[OSL | CCIE_Voice] Testing SRST
I have been looking for quick and easy ways to test SRST, and I've found many different waqys of doing this. With the exception of pulling the WAN interface, they all seem to take a lot of time and effort to accomplish. Anything from creating access-lists to block the traffic to creating new call manager groups and shutting down one of the CallManager services. I have found that a couple of simple static routes to the null 0 interface works very well. ip route 10.10.210.10 255.255.255.255 null 0 ip route 10.10.210.11 255.255.255.255 null 0 no ip route 10.10.210.10 255.255.255.255 null 0 no ip route 10.10.210.11 255.255.255.255 null 0 Add them to a notepad and the no statements as well and you can quickly send your devices into srst mode. Now if you have any VoIP dial-peers that point to other addresses across your WAN you may have to add those as well. What do you guys use? HTH Bill. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CME as SRST help.
It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME as SRST help.
I had a similar issue recently which ended up being a DB replication problem. You could check the phones config file: 10.10.210.11:6970/SEP123456789123.cnf.xml Check subscribers copy of the file Right-click, select view source and search for 'srst' and see what it has there. I could be missing something but it looks like you have enough to get them registered. Marty On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME as SRST help.
Hi Bill, I believe this might be related to a bug with using octo lines in CME SRST. Come out of SRST and reload the router this might resolve the issue. Regards, Ramcharan Arya CCIE # 28926 ( Voice/Routing Switching) On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME as SRST help.
Ramcharan, I'll give that a try. On Thu, Oct 17, 2013 at 10:18 AM, Ramcharan Arya ramcharan.a...@gmail.comwrote: Hi Bill, I believe this might be related to a bug with using octo lines in CME SRST. Come out of SRST and reload the router this might resolve the issue. Regards, Ramcharan Arya CCIE # 28926 ( Voice/Routing Switching) On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME as SRST help.
The reason I was even trying to register my phones in CME SRST is begause I cann't seem to get the CUE MWI to work in SRST and I wanted to see if I was missing something. I get a SIP 481 Call Leg/Transaction Does Not Exist when my CME sends mwi. In researching this issue I found that if the voice port does not have the MWI command on it, you'll get this error. Looking at the ephone-dn's created by SRST I see mwi sip on the DN. Below is the config I'm using. Result of show call-manager-fallback ephone-dn 3 octo-line number 1010 no-reg primary name 6178631010 description 6178631010 preference 0 secondary 9 huntstop huntstop channel 0 call-forward busy 3600 call-forward noan 3600 timeout 12 call-waiting beep cor incoming css-inter mwi sip no cti notify no cti watch Relevant config interface Service-Engine0/0 ip unnumbered Vlan400 service-module ip address 10.10.202.2 255.255.255.0 service-module ip default-gateway 10.10.202.1 ! ccm-manager fallback-mgcp ! dial-peer voice 3600 voip destination-pattern 3[16]00 session protocol sipv2 session target ipv4:10.10.202.2 dtmf-relay sip-notify codec g711ulaw no vad ! ! sip-ua mwi-server ipv4:10.10.202.2 expires 3600 port 5060 transport udp ! call-manager-fallback max-conferences 12 gain -6 transfer-system full-consult ip source-address 10.10.202.1 port 2000 max-ephones 4 max-dn 4 octo-line no-reg voicemail 3600 call-forward busy 3600 call-forward noan 3600 timeout 12 mwi relay moh music-on-hold.au cor incoming css-inter default On Thu, Oct 17, 2013 at 10:29 AM, Bill Hatcher wchatc...@gmail.com wrote: Ramcharan, I'll give that a try. On Thu, Oct 17, 2013 at 10:18 AM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Hi Bill, I believe this might be related to a bug with using octo lines in CME SRST. Come out of SRST and reload the router this might resolve the issue. Regards, Ramcharan Arya CCIE # 28926 ( Voice/Routing Switching) On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher wchatc...@gmail.comwrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.
