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- About Robert Craig
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Member since
10-16-2010
240
Posts
27
Helpful
5
Solutions
Created by
Robert Craig
in Call Control
in Call Control
07-10-2017
10:12 AM
07-10-2017
10:12 AM
Great. I'll give this a shot. Thank you very much! Robert
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Created by
Robert Craig
in Call Control
in Call Control
07-10-2017
06:54 AM
07-10-2017
06:54 AM
I guess it would be the request URI. The first first line in the invite. 2017-07-10 09:53:30.344 INVITE sip:5551112233@1.1.1.1;user=phone SIP/2.0 Robert
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07-07-2017
06:49 AM
I am experimenting with dialing directly to our Oracle SBC's (instead of tandeming through another product that does translations for us). However, our Verizon SIP service requires the INVITE to have a specific domain in the INVITE message. Right now the invite from CUCM just has 5551112222@1.1.1.1. Verizon wants to see 5551112222@verizondomain.com . Does anyone have a LUA scrip that can accomplish this? I've seen a lot of scripts out there, but nothing that changes the INVITE domain. Thanks for any tips. Robert
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Labels:
Created by
Robert Craig
in Call Control
in Call Control
07-07-2017
06:37 AM
07-07-2017
06:37 AM
OK, thank you very much! Robert
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Created by
Robert Craig
in Call Control
in Call Control
06-05-2017
02:16 PM
06-05-2017
02:16 PM
Hello. I'm looking to remove and replace the FROM header as the invite ingresses from Skype for Business into CUCM. Would the below script accomplish this? The invite currently just says " first.last@test.com :10.1.1.1". I'm looking to replace it with a phone number so its routable. Any help is appreciated. Robert M={} function M.outbound_INVITE(msg) local from = msg:getHeader("FROM") --Get FROM Header local b = string.gsub(from, "(<sip:.+@)", "<sip:5551112222") --New FROM Header msg:modifyHeader("From", b) --Replace FROM Header end return M
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- Tags:
- lua
- sip-normalization
Labels:
08-31-2015
02:46 PM
I'm sure there are plenty of articles out there, but I can't seem to find them. Can anyone point me in the right direction for a good article on configuring a Site to Site between an ASA (behind a nat device) to a Cisco Router (public ip)? I am not using standard crypto maps on the router. I am using VTI. That articles I have found just use plain crypto maps. Any help is appreciated. Robert
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Labels:
07-01-2015
07:19 PM
OK, I'll give that a shot. Is there any settings in that file that would overwrite custom things the user did (adjust volume and save, change ring tone, etc.)? I'm thinking on my test phone that it pulls the initial file and keeps it. But, when I make a change in the user settings from the phones menu, it pulls the file on the next interval, sees a difference, and then reboots. Robert
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06-29-2015
07:51 PM
OK, sorry for the delay. Attached is the latest version of the config file I am using. The phone is still rebooting. I'm thinking it is after something has changed under the user preferences (such as ringtone) or something that the user changed from the phone. Any ideas? Robert
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06-21-2015
12:37 PM
OK, so how would location bandwidth be used from headend to remote end? If that path is out of bandwidth and I can't use AAR, what would be an alternative routing method I could use from the PSTN - in? Robert
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Created by
Robert Craig
in IP Telephony and Phones
in IP Telephony and Phones
06-12-2015
11:12 AM
06-12-2015
11:12 AM
Hey folks. I understand CAC when used for calls between remote sites. However, we are migrating all sites to a centralized pair of SIP Cubes at the head office. Now, how can I implement CAC with AAR for that route? I've already tried implementing it, however when the bandwidth is exceeded for a remote site to make a call through the CUBE and on to ATT PSTN, it just says "Not enough bandwidth". The call does not reroute the call over the remote site GW. Is this possible or is it only limited to calls within CUCM? Below is call flow. Remote IP Phone - MPLS - Corporate Office - PSTN SIP Router (at coporate office) I'd like a call to reroute over the local GW at the remote office if the bandwidth is utilized instead of through the corporate office SIP router. Any help is appreciated. Robert
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Labels:
06-01-2015
07:13 AM
Yeah, thats the file I keep handy and then just copy and paste into a new xml file for every phone. So there is a duplicate option? Also, if I am not changing an option, do I have to keep it in the file? Robert
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05-30-2015
09:07 PM
I have been trying to push phone configuration out through the use of a TFTP server. While the phones download their config, they also reboot themselves every two minutes. Nothing changes on the phone, but the phones continuously reboot every time they pull a config file down. Below is a config file that I usually edit for the respective labels/usernames/passwords. Is there anything that sticks out that would cause this issue? Any help is appreciated. Robert
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Labels:
02-13-2015
08:23 AM
OK, I added mode transport on both the hub and the spoke. The tunnel is staying up now. Thanks for the tip! Robert
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02-13-2015
05:14 AM
OK, I added "mode transport" at the hub, no effect. I can still see the tunnel drop and a peer trying to form with the private IP. Think I should add it at the spoke now and see how it performs? Robert
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02-13-2015
04:26 AM
No, there is only one spoke behind the ASA at this site. I only used "mode transport" at the spoke side, but haven't tried using it at the hub first. I might just have to do a PAT on the ASA. If I end up doing a PAT, would it be UDP 500? Robert
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07-10-2017
I guess it would be the request URI. The first first line in the
invite.2017-07-10 09:53:30.344 INVI...
07-07-2017
I am experimenting with dialing directly to our Oracle SBC's (instead of
tandeming through another p...
06-05-2017
Hello. I'm looking to remove and replace the FROM header as the invite
ingresses from Skype for Busi...
08-31-2015
I'm sure there are plenty of articles out there, but I can't seem to
find them. Can anyone point me ...
07-01-2015
OK, I'll give that a shot. Is there any settings in that file that would
overwrite custom things the...
06-29-2015
OK, sorry for the delay. Attached is the latest version of the config
file I am using. The phone is ...
06-21-2015
OK, so how would location bandwidth be used from headend to remote end?
If that path is out of bandw...
06-12-2015
Hey folks. I understand CAC when used for calls between remote sites.
However, we are migrating all ...
06-01-2015
Yeah, thats the file I keep handy and then just copy and paste into a
new xml file for every phone. ...
05-30-2015
I have been trying to push phone configuration out through the use of a
TFTP server. While the phone...
02-13-2015
OK, I added mode transport on both the hub and the spoke. The tunnel is
staying up now. Thanks for t...
02-13-2015
OK, I added "mode transport" at the hub, no effect. I can still see the
tunnel drop and a peer tryin...
Public Statistics
| Date Registered | 10-16-2010 10:56 PM |
| Date Last Visited | 08-15-2019 11:36 PM |
| Total Messages Posted | 240 |
| Total Helpful Votes Received | 27 |
Helpful Votes Given To
| User | Helpful Count |
|---|---|
| 15 | |
| 15 | |
| 20 | |
| 10 | |
| 5 |
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.jpg "Hall of Fame Master"){kind=icon}
Helpful Votes From
| User | Helpful Count |
|---|---|
| 5 | |
| 5 | |
| 4 | |
| 5 | |
| 8 |
