Sip info (RFC 2796) dtmf not being sent

[Elihunter]

I'm trying to get Sip info dtmf working on VVX 550 and 450 phones and am not able to see info packets being sent to the server when monitoring with sngrep.  I tested with a softphone and receive and am able to use dtmf when setting dtmf method to info so I know nothing is blocking these packets between my network and the server.

On the Polycom phones I see 

[PpsHybridC]: MsgPpsDtmfViaInfo Rejected

in the diagnostic log.

I've tried firmware 6.4.1.2280 and 5.9.8.5760.

I have 

voIpProt.SIP.dtmfViaSignaling.rfc2976="1"

and 

tone.dtmf.viaRtp="0" 

in the config file.

I can't find any info on the MsgPpsDtmfViaInfo Rejected error.  Does anybody know what this error would be caused by?

Thanks!

[SteffenBaierUK]

Hello @Elihunter ,

welcome to the HP Poly community.

You have posted in an incorrect section and a volunteer has moved your post. It has been moved to a section more suitable for your request.

There is no VVX 550 phone.

Is this ObiEdition or UC Software?

For UC Software check our VoIP FAQ article here:

Oct 7, 2011 Question: Phone unable to send DTMF to an IVR system or how to troubleshoot DTMF issues

Resolution: Please check => here <=

Other volunteers may reply but if you need urgent help please work with our support organization in your region. Details on how to contact us are in my signature

Best Regards

Steffen Baier

[Elihunter]

Thanks for moving this Steffen.  Sorry it's a VVX500 and I'm using UC Software.

I changed the logging level for pps to debug and think this is relevant.  It repeats each time I press a digit during a call.

1011132633|pps |2|00|[PpsHybridC]: mediaSess2PpsCallInBandData NetCall(0x19d34d0) UsrCall(0x41a949e8) Type(0) Size(1) Data(1)
1011132633|pps |1|00|[PpsHybridC]: mediaSess2PpsDTMFViaInfo ppsCall(0x19d34d0) soCall(0x41a949e8)
1011132633|pps |2|00|[PpsHybridC]: MsgPpsDtmfViaInfo bGlobalForceDtmfInfo = 0
1011132633|pps |4|00|[PpsHybridC]: MsgPpsDtmfViaInfo Rejected

I've tried resetting the VVX 500 to factory and manually configuring the SIP server connection info then importing the dtmf_sip_info.cfg from the resolution you linked.  Still sngrep on the server shows no dtmf info packets being received. 

I do get show INFO allowed in the invite SDP.

[SteffenBaierUK]

Hello @Elihunter ,

I have no idea what sngrep is and I can only refer you to the troubleshooting in the FAQ. Our Phones have the capability to use remote Wireshark so I suggest you use this and/or the SIP and other logging levels.

The VVX 500 will most likely be out of warranty but if you still have a phone in warranty I suggest you contact our Support organization in your region. Details are in my Signature.

Best Regards

Steffen Baier

[Elihunter]

Thanks Steffen, I got a pcap from the endpoint with Wireshark and saw INFO is not included in the Allow in the SDP from Asterisk using PJSIP.  Found this on the Digium forums.

https://community.asterisk.org/t/asterisk-pjsip-using-sip-info/88422

So this ultimately looks like an Asterisk issue.  I'm guessing a feature request to override the behavior and use SIP INFO would require going through a vendor.  Unfortunately my most recent purchases weren't through an authorized vendor.  If you don't mind checking the vendor for a VVS 250 mac address 48:25:67:23:80:66 I'd appreciate it.

Thanks for all your help!

[SteffenBaierUK]

Hello @Elihunter 

the MAC shows as sold via jenne.com but the warranty ran out in December last year. I still do not understand the problem here but there is no feature request that would be accepted as the VVX 500 is at the end of its life so nothing new is being developed to get a feature request accepted for the VVX 250 we would need to get a business case and even more important how many phones are we not selling for "missing" this feature.

Best regards

Steffen Baier

[Elihunter]

Ultimately I'm trying to get direct media working with Asterisk so calls in an office can route RTP directly to each other.  One of the requirements is out of band DTMF.  The problem is Asterisk isn't sending INFO as allowed in the SDP so the Polycom phone is not sending out of band DTMF.  I can either force Asterisk to use INFO but then the Polycom refuses to send INFO packets so direct media works but DTMF does not.

It seems to behave this way on current and previous 5.x and 6.x UC versions.  It's understandable and expected behavior from Polycom's side however sending the info packet wouldn't hurt anything despite not being allowed in the SDP from the Asterisk server.

I will say I have this working with a Yealink phone and Microsip softphone.  They seem to ignore the SDP allowed field and send DTMF as configured regardless.

[Elihunter]

If anybody else runs across this post Digium has fixed this in Asterisk.  I'm not sure which version the fix will be included with, prob 18.21 and 20.6.

https://github.com/asterisk/asterisk/issues/376