Hi all, I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says: ------------------------------------------------------------------------ 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2 ^^^^^^ 00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented ^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available ---------------------------------------------------------------------------- ------------ Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law. So if I call the router from a normal telephone, the debugging looks as follows: ---------------------------------------------------------------------------- ---- 01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3 ^^^ 01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s ---------------------------------------------------------------------------- --------- whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law. I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this. Has anybody in this mailinglist the same experience? Any hints would be very appreciated. Thanks yang