Seifeddine, I've run that debug, but there is absolutly no output when I'm using CME-SRST. Bill On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili seifeddine.tl...@lvs1.com wrote: Can you send the output of debug ephone register? ** ** Thx ** ** *Kindly*** * * *Seifeddine Tlili* [image: Description: Description: Long View Systems] M.Eng CCIE # 26440 Systems Consultant .. * Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900 [image: Description: Description: Linkedin]http://www.linkedin.com/company/17908 [image: Description: Description: Twitter]http://twitter.com/LongViewSystems [image: Description: Description: Facebook]http://www.facebook.com/longviewsystems [image: Description: Description: Facebook]http://www.youtube.com/longviewsystems www.longviewsystems.com This message and any attached documents are only for the use of the intended recipient(s), are confidential and may contain privileged information. Any unauthorized review, use, retransmission, or other disclosure is strictly prohibited. If you have received this message in error, notify the sender immediately, and delete the original message. *** * ** ** ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher *Sent:* Thursday, October 17, 2013 9:29 AM *To:* ccievoice *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help. ** ** Marty, The weird thing is they work when I use call-manager-fallback. Looking at the cnf file, all seems well. ** ** On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan martinsloa...@gmail.com wrote: I had a similar issue recently which ended up being a DB replication problem. You could check the phones config file: 10.10.210.11:6970/SEP123456789123.cnf.xml Check subscribers copy of the file Right-click, select view source and search for 'srst' and see what it has there. I could be missing something but it looks like you have enough to get them registered. Marty ** ** On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07** ** srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.
Ok, I figured out my issue with the MWI not coming on in SRST more. Need to ass the key word unsolicited to the mwi-server command. Now to get the CME-SRST working. On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wchatc...@gmail.com wrote: Seifeddine, I've run that debug, but there is absolutly no output when I'm using CME-SRST. Bill On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili seifeddine.tl...@lvs1.com wrote: Can you send the output of debug ephone register? ** ** Thx ** ** *Kindly*** * * *Seifeddine Tlili* [image: Description: Description: Long View Systems] M.Eng CCIE # 26440 Systems Consultant .. * Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900 [image: Description: Description: Linkedin]http://www.linkedin.com/company/17908 [image: Description: Description: Twitter]http://twitter.com/LongViewSystems [image: Description: Description: Facebook]http://www.facebook.com/longviewsystems [image: Description: Description: Facebook]http://www.youtube.com/longviewsystems www.longviewsystems.com This message and any attached documents are only for the use of the intended recipient(s), are confidential and may contain privileged information. Any unauthorized review, use, retransmission, or other disclosure is strictly prohibited. If you have received this message in error, notify the sender immediately, and delete the original message. ** ** ** ** ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher *Sent:* Thursday, October 17, 2013 9:29 AM *To:* ccievoice *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help. ** ** Marty, The weird thing is they work when I use call-manager-fallback. Looking at the cnf file, all seems well. ** ** On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan martinsloa...@gmail.com wrote: I had a similar issue recently which ended up being a DB replication problem. You could check the phones config file: 10.10.210.11:6970/SEP123456789123.cnf.xml Check subscribers copy of the file Right-click, select view source and search for 'srst' and see what it has there. I could be missing something but it looks like you have enough to get them registered. Marty ** ** On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07* *** srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.
