Re: [cisco-voip] ATA 186 question
if money is a big problem. but you could get some ATA187s for about $20 apiece or ATA190s On Thursday, January 27, 2022, 02:49:53 PM MST, Lisa Notarianni wrote: Thank you all for your quick responses. We will go beg for funding and replace the ATAs. We cannot set speed to 10 since it is not an option according to our network engineer. Lisa From: Lelio Fulgenzi Sent: Thursday, January 27, 2022 4:26 PM To: Lisa Notarianni Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] ATA 186 question You might want to double check that. From this, 17.3, it shows 10 is a valid option. https://www.cisco.com/c/en/us/td/docs/switches/lan/catalyst9300/software/release/17-3/configuration_guide/int_hw/b_173_int_and_hw_9300_cg/configuring_interface_characteristics.html#task_uw1_3wc_g1b Sent from my iPhone On Jan 27, 2022, at 3:45 PM, Lisa Notarianni wrote: CAUTION: This email originated from outside of the University of Guelph. Do not click links or open attachments unless you recognize the sender and know the content is safe. If in doubt, forward suspicious emails toith...@uoguelph.ca We just began Cisco switch upgrade to 9300s and ran into an unexpected issue. We have many ATA 186s out there (about 100) and did not budget for replacement as we were gratefully approved for the massive overhaul of the switches. This ended up being an unforeseen glitch. We cannot get the ATA 186s to work on this switch with this IOS version: 1 65 C9300-48UXM 17.03.04 CAT9K_IOSXE INSTALL However, the 186s are working on this switch with this IOS version: 1 64 C9300-48P 16.9.5 CAT9K_IOSXE INSTALL I know it is not an apples to apples comparison but I was wondering if anyone had a workaround for the newer model/IOS. We believe the problem is that the ATA186s are hard coded at 10Mbps/half duplex. Even if we change the switch port parameter from 0x to 0x00ff the ATA will not register. The switch port is also configured at duplex auto but the slowest we can make the port is 100Mbps. Is this not possible? I am hoping you Cisco geniuses have a solution😊 Lisa Notarianni University of Scranton Telecommunications Engineer Infrastructure Services 800 Linden St. Scranton PA 18510 570.941.4325 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Centralized Cisco Cube Deployment
Centralized saves money and for me time.I don't do SRST at remote sites because if the network is down. most staff can't work or answer questions. and for 911 when the network is down, most people have cell phones.All the conferencing and resources are at the headend so that saves BW.and if you need more ports, you go into the vendor's portal and increase as needed.For 911, I use CER and use the network its on to get the correct ERLIN and save money by not having every number setup as 911. and if a phone is moved to another loacation, don't have to worry about updating 911. On Monday, June 7, 2021, 09:13:17 PM PDT, Kent Roberts wrote: Yes it would. Brings up other concerns…. 911 for example…. Local pots line… multiple WAN circuits…. To resolve that…. Can the site survive without PSTN if its single hosted? what happens if you loose the DC Wan connect? What do your sites do…. Do you have resources to handle those calls during the outage. (Vmail,, or a ton of ports in IVR) Do you have enough bandwidth from the DC to the branches? What codec will you support? G729/G711. (Remember not customers support G729, and not all carriers will “normalize” their network). Example ATT will route calls to other att customers without leaving their cloud, if they can’t do G729 and that is how you start the call, it will fail. (ATT is changing this FYI) QOS policy? Business support.. IE in we can save money, but a failure might mean loss of calls…. Are they ok with this. DSP resources. Do you have enough to support Conferencing…… etc Firewalls, Does your firewall team block access branch to branch… do you need to setup relays? VPN….. Jabber access… Concerns with voice parts across the branches.. Hope that helps. It is very much do-able, and can save lots of money. Just need to think it through and not rush…. Its really no different then say a branch being the data center for all the other sites… > On Jun 7, 2021, at 8:23 PM, LTGJAMAICA wrote: > > I want to know if a centralized cisco cube deployment would allows us to do > away with our existing voice gateways. > > I am trying to reduce costs by removing 6 cisco voice gateways and isdn pri > circuits located at 6 branch offices. I want to replace these individual > gateways with two cisco cubes/vcubes located at our hub location connected to > an ITSP via a Sip Trunk. Each branch is connected to an MPLS wan. > > > ___ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Outbound SIP connection failing in CUBE due to some timer... maybe.
