25 Apr
2003
25 Apr
'03
4:54 p.m.
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