Ok the re-boot worked and now I can register my phones to CME SRST on a WAN failure, and the CUE MWI works!! I did have to go in and edit the call-foward parameters on the ephone-dn with voicemail, and add my cor-list info to all the lines. On Thu, Oct 17, 2013 at 10:53 AM, Bill Hatcher wchatc...@gmail.com wrote: Ok, I figured out my issue with the MWI not coming on in SRST more. Need to ass the key word unsolicited to the mwi-server command. Now to get the CME-SRST working. On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wchatc...@gmail.comwrote: Seifeddine, I've run that debug, but there is absolutly no output when I'm using CME-SRST. Bill On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili seifeddine.tl...@lvs1.com wrote: Can you send the output of debug ephone register? ** ** Thx ** ** *Kindly*** * * *Seifeddine Tlili* [image: Description: Description: Long View Systems] M.Eng CCIE # 26440 Systems Consultant .. * Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900 [image: Description: Description: Linkedin]http://www.linkedin.com/company/17908 [image: Description: Description: Twitter]http://twitter.com/LongViewSystems [image: Description: Description: Facebook]http://www.facebook.com/longviewsystems [image: Description: Description: Facebook]http://www.youtube.com/longviewsystems www.longviewsystems.com This message and any attached documents are only for the use of the intended recipient(s), are confidential and may contain privileged information. Any unauthorized review, use, retransmission, or other disclosure is strictly prohibited. If you have received this message in error, notify the sender immediately, and delete the original message. * *** ** ** ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher *Sent:* Thursday, October 17, 2013 9:29 AM *To:* ccievoice *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help. ** ** Marty, The weird thing is they work when I use call-manager-fallback. Looking at the cnf file, all seems well. ** ** On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan martinsloa...@gmail.com wrote: I had a similar issue recently which ended up being a DB replication problem. You could check the phones config file: 10.10.210.11:6970/SEP123456789123.cnf.xml Check subscribers copy of the file Right-click, select view source and search for 'srst' and see what it has there. I could be missing something but it looks like you have enough to get them registered. Marty ** ** On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.
Bill, Here's a great reference for CME MWI: http://ciscovoiceguru.com/518/cue-mwi-notification-methods/ I've used this a lot through my studies. @Ramcharan - good call. I think the symptom with that bug is that the phones will register but display no DN's and if you issue 'show ephone reg' it will show the DN's as 'invalid' or something like that. I have also hit the other bug with SC MGCP after coming out of SRST the dial-peers are still chosen on inbound calls from the PSTN. I believe the fix for that is to globally set 'voice hunt 2' and under the ephone-dn's assign a higher preference (I use 9). On Thu, Oct 17, 2013 at 11:53 AM, Bill Hatcher wchatc...@gmail.com wrote: Ok, I figured out my issue with the MWI not coming on in SRST more. Need to ass the key word unsolicited to the mwi-server command. Now to get the CME-SRST working. On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wchatc...@gmail.comwrote: Seifeddine, I've run that debug, but there is absolutly no output when I'm using CME-SRST. Bill On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili seifeddine.tl...@lvs1.com wrote: Can you send the output of debug ephone register? ** ** Thx ** ** *Kindly*** * * *Seifeddine Tlili* [image: Description: Description: Long View Systems] M.Eng CCIE # 26440 Systems Consultant .. * Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900 [image: Description: Description: Linkedin]http://www.linkedin.com/company/17908 [image: Description: Description: Twitter]http://twitter.com/LongViewSystems [image: Description: Description: Facebook]http://www.facebook.com/longviewsystems [image: Description: Description: Facebook]http://www.youtube.com/longviewsystems www.longviewsystems.com This message and any attached documents are only for the use of the intended recipient(s), are confidential and may contain privileged information. Any unauthorized review, use, retransmission, or other disclosure is strictly prohibited. If you have received this message in error, notify the sender immediately, and delete the original message. * *** ** ** ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher *Sent:* Thursday, October 17, 2013 9:29 AM *To:* ccievoice *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help. ** ** Marty, The weird thing is they work when I use call-manager-fallback. Looking at the cnf file, all seems well. ** ** On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan martinsloa...@gmail.com wrote: I had a similar issue recently which ended up being a DB replication problem. You could check the phones config file: 10.10.210.11:6970/SEP123456789123.cnf.xml Check subscribers copy of the file Right-click, select view source and search for 'srst' and see what it has there. I could be missing something but it looks like you have enough to get them registered. Marty ** ** On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME as SRST help.