Was this working before?I had problems before on a 4331 CUBE. I had to redo the password on the CUBE to Lumen.it was the same password that was already there. and a reboot didn't fix it. On Friday, April 16, 2021, 08:32:22 AM PDT, Sreekanth Narayanan (sreenara) via cisco-voip wrote: Nick, What's the disconnect cause from the CUBE? 102? Do you have logs for this call? Would be clear which timers are expiring, causing the problem.debug ccsip messagedebug ccsip errordebug ccsip info -sreekanthFrom: cisco-voip on behalf of Nick Barnett Sent: Friday, April 16, 2021 6:53 PM To: cisco-voip Subject: [cisco-voip] Outbound SIP connection failing in CUBE due to some timer... maybe. Yes, very vague subject. Sorry about that. Some calls to certain wireless carriers on our ITSP connections have started failing. Win10 Jabber client (off of 12.5.su3) -> CUBE -> ITSP The call goes out Lumen, the 401 auth and challenge response are fine, the INVITE is then sent with SDP. We get a TRYING response which we immediately ACK. Up until this point, the entire call flow is NORMAL. If we don't receive a 18X response within 7 Seconds, the, the CUBE sends a cancel. Yes, the CUBE. It appears that the far end is taking too long to send the 18X message. we involved our carrier and they can see the 18X come back a split second later (sometimes), but our side has already closed the connection. I looked at all of the sip-ua timers and retry settings. nothing adds up to 7 seconds. Most timers are set to 500 msec. I'm not sure where to look? It's not on the sip profile. i tried bumping up the connect, update, info and trying timers (one at a time), but it didn't make any difference. Maybe I was supposed to do something to make sip-ua changes "kick in" like bounce the sip service which I didn't do... not sure on that part. Please tell me there is something simple I'm missing. Pointers? Thanks, Nick p.s. : some possibly relevant config and the timers and retries from my sip-ua retry invite 2 retry response 6 retry bye 10 retry cancel 10 retry prack 10 retry update 6 retry rel1xx 6 retry notify 10 retry refer 10 retry info 6 retry register 6 retry subscribe 6 retry keepalive 6 retry options 6 timers trying 500 timers expires 18 timers connect 500 timers connection aging 5 timers disconnect 500 timers prack 500 timers update 500 timers rel1xx 500 timers notify 750 timers refer 500 timers hold 2880 timers info 500 timers register 500 timers buffer-invite 0 timers keepalive down 30 timers keepalive active 120 timers dns registrar-cache 3600 timers options 500 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 5 digit and 10 digit dialing
Thanks Anthony, This will try it in the lab.Keep trying to make the system fool proof, but the fools keep getting on the system. Carlo From: Anthony Holloway To: Brian Meade Cc: Carlo Calabrese ; Cisco VoIP Group Sent: Thursday, April 12, 2018 12:48 PM Subject: Re: [cisco-voip] 5 digit and 10 digit dialing Re-reading my email, I realize I have at least one typo. I used 6120 as an example extension, and I should have used a fourth digit between 2-9 to make the example work. So, take 6126 as my new example extension. On Thu, Apr 12, 2018 at 12:43 PM Anthony Holloway wrote: It's surprising to me, how many people I hear these days, asking to remove the PSTN access code (typically a 9 in the US), but then still want intra- and inter-site abbreviated dialing habits to be supported. The usual defense is: "I don't have to dial a 9 on my cell phone." Though, these same people admit that they cannot 4 digit dial on their cell phones either. I have one customer who was talked into a design by someone else, and just the opposite of OP, they have to dial a "1" for all calls, local and national. And so, none of the intra- or inter-site abbreviated dialing habits start with a 1. So, in a way, they're still dialing a PSTN access code, it just happens to be "1" and not "9". Oh, and they cannot have a 0 operator extension, because of international dialing habits starting with 011. That's not great, but I get it. People ask for things from an ignorant place, and it's our jobs as experts to inform and lead design discussions. I think everyone should avoid inter-digit timeouts (aka post dial delay, aka T.302, aka "Why isn't my call working?") in their design, and Cisco has now given us the glorious check box for translation patterns: "Do Not Wait For Interdigit Timeout On Subsequent Hops" to help do just that. Now, in variable length numbering plans within E164, it's not un-avoidable, but in the US and for intra- and inter-site dialing, it is avoidable. Actually, it's not like what I think is the best design solution just because I think it is, but rather, it's actually a published design practice from Cisco in their Preferred Architecture for Enterprise Collaboration 11.6: "Starting the design process with an overview of all dialing habits makes sure that overlaps between any two dialing habits leading to inter-digit timeouts are detected and can be resolved before starting the dial plan deployment. Avoiding overlaps with any other (typically