Hi
I have a grandstream ht503 gateway. I configure a SIP account XXXX on fxo port.
When I run Dial(SIP/XXXX/YYYYYYYY) I immediately have the answer log on my asterisk CLI!! I mean it don’t wait until the YYYYYYYY answers!!
Which configuration should be changed?!
This is a question for Grandstream.
However, I cannot see any answer supervision or ringback detection options in their user guide, so, even if your PSTN provider supports answer supervision, there may be nothing that you can easily do. It does have some options for disconnect supervision.
Hi there,
Could you describe more detailed? I’m afraid I don’t fully understand what you are complaining.
What he is saying is that the HT503 (or the old 488) can’t get a ${DIALSTATUS} after the call has been bridged with the PSTN line. I have the same problem.
It occurs when you try to perform something automated outbound. In my case a recording for a data center alert.If I shove a call file in /var/spool/asterisk/outgoing to call everyone regarding a critical temperature alert, asterisk only sees that the SIP channel was answered. Meanwhile the PSTN line is still ringing after the bridge and thinks the call was answered. In my example below it’s playing the alert while the PSTN line is still ringing because as far as asterisk is concerned the SIP channel was answered.
A good example of this is if you try to use Playback() with the skip option.
extensions.conf
[alarm]
exten => s,1,Wait(3)
exten => s,2,Wait(2)
exten => s,3,Playback(newdigits/crit_temp2,skip)
exten => s,4,Wait(3)
exten => s,5,Hangup
verbose log
-- Attempting call on SIP/XXXXXXXXXX@ht488 for s@alarm:1 (Retry 1)
> Channel SIP/ht488-00000037 was answered.
-- Executing [s@alarm:1] Wait("SIP/ht488-00000037", "3") in new stack
-- Executing [s@alarm:2] Wait("SIP/ht488-00000037", "2") in new stack
-- Executing [s@alarm:3] Playback("SIP/ht488-00000037", "newdigits/crit_temp2,skip") in new stack
-- <SIP/ht488-00000037> Playing 'newdigits/crit_temp2.gsm' (language 'en')
-- Executing [s@alarm:4] Wait("SIP/ht488-00000037", "3") in new stack
-- Executing [s@alarm:5] Hangup("SIP/ht488-00000037", "") in new stack
If anyone has any insight or if I am just doing it wrong please let me know. I need to purchase a reliable ATA to make this work and Grandstream isn’t scoring any points with me at this point. 
[quote=“GS_yhan”]
Could you describe more detailed? I’m afraid I don’t fully understand what you are complaining.[/quote]
Answer supervision means that the network signals that the call has been answered. Disconnect supervision means that they signal that it has cleared. On a analogue line various methods are used, and having none available is quite likely on a line that is not intended to be used with a PABX.
For answer supervision to be passed to Asterisk, as well as being provided by the network operator, the equipment terminating the line must be able to recognize it. As waiting for answer supervision on a line that doesn’t provide it would prevent the call connecting, this has to be configurable. The documentation for your ATA does not include any options to configure it. I therefore conclude that it doesn’t support it.
In the absence of answer supervision, the SIP gateway needs to assume that the call has been answered immediately, so it generates SIP answer supervision immediately. Asterisk does support analogue line answer supervision, when used with directly attached analogue line interface cards, but when answer supervision isn’t configured, it still assumes an immediate answer.
Consequently, with your ATAs, the SIP call will always be answered no later than after the sending of the last digit.