HiFor telephony-serviceI used:srst mode auto-provision allsrst dn line-mode dualAnd for call-manager-fallback I used:max-ephone 4max-dn 8 dual-lineAm I wrong using these configurations?I avoid octo-lines for the bug mentioned.best regards!AlexOn Oct 17, 2013, at 10:18 AM, Ramcharan Arya ramcharan.a...@gmail.com wrote:Hi Bill,I believe this might be related to a bug with using octo lines in CME SRST.Come out of SRST and reload the router this might resolve the issue.Regards,Ramcharan Arya CCIE # 28926 ( Voice/Routing Switching)On Thu, Oct 17, 2013 at 9:34 AM, Bill Hatcher wchatc...@gmail.com wrote:It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager.telephony-servicesrst mode auto-provision allsrst ephone description auto provisioned ephone : Oct 17 2013 14:47:07srst dn line-mode octomax-ephones 4max-dn 4ip source-address 10.10.202.1 port 2000max-conferences 12 gain -6transfer-system full-consultcreate cnf-files version-stamp 7960 Oct 17 2013 16:07:18___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Base configs and Dial Plan mapping
My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wchatc...@gmail.com wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
I agree to setting the service parameters to default first. I was planning on doing that myself. As to changing the DSCP values, it all depends on what they ask for in the QoS section of the test is all. On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com probert...@gmail.comwrote: Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wchatc...@gmail.com wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
Hi, On service parameters, you may also want to check Vik's article http://blog.ipexpert.com/2010/10/13/common-ucm-service-parameters-to-change/ . On the comment section, Trinifox also mentioned Please add: Intraregion Audio Codec Default to G729 to avoid CSCsl74701 Bug. In my checklist, I also tweak Conference section esp. Drop Ad Hoc Conference. Regards, --Somphol. On Fri, Oct 18, 2013 at 7:14 AM, Bill Hatcher wchatc...@gmail.com wrote: I agree to setting the service parameters to default first. I was planning on doing that myself. As to changing the DSCP values, it all depends on what they ask for in the QoS section of the test is all. On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.com probert...@gmail.com wrote: Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wchatc...@gmail.comwrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension
Sorry for revisiting this old thread. The Calling Party Transformation at the Device Pool level would come in handy for this particular need. In the document starting 7.1.2, this is stated explicitly, http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fscallpn.html#wp1325305 . *Cisco Unity/Cisco Unity Connection* Because no calling party transformation options exist in Cisco Unified Communications Manager Administration for voice-messaging ports, make sure that you configure the calling party number transformations in the device pool that is associated with the voice-messaging ports. ... Table 7-8 Configuring the Calling Party Transformation CSS to Localize the Calling Party Number Also mentioned Use Device Pool Calling Party Transformation CSS as a method to Localize the Calling Party Number. ... ... The same document for 7.0.1 contained the table 7-8, but somehow doesn't have that explicit section on Cisco Unity/Cisco Unity Connection's calling party localization. (REF: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fscallpn.html#wp1276877) So, I am not so sure whether this is possible in CUCM 7.0.1/CUC 7.0.1. I don't have lab access to test this now, but would appreciate if anyone can help testing this. Note: I recall seeing some sort of Technotes outlining the strategy to perform Calling Party transformation for Call Manager 4.x or something that doesn't rely on Gateway's Calling Party Transformation.I can't locate it now, but if anyone could point me to the URL that would be great. Regards, --Somphol. On Sat, Mar 23, 2013 at 12:34 AM, Leslie Meade leslie.me...@lvs1.comwrote: Easy way of doing this is to copy the hunt pilot and give it another number.. set user caller ID and mask it to Then in the call-manager-fallback change the voicemail to the new hunt pilot and your done ** ** ** ** *Leslie Meade* .. * Mobile:778.228.4339* | *Main:* *604.676.5239* *Email:* leslie.me...