on-net) dialing habit is the key reason for using a PSTN access code (typically 9 in the US, as shown above)." Source: https://www.cisco.com/c/en/us/td/docs/solutions/CVD/Collaboration/enterprise/11x/116/collbcvd/control.html To answer the OP question more directly: well, it depends. The timer expiring has nothing to do with the pattern that was matched. For example, if you had the following two Translation patterns as potential matches: 1XXX Urgent135! Non-Urgent And you dialed 1350, you could not dial anymore digits after the 0, even though we've used the ! to indicate more digits could follow. Furthermore, the pattern with UP on it, is not matched, because 135! has less potential matches (aka is the closest match). So, with that knowledge, and with what Brian said, you could do this: Non-Urgent - Used for intra-site abbreviated dialing - Uses a CSS that can only reach internal extensions[2-9]XX[2-9]XX Urgent - Used for US Local and National PSTN dialing - Uses CSS inheritence to match next hop RP for PSTN routing So if the user dialed 6120, then waits for the timer to expire, CUCM selects the non-urgent pattern. Then, if the user dials 6125551212, CUCM selects the urgent pattern for PSTN and routes the call. In that way, your internal dialing uses one CSS, while your PSTN calls use another, and your internal dialing (which should be thought of as fast) will now have a post dial delay (default 15 seconds). Though, I must go back to the beginning of what I was saying, and say that I think you should review the documentation on dial plan design. There are even two great sessions at Cisco Live every year on the topic, which you can watch for free right now: Enterprise Dial Plan Fundamentals - BRKUCC-2008 Advanced Dial Plan Design for Unified Communications Networks - BRKUCC-3000 On Thu, Apr 12, 2018 at 10:00 AM Brian Meade wrote: Can't you just add those partitions to the existing CSS and make sure Urgent Priority isn't checked on the 5-digit extensions? On Thu, Apr 12, 2018 at 10:52 AM, Carlo Calabrese via cisco-voip wrote: Users are doing 10 digit dialing so any calls local or long distance are just 10 digits. they also want to do 5 digit dialing to the cube next door. I have * *but is there a way to look for a dial pattern in a different partition aft
[cisco-voip] 5 digit and 10 digit dialing
Users are doing 10 digit dialing so any calls local or long distance are just 10 digits. they also want to do 5 digit dialing to the cube next door. I have * *but is there a way to look for a dial pattern in a different partition after the inter-digit time out is reached. So they user would start dialing and if they only dial 5 digits and after the inter-digit timeout is reached, it would look at another CSS. Thanks. Carlo ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] route calls to VM after sending to Route list
I have a problem where I send a call to a route pattern, route list. if I get a busy signal, I want to send the call to voice mail or have UCCX pick it up.I have tried to use the same pattern with different partitions but it never goes to the second partition. I did this years ago but can't remember how and that was 8.5 Thanks Carlo ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCM UCCX email to Comcast
no this is to email reports for CUIC and RTMT. So it sounds like I just need to build an SMTP server. Guess what I get to learn this weekend. Thanks guys From: Anthony Holloway To: James Buchanan Cc: Carlo Calabrese ; "cisco-voip@puck.nether.net" Sent: Friday, February 16, 2018 12:41 PM Subject: Re: [cisco-voip] CUCM UCCX email to Comcast I saw UCCX Email in the subject and was thinking Agent Email. Maybe they just meant notifications via SMTP. Not sure. On Fri, Feb 16, 2018 at 2:21 PM James Buchanan wrote: Hello, Before Unity Connection supported Office365 for Single Inbox, I had to create my own SMTP server to relay emails over to Office 365. So, I created a Windows VM and added the Windows SMTP server to it, allowing only CUC to relay from it. It worked fine. It doesn't have to be a Windows SMTP server of course. Thanks, James On Fri, Feb 16, 2018 at 7:54 PM, Anthony Holloway wrote: So you basically need pubic DNS, a public IP Address, and Firewall rules to allow the external to external communication. I have not done this before either, just thinking out loud. On Fri, Feb 16, 2018 at 1:34 PM Carlo Calabrese via cisco-voip wrote: Hi all, I have a customer that we are installing a CUCM, CUC and UCCX. the problem is that what it total separate from their data network so I have a connection to Comcast. I need to setup the email for outbound. Has anybody done this.Or is this possible Thanks Carlo. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] CUCM UCCX email to Comcast
Hi all, I have a customer that we are installing a CUCM, CUC and UCCX. the problem is that what it total separate from their data network so I have a connection to Comcast. I need to setup the email for outbound. Has anybody done this.Or is this possible Thanks Carlo. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Phone equipment refresh?
