I have Asterisk Server and TalkSwitch IP PBX (TS). I created a SIP Trunk in Asterisk server connecting to TS.
The registration is successful, I can make outgoing calls and receive incoming calls without any problems.
I enabled the IVR into this SIP truck so that the caller can enter the desired extension.
The problem, SIP trunk does not recognize the dial tone from analog phones or PSTN connected to TS while it is working fine with IP Phones connected to TS.
The problem lies with your PSTN connectivity provider, i.e. the person with the FXO or ISDN interface. SIP is intended to operate with pre-keyed numbers.
The problem not with PSTN line only, it is also with analog phones connected to other PBX in same LAN.
I am using another PBX because I do not have FXO/FXS cards installed in asterisk server.
When pressing any key in analog phone does not have any effect.
The essential point is that SIP has no concept of dial tone, so any problem related to failing to recognize/wait for a proceed to dial signal has to be on the device with analogue connections, not that on the device that is purely SIP.
Sorry, I meant to say touchtone or DTMF not dial tone.
For more clarification below id my network diagram.
What have you got for DTMF options on that TalkSwitch? rfc2833? info? inband? I’d be curious why one TalkSwitch phone registered to Asterisk is fine and another TalkSwitch phone registered to TalkSwitch SIP’d to Asterisk is fine, but an analog phone tied to TalkSwitch isn’t. What DTMF mode are you using for the SIP connection between the TalkSwitch and Asterisk?
I just came across a Talkswitch 480s and would like to configure it to operate the same as you did; which is to use the talkswitch with a PTSN line connected to it as a trunk for my asterisk system. I tried to mimic your setup to no avail.
Did you use pjsip or sip for the trunk and extension? did you have “direct dial” turned on/off on the talkswitch?