IPBlue phone continues ringing after call is picked up…

As a continuation of the previous post Configuring Asteriskfrom the Ground Up (1/2), this post is about an issue I encountered and its resolution. 


When calling IPBlue (Cisco softphone) from Xlite (SIP client), after I picked up the call (on IPBlue) the ring-tone continues and the call could not be established. 


Reviewing the console, when IPBlue is first registered, its capability is set to '0'.
Also, a warning message "Don't know any of 0x0 formats" shows up after the call is picked up. 

-- Device 'SEP00215D2B1296' successfully registered Device capability set to '0' Adding button: 9, 1     -- Executing [1000@DLPN_DialPlan1:1] Dial("SIP/6000-2a2fc000", "Skinny/1000@ipblue") in new stack     -- skinny_request(1000@ipblue)     -- Called 1000@ipblue     -- Skinny/1000@ipblue-4 is ringing [Jan 22 16:02:50] WARNING[1490]: channel.c:700 ast_best_codec: Don't know any of 0x0 formats

The warning message suggests it may has to do with the codec and the resolution is as follows.

1. Allow all audio codecs for both IPBlue and SIP phones:

skinny.conf

[ipblue] type=7960 ;context=default line=1000            ; Dial(Skinny/1000@ipblue) device=SEP00215D2B1296 allow=all

sip.conf

;;; SIP Client - J Lam, Jan 2012 [xlite] ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend                     ; Context used for both inbound and outbound calls regexten=1234                   ; When they register, create extension 1234 host=dynamic                    ; This device needs to register allow=all

2. On both IPBlue and Xlite clients, permit G.711 is used:

IP Blue’s setting:

X-Lite Preferences:

Since I am not sure which of the G.711 codecs (alaw, ulaw) is used by IPBlue, I have enabled both on X-Lite.

3. Reload SIP module to enable sip.conf, and force skinny client to re-register:

The following messages appear on Asterisk console (asterisk –rvvvvvgc)

home-unix6*CLI> sip reload  Reloading SIP>   == Parsing '/usr/local/etc/asterisk/sip.conf': Found   == Parsing '/usr/local/etc/asterisk/users.conf': Found   == Parsing '/usr/local/etc/asterisk/sip_notify.conf': Found home-unix6*CLI> skinny reset SEP00215D2B1296     -- Device 'SEP00215D2B1296' successfully registered Device capability set to '268' Adding button: 9, 1

Note: ensure skinny client’s device capability is NOT set to ‘0’…

4. SIP and Skinny clients can now call each other successfully. i.e. no more ringing tone after Skinny client picks up the phone.

Voice is also working fine.