We face the same problem, so we have been replacing a site at a time, so people
don't get phone envy. I like the 7841 for staff and 8851 or 8861 for
receptionist areas. If you have someone who is using wireless headsets, then
look at the 8851. It has Bluetooth built in so you don't need to spend an extra
$200 dollars on a headset unit or get the overprices cables.Also, on either one
of these phones, make sure to update the firmware to a 12.X code. there is a
problem with the 11.7 code where you get crackling between Cisco phones. (that
was a fun one)
Good luck, we all feel your pain.
From: Lisa Notarianni
To: "cisco-voip@puck.nether.net"
Sent: Monday, January 29, 2018 11:34 AM
Subject: [cisco-voip] Phone equipment refresh?
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{}#yiv8886688008 We are currently at CUCM 11.5. We will soon be able to
provide Gig to desktop but our current phones will prevent us from doing so.
We need to devise a plan and wondered what others do for their equipment.
The majority of phones we have are 7900 series. They are working well for the
most part so we replace as needed. We are concerned that the 7900 series may be
eliminated from a future CUCM version and prevent us from upgrading. Does
anyone have a plan for a refresh on Cisco VoIP phones? Do you lease or did
you purchase? We appreciate any information you can offer. Thank you.
Lisa Notarianni Telecommunications Engineer The University of Scranton (570)
941-4325 ___
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Re: [cisco-voip] UCCX adding ring before hold music
Before you accept the call, Check to see if all agents are busy, if so, add a delay of 1 or 2 seconds before you accept the call. From: ccie collan To: cisco-voip@puck.nether.net Sent: Friday, December 1, 2017 8:29 AM Subject: [cisco-voip] UCCX adding ring before hold music Hello , when all of the agents are busy on the calls, when calls come into UCCX customers here music on hold , can you please advise how i can rings before the call enters the hold cycle ? Thank you . ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX Place Call, Get Digit String and Play Prompt
I have this working and it waits till an agent gets the call. it take two scripts to run it I can send you the scripts if you want I have it running on 10.6 I do have a lot of it documented in the script.Let me know. Carlo From: Ray Maslanka To: cisco-voip@puck.nether.net Sent: Wednesday, August 9, 2017 7:43 AM Subject: [cisco-voip] UCCX Place Call, Get Digit String and Play Prompt Gentlemen, Much like the BaseLineAdvQueueing.aef script in the script repository, I have scripts in production that allow callers the ability to leave a message if they choose and receive a call back. UCCX records the callers message, terminates the call, calls a trigger, waits for an agent to answer and prompts them to press a digit to confirm they want to hear the recorded message. The agent is then free to do what they want with that information. It seems what really happens after the message is recorded and the call is made to the trigger to deliver it to an agent is that the script starts playing the prompt "Press a digit to hear a message", regardless of whether an agent has actually answered. That prompt plays and waits a given amount of seconds for the agent's digit input, and then loops, courtesy of the timeout function related to the initial digit timer. If no agents are available, the script will continue to play the prompt and wait for digits while listening to hold music, delay prompts or whatever may be presented while waiting for an agent. There is no real harm done here though. The issue is when an agent does answer, depending on when they answer during the looping process, they may hear "Press a digit to hear a message" or "to hear a message" or silence as long as the initial timeout value in the Get Digit String step before "Press a digit to hear a message". In higher volume environments, that timer and the possible related silence after answering may be unacceptable. Three seconds of nothing may be enough to trigger an agent to assume it is an abandoned call and hang up. I am hoping someone has a technique to have UCCX only start playing the "Press a digit to hear a message" when the agent actually answers the call that UCCX made into their queue. If what I am experiencing is expected, confirming that would be great too and I'll try to find an acceptable timer or different recordings, etc. If you believe what I am experiencing is not correct behavior, any suggestions on what is wrong with that sample would be appreciated. Running into this on fully patched UCCX 11, CUCM 11 and 8800 series endpoints. Thanks in advance for any feedback. Ray Maslanka ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] FXO port to Viking C-1000B door entry controller
