DTMF issue

Dear friends, i have a question … I have a SIP trunk running and working.
If I can an internal extension, when the call party places the call, if i press any number at phone’s dialpad, the call party can cleary listen it. If I call somene using This sip trunk, my call party cannot listen the tone if the call party presses some key i can listen the tone. It’s causing a problem because i’m unable to dial to electronic services like credit card operator / phone operator that uses tones.

Can someone please help me on solving it?

Many thanks

Assuming it’s a PJSIP trunk, you can try playing with the DTMF Mode option by editing the trunk, and going to the pjsip Settings’ Advanced tab. It should be ‘auto’ by default. Typically, you shouldn’t have to change this, but it’s worth trying. I think RFC 4733 is common for most providers these days. Do you have any issues with DTMF on inbound calls over the same trunk, for example to an IVR? Also, can you give the name of the provider, or any asterisk system based setup guides they offer? It could help to see if they require a setup that’s different from the usual.

Thank you so much for your reply! I Tried changing Dtmf settings under trunk / pjsip/ advanced options and performed an asterisk restart after each changes. The issue persists. I also tried to disallow all codecs and allowing only ulaw once my internet link is 200mb/s and ping latency is low considering that my provider is overseas. If the person that receives the call presses a key, I can hear. If I transfer this call to a test menu i have created, this person can press the keys and FreePBX can easily understand the commands, directing the call according to the menu options. The problem only happens when I try to call some number using an extension. The other side can’t receive my dtmfs. I Have made calls to some 1800 services like credit cards and customer services like bestbuy. These services are not receiving my dtmf signal. O also tried to call some friends and pressed some keys in my phone. All them told me they didn’t hear anything. So I really don’t know if it could be a problem with my ata config (grandstream ht814) or if it could be something with my extensions config (pjsip) or something with my provider. It’s a cheap calls provider from betamax (calleasy.com in my case) they don’t give so many parameters to be set, the informations are so basic:

Software configuration

SIP port : 5060
Registrar : sip.calleasy.com
Proxy server : sip.calleasy.com
Outbound proxy server : leave empty
Account name : your CallEasy username
Password : your CallEasy password
Display name/number : your CallEasy username or voipnumber
Stunserver (option) : stun.calleasy.com

Below are some examples of the software configuration of various popular SIP devices. Please also consult the manual that came with your SIP device

G.711 (64 kbps)
G.726 (32 kbps)
G.729 (8 kbps)
G.723 (5.3 & 6.3 kbps)
GSMFR (13.2 kbps)

If you have audio problems:
Use a STUN server (e.g. stun.calleasy.com) with port 3478 (if supported by your device)
Use the G.711 codec

I really don’t know how to solve it, I will try new tests here like using a softphone to call some number and test, instead of an ata, I will also try using a Cisco iP phone connected to another extension. I will also try to delete the trunk and create it again. I will keep this post updated and I will be so thank if you could give me more tips/ideas to try here.

Have a great week

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