@lvs1.com ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Chadi H Hassoune (chassoun) *Sent:* Thursday, March 21, 2013 7:10 PM *To:* Pixar Perfect; Mark Thrash (marthras); Steve Keller *Cc:* CCIE Voice OSL *Subject:* Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension ** ** Calling Party Xform and assign it to the CUC Device Pool works fine for me. ** ** HTH ** ** *From: *Pixar Perfect pixarperf...@live.com *Date: *Wednesday, March 20, 2013 11:43 PM *To: *Mark Thrash (marthras) marth...@cisco.com, Steve Keller skeller...@gmail.com *Cc: *CCIE Voice OSL ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension ** ** the requirement is always for SiteB calling into SiteA voicemail by hitting Messages button. SiteA is always MGCP gateway. Calling Party Xform isnt any use on MGCP gateway for incoming calls. ** ** here is another way of doing it ... ** ** Voicemail Pilot for CUC is 2200 ** ** call-manager-fallback voicemail 2777 --- siteB specific ** ** translation-pattern on CUCM to convert 2777 into 2200 and mask calling number . The CSS of the translation pattern should have access to 2200. ** ** ** ** ** ** there is no definitive answer as to which solution is graded positively. there is a reason why many leading CCIE instructors say this is not a test of best practices but a test of how like able is your solution to the script. .. :) ** ** ** ** ** ** ** ** -- From: marth...@cisco.com To: skeller...@gmail.com Date: Thu, 21 Mar 2013 03:59:48 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST to voicemail without Alternate Extension What about a calling party transform mask on the incoming gateway? Sent from my iPhone On Mar 20, 2013, at 10:43 PM, Steve Keller skeller...@gmail.com wrote: Thanks Bill, I like this option pretty well as it seems to limit treatment of calls this way to CUC when site B is in SRST mode only. I will try to lab this up tomorrow morning. Question for you, will this only solve my issue of pressing the VM button to access my mailbox to retrieve a message. Meaning when PSTN calls in to site B phone and then gets forward(redirected) to voicemail, I use a dial-peer that provides RDNIS capabilites to route the caller to the correct mailbox and not the opening greeting. So with this would i still want to use the following to get the caller into my mailbox? dial-peer voice 2600 pots destination-pattern 2600 port 0/0/0:23 no digit-strip prefix 202555 ( assuming no LD code at this site ) this is the way i get callers into my
Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping
I've found the QoS questions are very specific to test a certain area of knowledge. They are not looking for what we would consider a best practice system wide. I think we could skip setting the DSCP values in CUCM. If you think the question calls for it you can have your class-map match both AF31 and CS3 for signaling. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Hatcher Sent: Thursday, October 17, 2013 3:14 PM To: probert...@gmail.com Cc: ccievoice Subject: Re: [OSL | CCIE_Voice] Base configs and Dial Plan mapping I agree to setting the service parameters to default first. I was planning on doing that myself. As to changing the DSCP values, it all depends on what they ask for in the QoS section of the test is all. On Thu, Oct 17, 2013 at 2:24 PM, probert...@gmail.commailto:probert...@gmail.com probert...@gmail.commailto:probert...@gmail.com wrote: Hi, I think my strategy will be to set all Service Parameters to default before making changes. This way I can avoid and undesirable presets. Let me know your thoughts on this. Also why are you setting DSCP for Phone Configuration and DSCP for Cisco CallManager to Device Interface to AF31? Default CS3 should be good, let me know if I'm wrong on this? Rob On Thu, Oct 17, 2013 at 1:39 PM, Bill Hatcher wchatc...@gmail.commailto:wchatc...