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I already posted but no reply;

here is my problem when I try to use the sip phone after 20 second it drops, it's not about sip phone or codecs it's asterisk it drop the call the message is

php: line:1592 - Requesting DTMF ::> Len-8
Apr 19 12:26:07 WARNING[4065]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission c9461229f652b379Mzc1ZWZkOWEzNjhiMDk4MDM3YjVmZjg0MTYxZmM3ZjE. for seqno 2 (Critical Response)
Apr 19 12:26:07 WARNING[4065]: chan_sip.c:1245 retrans_pkt: Hanging up call c9461229f652b379Mzc1ZWZkOWEzNjhiMDk4MDM3YjVmZjg0MTYxZmM3ZjE. - no reply to our critical packet.


and always you found after that enable to write frame

please help

I posted a fix here for that issue on Asterisk 1.2.

Whilst I don't run it, I've been looking at the source of Asterisk 1.4 today and it seems it was changed to behave in a similar way for v1.4.0. There have since been other changes to what is considered a critical packet between 1.4.4-1.4.5. Those changes look quite reasonable to me, but then so did the original patch for 1.2. Beware and test thoroughly before deploying a new build of Asterisk to production.

Could be a nat/firewall issue but I've seen that problem in the past where Asterisk was not receiving the responses from the UA. In this case it was a configuration issue. The UA should be configured to use a port such as 5070 and not the default 5060. Then this port is configured in Asterisk peer for this UA eg.

[7000]
username=7000
secret=700021
host=dynamic
port=5070
...
What this means is that the UA will signal out on its local 5070 towards Asterisk 5060 and asterisk will respond to the UA IP/5070.

asterisk : source port =5060 , destination port = UA/5070
UA : source port =5070 , destination port = asterisk/5060

Its best to let asterisk be the only device on the network using 5060 for SIP outbound signalling.

In the above context, port 5070 should be opened for outbound in the firewall if one exists between Asterisk and the UA.