I take it you press 6 or something to allow the gate to open? can you send a copy of the MGCP statements on the gateway? looking to see what the dtmf-relay is set to. mgcp dtmf-relay voip codec all mode nse On Tuesday, March 28, 2017 6:46 AM, Brian Meade wrote: We'd need some debugs to see what is happening like "debug vpm signal" to see what tones are sent on the line. On Tue, Mar 28, 2017 at 9:35 AM, Kevin via cisco-voip wrote: Hi, I was wondering if a solution was found. I have a similar issue. We have a Viking gate opener that was connected to a Nortel Option 11c analog port. We have migrated to a CUCM. The local analog port is now an FXS port on a Cisco 4331 router using MGCP. The call box at the gate works in that I can call from the call box to a Cisco IP phone and a two way connection works. But when the code is entered on the IP phone to open the gate, the call drops and the gate does not open. Thanks in advance. Kevin Sent from my iPhone __ _ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/ mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Anti-Tromboning Tie Lines to PBX
I found the link. it has what needs to be changed in the PBX and what is needed on the gateway and in the gateway configs on CUCM. http://www.cisco.com/c/en/us/solutions/collateral/enterprise/interoperability-portal/712433.pdf Carlo On Thursday, March 9, 2017 7:43 AM, Carlo Calabrese via cisco-voip wrote: QSIG with MGCP. I use this to connect to a few Nortel PBX's. The main reason was to get rid of Meridian Mail and put it on Unity. Started this bad in the 4.2 days. When the call transfers out it will release the channels and the PRI is no longer in the call.Easy to set up. just a couple of configs on the gateway and a few options changes on the PBX. Cisco has a document on it somewhere. I am trying to find it on my PC, but that was 10 years ago. On Tuesday, March 7, 2017 3:25 PM, Brian Meade wrote: Also known as 2 B-channel Transfer or TBCT (should help your searching). On Tue, Mar 7, 2017 at 5:54 PM, Anthony Holloway wrote: I'm not sure how often this gets configured out in the wild, but I do know that there are few different flavors of this feature, depending on your scenario. E.g., CUBE Media Anti-Tromboning The one I'm interested in would involve a tie line to a legacy PBX, and I'm not married to the IP technology: SIP, H323, MGCP, whichever will get the job done. CUCM ---IP---> VGW ---T1/PRI---> PBX The call scenario I'd like to see if we can avoid is when a call is established from CUCM to the other PBX, and then a phone on the far side transfers the call to an IP Phone on the CUCM side, I'd like the PRI usage to drop to 0 trunks in use, instead of nailing up 2 trunks for the duration of that call. I'm looking for your experience and feedback in configuring anti-tromboning in this scenario. One of the hardest parts in researching this, is knowing what terms to search for. I think I've narrowed down the terminology to QSIG Path Replacement, as described in the CUCM SRND. Though, without real-world working experience, it's hard to know if I'm right or not. So, I know there's some good legacy telco knowledge out on this list. What do you know? Thanks! __ _ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/ mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Anti-Tromboning Tie Lines to PBX
QSIG with MGCP. I use this to connect to a few Nortel PBX's. The main reason was to get rid of Meridian Mail and put it on Unity. Started this bad in the 4.2 days. When the call transfers out it will release the channels and the PRI is no longer in the call.Easy to set up. just a couple of configs on the gateway and a few options changes on the PBX. Cisco has a document on it somewhere. I am trying to find it on my PC, but that was 10 years ago. On Tuesday, March 7, 2017 3:25 PM, Brian Meade wrote: Also known as 2 B-channel Transfer or TBCT (should help your searching). On Tue, Mar 7, 2017 at 5:54 PM, Anthony Holloway wrote: I'm not sure how often this gets configured out in the wild, but I do know that there are few different flavors of this feature, depending on your scenario. E.g., CUBE Media Anti-Tromboning The one I'm interested in would involve a tie line to a legacy PBX, and I'm not married to the IP technology: SIP, H323, MGCP, whichever will get the job done. CUCM ---IP---> VGW ---T1/PRI---> PBX The call scenario I'd like to see if we can avoid is when a call is established from CUCM to the other PBX, and then a phone on the far side transfers the call to an IP Phone on the CUCM side, I'd like the PRI usage to drop to 0 trunks in use, instead of nailing up 2 trunks for the duration of that call. I'm looking for your experience and feedback in configuring anti-tromboning in this scenario. One of the hardest parts in researching this, is knowing what terms to search for. I think I've narrowed down the terminology to QSIG Path Replacement, as described in the CUCM SRND. Though, without real-world working experience, it's hard to know if I'm right or not. So, I know there's some good legacy telco knowledge out on this list. What do you know? Thanks! __ _ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/ mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] TAPS - 3rd party SIP Phones?
LOL.
Thanks, Now I can tell the manager that not a good way to go.
On Friday, January 27, 2017 5:56 AM, "Heim, Dennis"
wrote:
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1.0in 1.0in;}#yiv9820845464 div.yiv9820845464WordSection1 {}#yiv9820845464 If
you sell the 3rd party phone on ebay and buy a Cisco one. Then you can.
