Inbound DTMF signaling

[Roy Reynolds]

Hello, 

The IVR on an installed system worked well for inbound calls over the years.  Suddenly it's only working if the calls are over an originating sip trunk or landline. Some mobile calls fail to signal the ivr.

The carrier is saying we need to implement transcoding at our end.  I don't understand their reasoning but want to know what options are available to allow dtmf to work consistently.

Thanks.

[Support]

Mobile networks may use EVS (hidef), AWR-WB (G722.2 wideband) and AWR-NB (legacy narrowband) codecs.  Whereas most sipX implementations rely on G711 PCMU in the US.

So the SBC you are using towards PSTN, either directly or from a PSTN connectivity provider, must correctly transcode those codecs to PCMU, including DTMF SDP parameters. And sometimes it is also necessary to strip or modify SDP parameters in the SBC to something the Freeswitch IVR accepts.

We have done a number of direct mobile network SBC integrations with sipX towards carrier grade SBCs that transcode the above mentioned EVS / AWR-* codecs towards PCMU and PCMA , also on the 24.01 CentOS7 branch, so if you use the built in sipX SBC it should handle DTMF correctly from cell networks. 

--
You received this message because you are subscribed to the Google Groups "sipxcom-users" group.
To unsubscribe from this group and stop receiving emails from it, send an email to sipxcom-user...@googlegroups.com.
To view this discussion visit https://groups.google.com/d/msgid/sipxcom-users/CALz8Txn4a_wF3b_cta%2BL5hdoxTisid9m%2BPpoN4G8QTP9eiUHdA%40mail.gmail.com.

[Roy Reynolds]

We use the built in sbc all the time.  Since this system has been running for a very long time, I don't think it's on release 24.x.

If that's the only fix, then we have to look at an upgrade path. 

Thanks for the response.

[Support]

We are of course happy to help. Other upgrades from e.g. the Ezuce UniteMe to 24.01 have been quite smooth, and not e.g. required any backup / restores procedures etc.  

If you have remote SIP extensions / users we also recommend transitioning to SIP TLS and SRTP end to end while you are at it. 

In addition to security benefits this also circumvents the very cumbersome SIP ALG function that is hardcoded and / or default enabled in many home routers these days. The reason is the router ALG mechanism cannot break / modify TLS encrypted SIP packages.

[Roy Reynolds]

I'm honored by the offer to help.

The system maybe on CentOs 6 so it's more than just sipxcom upgrade. However, if the os is 7, it sounds like we can set the repository to sipxcom 24.x and do a yum upgrade with no issues. I was of the impression that the systems had to be 19.4 or above to have a smooth upgrade. 

I will look into the srtp and tls for remote extensions as well. 

Thanks.

[Oyibo Dickson]

Hello bro I can help you setup any country sip trunk did number ivr setup for both inbound outbound number 

Inbox me on telegram @isellsiptrunk

[Donkiss Boss]

Licensed Pharmaceuticals And Psychedelics Online Store
Buy the best Pharmaceuticals And Psychedelics products in our website with safe and guaranteed shipping and delivery Worldwide
Check out our website below if interested in our products.
https://shorturl.at/BvDVf
https://shorturl.at/ITBJW
https://shorturl.at/wCGS4
https://shorturl.at/RnP0b
https://shorturl.at/GTr8Q