10-01-2020 02:35 PM
Hello all,
I have a Cisco VG350 which has one of its FXO ports connecting to another PABX which is configured as a hotline (ie. has no digit matching/route pattern in it's config, it just looks for a loop on the line then routes the call).
The intial config had an entry into its Route Paterrn (for the sake of this topic the DN is 1234) which then routes to the specific FXO port. The far end PABX we thought would disregard any digits sent as there is a loop on the line so doesn't care about the digits, but the conclusion was the PABX sent back what sounded like an NU tone and I think it was getting confused by the digits being sent. As the far end system isn't ours we cannot modify that config so we have to change ours to suit.
We then decided to use PreDot trailing on the CUCM RP (i.e. '12345.') so no digits are sent. This worked ringing the far end, however the initiaing 3rd party SIP phone is a trader type touchscreen which needs to know the DN in order to identify what far end phone is being contacted (customer requirement).
Hence my question; could the FXO complete the digit stripping instead? The 3rd party SIP phone has no logial interface to the VG350 during the call initiation so we could change the CUCM RP to '12345', remove the PreDot digit stripping on CUCM and the third party phone would be none the wiser.
The Cisco VG350 communicates to CUCM via MGCP. The port in question is FXO 0/0/1, specific config from the VG350 below (all completed on CUCM and pushed to the VG350):
!
mgcp
mgcp call-agent 10.100.10.2 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
ccm-manager music-on-hold
!
ccm-manager redundant-host 10.100.10.3 10.100.10.4
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager config server 10.100.10.5 10.100.10.1
ccm-manager config
!
voice-port 0/0/1
no battery-reversal
cptone GB
timing hookflash-out 50
impedance complex2
!
dial-peer voice 999001 pots
service mgcpapp
port 0/0/1
!
Thanks in advance!
10-01-2020 10:42 PM
You won’t be able to do this with MGCP controlled port as it’s a client/server relation style of control. You’d need to have the control of the port transfer to IOS to make modifications to called number, or calling if this was your requirement, in the gateway. For this you’d need to use SIP or H.323, where the preferred these days would be SIP.
10-02-2020 12:47 PM
Using MGCP protocol for VG, digit strip will be from CUCM. As @Roger Kallberg mentioned you can try SIP or H323.
10-03-2020 12:50 AM
Thanks both for your reply.
I have a few more follow up questions:
https://frgtech.wordpress.com/2013/11/20/configure-voice-gateway-vg224-in-sip-with-cucm/
Using the above as an example, what addititional config would be needed to strip 28010 from the FXO port (equivalent of CUCM RP '28010.' PreDot stripping)?
voice-port 2/20
ring frequency 50
cptone FR
description **telephone analogique**
station-id number 28010
!
dial-peer voice 28010 pots
description **telephone analogique**
destination-pattern 28010
port 2/20
!
dial-peer voice 201 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
!
dial-peer voice 200 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 10...
session protocol sipv2
session target sip-server
codec g711alaw
!
sip-ua
sip-server ipv4:10.20.2.27
!
3. Is there a guide of how I can configure SIP on CUCM to the VG?
10-03-2020 04:16 AM - edited 10-04-2020 12:09 AM
It is absolutely possible to use both protocols simultaneously. In CM you just configure a SIP trunk that points to the IP of the gateway and create a route pattern for the number you have on the FXO port.
In the gateway you remove the service mgcpapp from the dial peer and define a destination pattern for the directory number that you would send to the FXO. For a pots dial peer the default behaviour is to consume any digits that are specifically matched by the destination pattern, so the digit striping should by this be automatically solved for you. You would also need to define dial peers for the connection to/from your CM.
10-03-2020 08:13 AM - edited 10-03-2020 08:32 AM
Went in and took the parts that where of interests from our gateway configuration template for you to use as a reference if you want.
voice service voip ip address trusted list ipv4 10.138.16.32 ;CPE Subscriber ipv4 10.138.16.33 ;CPE Subscriber ipv4 10.138.16.34 ;CPE Subscriber ;add as many line as there are CPE nodes in the CM cluster sip bind control source-interface GigabitEthernet0/0/0 bind media source-interface GigabitEthernet0/0/0 ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 codec preference 4 ilbc ! voice class e164-pattern-map 1 description E164 Pattern Map for called number to CUCM e164 +46555777T ! voice class server-group 1 ipv4 10.138.16.32 preference 1 ipv4 10.138.16.33 preference 2 ipv4 10.138.16.34 preference 3 description Inbound calls from PSTN to CUCM ! voice class sip-options-keepalive 1 description Used for Server Group SIP OPTIONS PING ! sip-ua g729-annexb override retry invite 2 timers trying 300 no remote-party-id ! controller E1 0/0/0 clock source line primary pri-group timeslots 1-31 description DID Range: +46 555 777 (XXXX) no shutdown ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn negotiate-bchan resend-setup no cdp enable no shutdown ! voice-port 0/0/0:15 bearer-cap Speech
cptone SE ! dial-peer voice 1000 voip description Outbound calls from UCM voice-class codec 1 session protocol sipv2 incoming called-number . dtmf-relay rtp-nte sip-kpml no vad ! dial-peer voice 1010 voip description Inbound calls to CUCM subscribers modem passthrough nse codec g711ulaw session protocol sipv2 session server-group 1 destination e164-pattern-map 1 voice-class codec 1 voice-class sip options-keepalive profile 1 dtmf-relay rtp-nte sip-kpml no vad ! dial-peer voice 100 pots tone ringback alert-no-PI description Inbound calls from PSTN incoming called-number . direct-inward-dial ! dial-peer voice 110 pots tone ringback alert-no-PI description Outbound calls to PSTN destination-pattern 0T progress_ind setup enable 3 progress_ind alert enable 8 progress_ind progress enable 8 progress_ind connect enable 8 progress_ind disconnect enable 8 no digit-strip port 0/0/0:15 no sip-register !
