Hi all,
Asterisk have a two NICs: one NIC to wan and second NIC to lan.
Within the lan segment are phones (Linksys SPA921 & SPA922).
I have a problem with extension dial in outgoing calls. When dialing extension 589 (for example), asterisk send to SIPtrunk only 58.
tcpdump on nic “wan” and nic “lan” show that:
on dump lan i see 519
on dump wan i see 51
digit “9” is skipped.
Ie, before asterisk all dtmf are present in rpt flow, ad after asterisk some digits are skipped.
I’m use G711alaw codec and dtmf rfc2833. I’m try to change dtmf mode to info and inband.
I’m use asterisknow with asterisk 1.6.2.18.
Any ideas?
Thanks in advance.
I shared screenshots of wireshark (analyze voip calls):
http://hostingkartinok.com/image/01201106/36751a235c2fd9efcb27bdb8c9bbf703.png - from phone to asterisk. DTMF = 589.
http://hostingkartinok.com/image/01201106/25792fd81df06159b543acb37ee91024.png - from asterisk to trunk. Digit “9” is skipped.
