL BACK TO GATEKEEPER destination-pattern 3...$ session target ras codec g711ulaw gatekeeper zone local UCM proctorlabs.com zone local VIA proctorlabs.com zone local UCME proctorlabs.com outvia VIA zone prefix UCM 1... zone prefix UCME 3... zone prefix UCM 5... gw-type-prefix 1#* default-technology no shutdown telephony-service sdspfarm units 5 sdspfarm transcode sessions 4 sdspfarm tag 1 HQ-XCODER max-ephones 1 max-dn 1 ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jun 20 2010 23:20:02 ! HQ-RTR#sh gatek end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.10.110.2 1720 10.10.110.2 65228 VIA H323-GW H323-ID: BR1-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.110.3 1720 10.10.110.3 57209 UCME H323-GW H323-ID: BR2-RTR Voice Capacity Max.= Avail.= Current.= 0 10.10.200.3 1720 10.10.200.3 51074 VIA H323-GW H323-ID: HQ-RTR Voice Capacity Max.= Avail.= Current.= 0 192.168.1.1036728 192.168.1.1033279 UCM VOIP-GW H323-ID: gk-trunk_1 Voice Capacity Max.= Avail.= Current.= 0 192.168.1.1135438 192.168.1.1132790 UCM VOIP-GW H323-ID: gk-trunk_2 Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 5 HQ-RTR#debug gatek main 10 HQ-RTR# Jun 21 00:15:32.991: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 21 00:15:32.995: ////GK/gk_rassrv_arq: arqp=0x47D96338,crv=0xB0, answerCall=0 Jun 21 00:15:32.995: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/gk_dns_query: No Name servers Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_get_addrinfo: (1#3001) Matched tech-prefix 1# Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_get_addrinfo: (1#3001) Matched zone prefix 3 and remainder 001 Jun 21 00:15:32.995: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4A68299C Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: matched zone is UCM, and z_invianamelen=0 Jun 21 0 HQ-RTR#0:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4B9C9910 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: matched zone is UCME, and z_outvianamelen=3 Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone and z_outvianamep=VIA Jun 21 00:15:32.995: //8000830CB000/8000830CB000/GK/rassrv_arq_select_viazone: Received ARQ for a zone (UCME) that has an outviazone (VIA) specified. Pick an IP-IP gateway in that viazone. Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: zonep: 0x4A682E7C, tpp: 0x4A62E180, current_endpt: 0 Jun 21 00:15:32.995: ////GK/gk_gw_select_ipipgw_random: Gateway
[OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None
I have BR1 phones configured with a shared line. During normal operation when phone 1 calls a number (ie 911), phone 2 can see the number of the caller who is connected to phone 1. So privacy is off. This was done by changing the service parameter in callmanager and leaving the phones to default for privacy. The problem is when the phones go into SRST (srst mode auto-provision none) this behavior no longer exists. Its as if privacy is enabled. Neither phone can see who the other phones shared line is connected to. Under telephony-service no privacy is configured. Has anyone ran into this issue before ? Is this a proctorlabs limitation somehow ? I am using 7961s, is that a problem ? I do not know how to fix this without changing to srst mode auto-provision all. Is this a limitation of none ? Your thoughts would be greatly appreciated. Mike ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper CUBE - Pulling My Hair Out!
Hi CCIE Voice (hopefully, I wish you to start using ur number soon ;-) ), Well, I may suggest something different, I have two suggestions: 1st, assign an ip address for your gatekeeper in first zone local command 2nd, use different interface for your VIA zone and your gatekeeper I think the gatekeeper is not being able to sense the topology, and this is why it can detect that it has to use an IPIPGW within VIA zone, but it cannot find it BTW, I have just learnt this debug command from you, it is great, thank you a lot Best regards, Mouhammad _ 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