10-03-2020 03:18 PM
Thanks for the example config, very useful!
Are you saying that by default the digits are stripped unless you add the 'no digit-strip' command?
Also, on the ports that will remain as MGCP, do I need to add the 'no sip-register' command?
10-04-2020 12:00 AM - edited 10-04-2020 12:07 AM
You’re welcome.
To answer your questions, yes a pots dial peer will consume any explicitly matched digits if you don’t have that command. As an example if you have a destination-pattern of 1234 it would consume (drop) all of these. So either you use that built in functionality or you can create a voice translation profile that you set in the outbound direction on your voice port or dial peer to drop the digits.
For the other question, no you should not need to add that as it pertains to the specific case where pots dial peers with SIP tries to register to SIP SRST on the router and even so I don’t think it would apply to the MGCP controlled pots dial peers.
10-04-2020 11:25 AM
Thank you.
So from the config above (I've copied below for convenience), what would happen if a call was made to 28010 whne it routes through port 2/20?
voice-port 2/20
ring frequency 50
cptone FR
description **telephone analogique**
station-id number 28010
!
dial-peer voice 28010 pots
description **telephone analogique**
destination-pattern 28010
port 2/20
!
dial-peer voice 201 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
!
dial-peer voice 200 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 10...
session protocol sipv2
session target sip-server
codec g711alaw
!
sip-ua
sip-server ipv4:10.20.2.27
!
10-04-2020 12:19 PM - edited 10-05-2020 01:41 AM
Change dial peer 201 to not have the session target as this is for an outbound dial peer and add codec g711alaw, without it you’ll get g729, and add incoming called-number . (dot). Also add no vad to both voip dial peers.
You’d likely want to have some dtmf relay function defined on these, like I have in the example.
Do you only have one CM? If not I’d recommend you to use the server group from my configuration example or multiple dial peers towards CM for each CPE node. Otherwise the routers security features for voip will reject the call if it comes from any of the other CM nodes.
When a call hit dial peer 28010 to directory number 28010 it should route it out via port 2/20 with an empty called number.
Edit:
Added the configuration as I would have made it.
voice service voip
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
!
voice class server-group 1
ipv4 10.20.2.27 preference 1
description Server group for CUCM
!
voice class codec 1
codec preference 1 g711ulaw
!
voice-port 2/20
ring frequency 50
cptone FR
description **telephone analogique**
station-id number 28010
!
dial-peer voice 28010 pots
description **telephone analogique**
destination-pattern 28010$
port 2/20
!
dial-peer voice 201 voip
description **Incoming Call from CUCM SIP Trunk**
voice-class codec 1
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte sip-kpml
no vad
!
dial-peer voice 200 voip
description **Outgoing Call to CUCM SIP Trunk**
session protocol sipv2
session server-group 1
destination-pattern 10...
voice-class codec 1
dtmf-relay rtp-nte sip-kpml
no vad
!
sip-ua
g729-annexb override
retry invite 2
timers trying 300
no remote-party-id
10-08-2020 10:15 AM
It looks like I can get what I need to working if I leave the RP as '12345.', but instead of PreDot stripping on the RP I perform NANP: PreDot on the RG.
What is the difference between performing PreDot on the RP and NANP PreDot on the RG?
10-08-2020 10:52 AM - edited 10-14-2020 12:57 PM
Primary difference is how the calling endpoint is updated with any transformation done on the called number. Doing it on the RP level will update the calling device display of called number for changes done, whereas putting it on RG level can result in no update to the calling endpoint. What can influence this on the RG level is what control protocol that is used for the gateway. Especially this is so if H.323 is used.
10-08-2020 01:19 PM
Thanks for your reply.
The 3rd party SIP phone cmmunicates to CUCM via SIP (obviously!) and the VG350 communciates to CUCM via MGCP.
It seems when the PreDot digit stripping is placed on the RP, the 3rd Party SIP phone loses visibility of the DN (12345) and therefore cannot identify who they are calling.
When the NANP:PreDot stripping takes place on the RG, the 3rd party SIP phone behaves as exepected.
Why is the case when all I have done is moved the PreDot stripping from the RP to the RG? I thought the call ins't processed to the VG350 FXO port until after the RG is completed.
10-08-2020 10:31 PM - edited 10-15-2020 01:36 AM
As I wrote before the CM will in most instances not pass along the update of number manipulation when it’s done on the RG level. This behavior depends upon what control protocol that is used for the entities in the RG. You’ll get different behavior for example with a H323 gw if you haven’t configured it with a command that “tells” CM to not pass along the number transformation information. As you use MGCP you should not need to worry about this as it’s the default behavior for a MGCP controlled gateway.
Sorry for forgetting about this specific earlier, it’s been a quite long time since I did anything that didn’t involve SIP gateways and there you would naturally be doing this directly in the gateway.
10-14-2020 12:34 PM
I can confirm adding 'NANP PreDot' on the RG config has solved the prolem!
I still don't understand why PreDot on the RP didn't work but PreDot on the RG does, I would've thought the SIP reply to the SIP phone would take place after the RP, RL and RG process took place?
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