Hi All, I struck with codec config issue. Please suggest me if you have any idea with codec configuration of asterisk. I have simulated Chan phone code of asterisk to my own driver and able to make calls between my own driver and external sip [X-lite: Configured alaw, ulaw] using asterisks pbx. Added ulaw, alaw, g726, and g729 codecs as capabilities of ast_channel_tech and able to make both incoming and outgoing calls with u-law. But not able to make outgoing call [from Phone to X-lite] with *a-law* where as incoming call is working fine.Config files are shown below, Sip.conf [siptrunk] username = 3300 fromuser = 3300 type = friend outboundproxy = 192.168.5.100 fromdomain = 192.168.5.100 host = 192.168.5.100 secret = 12345 qualify = no insecure = invite port = 5060 session-timers = refuse context = EXT_3300 *disallow = all* *allow=alaw* canreinvite = no Outgoing call failed with ?*No* *audio format found to offer. Cancelling call to EXTEN*? and also some times I am seeing below print from asterisk CLI. ?*Dropping* *incompatible voice frame on Phone/1 of format unknown since our native format has changed to 0x90c (ulaw|alaw|g726|g729)*? Observations: 1. If I configure ast_channel_tech capabilities as only *a-law* then I am not seeing this issue and able to make both incoming and outgoing calls. 2. If I add g723 also in capabilities [ulaw, alaw, *g723*, g726, g729] then I am seeing set_format always selecting G723 codec and trying to do some translation. In this case ulaw also is not working. 3. Tried *core show translation* from asterisk CLI and it shown all codecs [ulaw,alaw,g723,g726,g729?.]. Consider: - Is any thing need to configure from conf files [sip.conf / phone.conf: Actually I am not reading any config from phone.conf]. - Is phone driver restricted to use only one codec at a time [I mean, need to register only a-law as capabilities] - I have tried with other codec [g726] and I am facing same outgoing issue. I will check with g729 also. I found similar issues reported by other people also, but I did not get any root cause for this issue. Please help me. Thanks in advance. Regards, garge -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100825/b89217a5/attachment.htm