From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]On Behalf Of Brian
Meade
Sent: Thursday, January 26, 2017 5:52 PM
To: Carlo Calabrese
Cc: Cisco VoIP Group
Subject: Re: [cisco-voip] TAPS - 3rd party SIP Phones? I wouldn't think so.
3rd party SIP phones usually are manually configured on the phone side to use a
certain extension/username/password. Some 3rd party SIP phones have
provisioning servers or can use any TFTP/Web server but nothing that CUCM is
going to be able to update automatically after a successful TAPS. On Thu,
Jan 26, 2017 at 4:45 PM, Carlo Calabrese via cisco-voip
wrote:
can you tap in 3rd party SIP phones? I need to add a site to a current
system and would like to use their phones before we can order more 7841s
Thanks
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[cisco-voip] TAPS - 3rd party SIP Phones?
can you tap in 3rd party SIP phones? I need to add a site to a current system and would like to use their phones before we can order more 7841s Thanks___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 8851 and Transfers
Are you running 10-3-1-20? I think that is the one with the bugwe are running
10-2-2-16
On Wednesday, October 5, 2016 2:28 PM, "Haas, Neal"
wrote:
#yiv2085856756 #yiv2085856756 -- _filtered #yiv2085856756
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{}#yiv2085856756 No side Car, and this is a shared line across multiple phones.
Thank You, Neal Haas From: Carlo Calabrese
[mailto:carlo_calabrese2...@yahoo.com]
Sent: Wednesday, October 5, 2016 2:17 PM
To: Haas, Neal ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] 8851 and Transfers Does this have a Side Car? we
had a problem with this awhile back, not sure if same model but we had to roll
back. On Tuesday, October 4, 2016 7:16 PM, "Haas, Neal"
wrote: We have 8851 phones, just upgraded to the new
firmware, now we have issues.
When a phone call is in the process of being transferred and a new call comes
in the phone will not let you finish the transfer. How can we stop this issue?
Anyone have a solution - besides rolling back the firmware?
Neal Haas
?
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Re: [cisco-voip] 8851 and Transfers
Does this have a Side Car? we had a problem with this awhile back, not sure if same model but we had to roll back. On Tuesday, October 4, 2016 7:16 PM, "Haas, Neal" wrote: We have 8851 phones, just upgraded to the new firmware, now we have issues. When a phone call is in the process of being transferred and a new call comes in the phone will not let you finish the transfer. How can we stop this issue? Anyone have a solution - besides rolling back the firmware? Neal Haas ? ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Cisco ATA 186
Have you tried changing the mac address on CUCM and let the ATA auto register? do both MAC address On Wednesday, October 5, 2016 8:42 AM, Asim Mekki Basheer wrote: Yes allowed since the IP PHONES connected to the same port. Regards ASIM MEKKI From: Lisa Notarianni Sent: Wednesday, October 5, 2016 6:23 AM To: Asim Mekki Basheer Subject: RE: [cisco-voip] Cisco ATA 186 Are you sure the IP and port are configured to allow the device to work in the voice VLAN? From: Asim Mekki Basheer [mailto:asim_...@hotmail.com] Sent: Wednesday, October 05, 2016 9:23 AM To: Lisa Notarianni ; Ashwani Ranpise ; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Cisco ATA 186 Yes its ATA 186 From: Lisa Notarianni Sent: Wednesday, October 5, 2016 6:15 AM To: Asim Mekki Basheer; Ashwani Ranpise; mailto:cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Cisco ATA 186 I have had this issue but the network configuration did not allow it to register or the device type is different than what you are actually trying to configure. It is a ATA 186 correct? From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net]On Behalf Of Asim Mekki Basheer Sent: Wednesday, October 05, 2016 9:13 AM To: Ashwani Ranpise ;mailto:cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Cisco ATA 186 please see the attached still showed rejected even after assigned The DN Thanks ASIMFrom: Ashwani Ranpise Sent: Wednesday, October 5, 2016 5:55 AM To: Asim Mekki Basheer Subject: Re: [cisco-voip] Cisco ATA 186 Asim,I dont see any DN has been assigned. Assign the DN to your ATA-186 device and it will work.Thanks,Ashwani On Wed, Oct 5, 2016 at 8:50 AM, Asim Mekki Basheer wrote: Hello Everyone we have CCM 7.5 ,recently we faced issue with fax the ATA showed rejected in the call manager we make factory reset the same issue Regards ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Finesse and web page dip
I have a group that does a web page dip thru CAD. Now I need to get them ready for Finesse, Can Finesse do the web page dip? trying to find documentation on it. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX Queuing and Over flow Queueing Best Practice
