Shorten the Incoming call ring ( from PSTN )


I have problem on setting up asterisk@home. I would like to shorten the incoming call ring from PSTN. Currently, when somebody call my pstn line and get into the Digium card, they need to wait 3 rings then they will hear the AA ( digital receptionist ). Can I shorten it or probably if somebody call, he will hear the digital receptionist straight way ? I have tried to set callprogress=no or yes, but could not help me.
Thank you

Try setting any CallerID settings in zapata.conf to no.

The caller ID sent from the PSTN is sent as a burst of data signals between the first and second ring. If you intend to use the incoming caller ID information, you’ll have to tolerate the three ring wait to pickup.

If you tell the Digium card that you don’t have caller ID or not to use it, it should pick up sooner.

The setting for the incoming ports would be:



thanks for the reply. is that the limitation of asterisk with analog card ? I wan to setup for my own office. It will have 4 analog line and I also want to enable the called ID. any other way to do ? btw, i have set usecallerid to no. it could not help.
Many thanks

After you set the parameter to:


You have to restart the zaptel service on the linux box for it to take effect.

I suppose it’s a limitation of sorts. The caller ID has to be delivered, so you need at least 2 rings from the PSTN. If the call is going to be answered by an IVR application that will need the caller ID data, you can’t have it answered immediately.

I guess since the callerID information is sent in the call setup message for SIP phones you have to wait for two rings there too.

The option is to get yourself a PRI. It’s a pricy solution though.


okie thanks for your suggestion. But I dont think the similiar issue happens to Key phone or PBX systems that uses analog line that are hunted together. So Its not the analog line, isnt it ?


Yes, it actually would happen with other systems.

Ringing a SIP phone system and ringing an analog phone system are two very different things, but with analog phone lines, the deilvery of caller ID data remains the same.

When an analog key system that can use caller ID gets an incoming call, the ringer voltage is detected by the key system, which signals the system to ring it’s phones. After the first ring, the caller ID data is received, and sent to all of the phones, followed by the second and subsequent rings. Everything happens pretty much at the same time it happens from the PSTN. A person who’s answering a phone has the option to answer right away, or wait for two rings to see the caller ID data.

However, even with an IVR (not an analog phone) as the destination, you would STILL have to wait at least two rings. If the IVR answered on the first ring, the caller ID would not be delivered. If you want the IVR application to have caller ID information you HAVE to wait two rings for it to be delivered. This also happens with your SIP system. It’s not different at all.

When delivering calls to phones with a SIP phone system, you can’t start the ringing right away. The message sent to a SIP phone does everything (delivers the caller ID, starts the phone ringing) and is sent and received just once. You can’t send a message to start the ring, then send a message to deliver the caller ID, then another message to keep it ringing. The SIP phone would see that as three incoming calls.

So, you have to wait for the caller ID delivery from the PSTN to happen before you have all the information you need to form the SIP message you send to the SIP phone, or use in your IVR application.

As I mentioned, the fix for this is an ISDN line. Like a SIP message, an incoming ISDN call setup message delivers the caller ID and an instruction to start ringing all at once. The Asterisk system can then begin offering it’s services right away.

thank you so much for the explanation.


Although I agree with the explanation of how analog caller id is delivered, the hardware and software should still have the ability to answer as soon as the caller id is received. This would be exactly 1 and 1/2 rings. The system should be fast enough to collect caller id and route the call. There is no mechanism to re-send caller id if we did not get it right the first time, so halfway between the first and second ring we got it or not!


Remember that there are readers on these forums from many countries, and not all Caller ID works the same way across those countries! IIRC in the UK it come in prior to the 1st ring.



I have problem again. After I enabled the telephone line with caller id display. The caller will hear the AA longer than before. Now it will hear the welcome message after 8 seconds. Before I enabled with caller id, it only took 2 secs. Any suggestion or there are some configs i need to change.