@gmail.com wrote: My test is just a couple of weeks away, and I've been reading different blogs on how to maximize your time. The one thing I'm really struggling with is mapping out my dial-plan during my read through of the lab. I would love to hear what others are doing. I have also been building base router configs for h323, gatekeeper, mgcp, srst,sip, etc so that I can practice quickly configuring those on the routers. One of the things I haven't really been keeping track of are some of the service parameters that I should adjust out of habit. Here are a few that I can think of off the top of my head that I plan on tweaking at the start of the exam. Please feel free to add to them. Enterprise Parameters DSCP for Phone Configuration - Set to AF31 DSCP for Cisco CallManager to Device Interface - Set to AF31 Change the Phone URL's to IP's Organization Top Level Domain Cluster Fully Qualified Domain Name Service Parameters - CallManager T302 Time - Know where it is if you need ot change interdigit timeout Call Classification - Offnet Builtin Bridge Enabled - True Device Name of GK-controlled Trunk That Will Use Port 1720 - Set as needed. Transfer On-hook Enabled - True (Also a great thing to do in production when migrating users from other phone systems) Block offnet to offnet transfers - Know where it's at. Auto Call Pickup Enabled - True Call Back Enabled Flag - True (Verify) Single Button Barge/CBarge Policy - Set to Barge unless otherwise directed. Stop routing on Unallocated Number Flag - False - H323 redundancy Preferred G.711 Millisecond Packet Size - 20 (Verify) Preferred G.729 Millisecond Packet Size - 20 (Verify) G722 Codec Enabled - Disabled (Unless otherwise directed) Intraregion Audio Codec Default - G711/G722 (Verify) Interregion Audio Codec Default - G729 (Verify) Automated Alternate Routing Enabled - True (This one gets me every time on AAR so I turn it on by default now) Enable Mobile Voice Access - Set as required Mobile Voice Access Number - Set as required System Remote Access Blocked Numbers - Set as required Service Parameters -Cisco IP Voice Media Streaming App Supported MOH Codecs - G711 mulaw and G729 Annex A HTH Bill Hatcher ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com Important Notice: This email message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of Core BTS. Core BTS specifically disclaims liability for any damage caused by any virus transmitted by this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Testing SRST
An easier way to test for THEO and such is to just shut down the voice-port (not the controller or serial). Quick and easy and perhaps not as easy to overlook when troubleshooting. I've left my null routes in a couple of times without realizing it. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bill Hatcher Sent: Thursday, October 17, 2013 11:53 AM To: ccievoice Subject: Re: [OSL | CCIE_Voice] Testing SRST If you are talking about testing redundancy I'll do the same thing on the gateway I want to simulate as being down. For example when doing TEHO where if the remote gateway is down we want to fail to the local gateway, I'll go to the remote gateway and put in the static routes. On Thu, Oct 17, 2013 at 11:46 AM, Alex Mendoza aa.mend...@icloud.commailto:aa.mend...@icloud.com wrote: Hi Bill I use your way to test SRST, I'm wondering what are you using for test Route List when they have 2 route groups. best regards. Alex On Oct 17, 2013, at 09:23 AM, Bill Hatcher wchatc...@gmail.commailto:wchatc...@gmail.com wrote: I have been looking for quick and easy ways to test SRST, and I've found many different waqys of doing this. With the exception of pulling the WAN interface, they all seem to take a lot of time and effort to accomplish. Anything from creating access-lists to block the traffic to creating new call manager groups and shutting down one of the CallManager services. I have found that a couple of simple static routes to the null 0 interface works very well. ip route 10.10.210.10 255.255.255.255 null 0 ip route 10.10.210.11 255.255.255.255 null 0 no ip route 10.10.210.10 255.255.255.255 null 0 no ip route 10.10.210.11 255.255.255.255 null 0 Add them to a notepad and the no statements as well and you can quickly send your devices into srst mode. Now if you have any VoIP dial-peers that point to other addresses across your WAN you may have to add those as well. What do you guys use? HTH Bill. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com Important Notice: This email message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the named addressee, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. Please note that any views or opinions presented in this email are solely those of the author and do not necessarily represent those of Core BTS. Core BTS specifically disclaims liability for any damage caused by any virus transmitted by this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