Because I don’t let supervisors onto the system to change agent skill levels, I set up 2 or 3 logins for each agent with different skill levels for what they are working on that day. So they can be primary one day and backup the next. I don’t deque them when I add them to the overflow ques so if someone because available, they will get the call and not a backup person. This seems to work well for the group and they can run reports to see when and how many calls are going to overflow staff. On Thu, 9/8/16, Max Harmony wrote: Subject: [cisco-voip] UCCX Queuing and Over flow Queueing Best Practice To: "Cisco VOIP" Date: Thursday, September 8, 2016, 12:11 PM I have a customer that is requesting to have a main Queue, and two overflow queues, and they intend to have the same agents in all three queues with the same skills. I am not very confortable with the user experience but can anyone advice on what would be the best recommendation for this situation? Please see below logicCaller --- goes to Queue A---35secs delay --->redirected to OverflowQueueAOverflow--5MinutesDelay--->Redirected to QueueAGeneralOverflow--5MinitesDelay-- -- -- Grace Maximuangu CloudPOP/InvictaCloud www.cloudpop.com “Go beyond your limits, push yourself, be the best you can be.Experience new cultures, broaden your horizons, stay connected.” -Inline Attachment Follows- ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Bug CSCug20588 - Phone registration error
Has anyone run into this bug? I can't find anything on it and was hoping to fix it before I have to open a ticket with TAC. I am running 10.5 SU2 and we started reinstalling a bunch of phones we removed awhile back. Thanks Carlo ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Time of day routing on UCCX 10.6
XML would be good. just need a few variables. date StartTime EndTime. The times would be in the form of = T[9:00:00 AM PST] Then do a if statement On Mon, 7/4/16, norm.nichol...@kitchener.ca wrote: Subject: [cisco-voip] Time of day routing on UCCX 10.6 To: cisco-voip@puck.nether.net Date: Monday, July 4, 2016, 7:15 AM I have a CCX group that works daily from 07:00 to 21:00 . They would like change their opening and closing times over July and August to some days opening at 08:00 and some days closing at 16:00 because of staffing issues. Can this be built in an .xml file like a holiday schedule or do we need to program this logic into the script. Thanks Norm Nicholson Telecom Analyst City of Kitchener (519) 741-2200 x 7000 -Inline Attachment Follows- ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Upgrading to UCCX 10.5 and sotware updates
We are planning to go from 8.6 to 10.5 for our CUCM and UCCX. My question is when you start the CAD software and it wants to upgrade it, Do you need to do this as an administrator on the PC? We are running Windows 7 & 8. Thanks Carlo ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DID Best Practice
It will depend on the size and complexity of your network. I like doing all routing in Call Manger. I route about 35,000 DNs to over 10,000 VoIP phones and 300 key systems with multiple connections to the PSTN. This way everything is standard and also staff can do most of the work (Add moves and changes). I don’t use gatekeepers as the CM can do it all. This way all calls are recorded in CDR and I forward that to an accounting system. On Thu, 1/14/16, David Zhars wrote: Subject: [cisco-voip] DID Best Practice To: "cisco-voip@puck.nether.net" Date: Thursday, January 14, 2016, 5:19 AM What is the best way to route DIDs? I see some setup on 2801s with num-exp DID# INTERNAL_EXT# They always seem to work, but someone told me the preferred way is through CM, apparently setting up a route point. Thoughts? -Inline Attachment Follows- ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Rightfax transmission errors
Can you send a part of the configs on the 3925, include the voice service voip voice class codec dial-peers I connect to 3 rightfax servers with no problems. The one problem we did run into on each one of them was the servers firewall settings. Try turning that off and see if it connects. Here is what I have on my 3945s. voice service voip ip address trusted list ipv4 172.xx.xx.146 ipv4 172.xx.xx.162 ipv4 172.xx.xx.54 ipv4 172.xx.xx.9 ipv4 172.xx.xx.100 ipv4 172.xx.xx.50 ipv4 172.xx.xx.82 ipv4 172.xx.xx.5 ipv4 172.xx.xx.20 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 3 hs-redundancy 2 fallback none h323 h225 timeout setup 5 modem passthrough nse codec g711ulaw sip bind control source-interface Loopback0 bind media source-interface Loopback0 early-offer forced ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! dial-peer voice 112 voip corlist incoming TOPSTN corlist outgoing TODVS tone ringback alert-no-PI description Inbound 10-digit VOIP peer preference 1 destination-pattern .T progress_ind setup enable 3 progress_ind progress enable 8 progress_ind connect enable 8 session protocol sipv2 session target ipv4:172.XX.xx.50 incoming called-number .T voice-class codec 1 dtmf-relay rtp-nte fax rate 14400 ! ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] configure Transcoders in AS5400XM router...
So for the router, enable the dsp farm under voice-card for each of the cards you are going to use. Then define the Call Mangers Then the groups Then the functions. I like to use the first letter to define what it is doing, C - Conference X - Transcoding M - Media Termination point Then in Call Manger, you will use the name for the gateway to register. You will need to reset them after you define them. Hope this is what you are asking. Carlo ! voice-card 0 dsp services dspfarm ! voice-card 2 dsp services dspfarm ! voice-card 3 dsp services dspfarm ! sccp local GigabitEthernet0/0 sccp ccm 10.1.1.10 identifier 1 version 7.0 sccp ccm 10.1.2.20 identifier 2 version 7.0 sccp ccm 10.1.3.30 identifier 3 version 7.0 sccp ! sccp ccm group 2 bind interface GigabitEthernet0/0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 5 register C-CCM-Gateway1 associate profile 1 register X-CCM-Gateway1 associate profile 3 register M-CCM-Gateway1 ! dspfarm profile 5 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 24 associate application SCCP ! dspfarm profile 1 transcode codec g711ulaw codec g729ar8 codec pass-through codec g729br8 codec g729r8 codec g729abr8 maximum sessions 50 associate application SCCP ! dspfarm profile 3 mtp codec pass-through codec g711ulaw maximum sessions software 100 associate application SCCP ! On Wed, 8/12/15, Mudassar M via cisco-voip wrote: Subject: [cisco-voip] configure Transcoders in AS5400XM router... To: "cisco-voip@puck.nether.net" Date: Wednesday, August 12, 2015, 2:18 PM Hello folks, trying to configure 5400 router as my media resource box for transcoders and conference bridges but unable to find sccp, dsp commands etc. IOS version is c5400-ik9s-mz.151-3.T4.bin and inventory is listed down. any one has idea about how to configure these PVDMS in 5400?Thanks for your help,MAK KKVG1#show inventory NAME: "AS5400XM chassis", DESCR: "AS5400XM chassis"PID: AS5400XM , VID: V05, SN: JAE162603MG NAME: "AS5X-FC_1", DESCR: "AS5000 Feature Card with Six AS5X PVDM DSP Module Slots"PID: AS5X-FC , VID: V01, SN: JAE163007ZF NAME: "PVDMII DSP SIMM with four DSPs 1/0", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221YC NAME: "PVDMII DSP SIMM with four DSPs 1/1", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221Z6 NAME: "PVDMII DSP SIMM with four DSPs 1/2", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221VH NAME: "PVDMII DSP SIMM with four DSPs 1/3", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221Y7 NAME: "PVDMII DSP SIMM with four DSPs 1/4", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221PH NAME: "PVDMII DSP SIMM with four DSPs 1/5", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC1632220Q NAME: "AS5X-FC_2", DESCR: "AS5000 Feature Card with Six AS5X PVDM DSP Module Slots"PID: AS5X-FC , VID: V01, SN: JAE163007ZD NAME: "PVDMII DSP SIMM with four DSPs 2/0", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221Z7 NAME: "PVDMII DSP SIMM with four DSPs 2/1", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221VF NAME: "PVDMII DSP SIMM with four DSPs 2/2", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221ZF NAME: "PVDMII DSP SIMM with four DSPs 2/3", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221W1 NAME: "PVDMII DSP SIMM with four DSPs 2/4", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221Q7 NAME: "PVDMII DSP SIMM with four DSPs 2/5", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221ZW NAME: "AS5X-FC_3", DESCR: "AS5000 Feature Card with Six AS5X PVDM DSP Module Slots"PID: AS5X-FC , VID: V01, SN: JAE16300812 NAME: "PVDMII DSP SIMM with four DSPs 3/0", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC16322200 NAME: "PVDMII DSP SIMM with four DSPs 3/1", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC1632220C NAME: "PVDMII DSP SIMM with four DSPs 3/2", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221PN NAME: "PVDMII DSP SIMM with four DSPs 3/3", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC163221VE NAME: "PVDMII DSP SIMM with four DSPs 3/4", DESCR: "PVDMII DSP SIMM with four DSPs"PID: AS5X-PVDM2-64 , VID: V01, SN: FOC16322203 NAME: "PVDMII DSP SIMM with fou
