Hi all,
I have a PPC box (IBM RS6000 43P-150, bigendian afaik) which runs Fedora
Core 5 Test1 and zaptel, libpri and asterisk 1.2.0. I also installed
chan_capi (0.6.1) so I can use my Eicon Diva Server BRI card. Asterisk
was compiled with DEBUG=-g and DEBUG_THREADS = -DDUMP_SCHEDULER
-DDEBUG_SCHEDULER-DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS. Next I
did make clean, make valgrind, make install. Asterisk runs as user/group
asterisk/asterisk.
SIP <--> SIP calls are fine, Calls from SIP out to the PSTN via
CAPI/ISDN are fine too. ISDN/CAPI --> SIP calls don't work. Example
output of the issue is below. Anyone have an idea how I fix this?
Thanks and regards,
Patrick
chan_capi registers fine:
**********************************************************************
[chan_capi.so] => (Common ISDN API for Asterisk)
== This box has 1 capi controller(s).
== Reading config for BRI1
-- ast_capi_pvt BRI1-pseudo-D (<MSN1>,<MSN2>,capi-in,0,2)
(1,4,128)
-- ast_capi_pvt BRI1 (<MSN1>,<MSN2>,capi-in,0,2) (1,4,128)
-- ast_capi_pvt BRI1 (<MSN1>,<MSN2>,capi-in,0,2) (1,4,128)
-- listening on contr1 CIPmask = 0x1fff03ff
== Registered channel type 'CAPI' (Common ISDN API Driver ($Revision:
1.115 $) )
== Registered application 'capiCommand'
== Registered custom function VANITYNUMBER
Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2):
**********************************************************************
== BRI1: Incoming call '<my GSM>' -> '<MSN2>'
-- Executing Macro("CAPI/BRI1/<MSN2>-0",
"stdexten|1003|SIP/1003")
in new stack
-- Executing Dial("CAPI/BRI1/<MSN2>-0",
"SIP/1003|10|TtwW") in new
stack
Dec 6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No
translator path exists for channel type SIP (native 65535) to 0
Dec 6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to
create channel of type 'SIP' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("CAPI/BRI1/<MSN2>-0",
"s-CHANUNAVAIL|1") in new
stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
-- Executing Goto("CAPI/BRI1/<MSN2>-0",
"s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing Answer("CAPI/BRI1/<MSN2>-0", "") in
new stack
== BRI1: Answering for 703241494
-- Executing Wait("CAPI/BRI1/<MSN2>-0", "1") in
new stack
Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our
native format has changed to unknown
Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our
native format has changed to unknown
[snipped tons more of these]
Dec 6 02:30:48 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our
native format has changed to unknown
-- Executing VoiceMail("CAPI/BRI1/<MSN2>",
"u1003") in new stack
Dec 6 02:30:48 WARNING[28889]: channel.c:2313 set_format: Unable to
find a codec translation path from unknown to gsm
Dec 6 02:30:48 WARNING[28889]: file.c:820 ast_streamfile: Unable to
open vm-theperson (format unknown): No such file or directory
== BRI1: CAPI Hangingup
> CAPI INFO 0x3490: Normal call clearing
Hello Patrick,
I have an Eicon Diva PRI-30M card and use the Eicon Linux drivers with
chan_capi_cm.
I am able to do ISDN to SIP calls with this.
Have you tried using the Eicon drivers instead, rather than zaptel and zib pri.
Instruction for doing this can be found here:
http://www.voip-info.org/wiki/view/Asterisk+Eicon+Diva+CAPI+ISDN
I hope this helps
Thanks David
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Patrick
Sent: 06 December 2005 01:40
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk on PPC & chan_capi issue
Hi all,
I have a PPC box (IBM RS6000 43P-150, bigendian afaik) which runs Fedora
Core 5 Test1 and zaptel, libpri and asterisk 1.2.0. I also installed
chan_capi (0.6.1) so I can use my Eicon Diva Server BRI card. Asterisk
was compiled with DEBUG=-g and DEBUG_THREADS = -DDUMP_SCHEDULER
-DDEBUG_SCHEDULER-DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS. Next I
did make clean, make valgrind, make install. Asterisk runs as user/group
asterisk/asterisk.
SIP <--> SIP calls are fine, Calls from SIP out to the PSTN via
CAPI/ISDN are fine too. ISDN/CAPI --> SIP calls don't work. Example
output of the issue is below. Anyone have an idea how I fix this?
Thanks and regards,
Patrick
chan_capi registers fine:
**********************************************************************
[chan_capi.so] => (Common ISDN API for Asterisk)
== This box has 1 capi controller(s).
== Reading config for BRI1
-- ast_capi_pvt BRI1-pseudo-D (<MSN1>,<MSN2>,capi-in,0,2)
(1,4,128)
-- ast_capi_pvt BRI1 (<MSN1>,<MSN2>,capi-in,0,2) (1,4,128)
-- ast_capi_pvt BRI1 (<MSN1>,<MSN2>,capi-in,0,2) (1,4,128)
-- listening on contr1 CIPmask = 0x1fff03ff
== Registered channel type 'CAPI' (Common ISDN API Driver ($Revision:
1.115 $) )
== Registered application 'capiCommand'
== Registered custom function VANITYNUMBER
Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2):
**********************************************************************
== BRI1: Incoming call '<my GSM>' -> '<MSN2>'
-- Executing Macro("CAPI/BRI1/<MSN2>-0",
"stdexten|1003|SIP/1003")
in new stack
-- Executing Dial("CAPI/BRI1/<MSN2>-0",
"SIP/1003|10|TtwW") in new
stack
Dec 6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No
translator path exists for channel type SIP (native 65535) to 0
Dec 6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to
create channel of type 'SIP' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("CAPI/BRI1/<MSN2>-0",
"s-CHANUNAVAIL|1") in new
stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
-- Executing Goto("CAPI/BRI1/<MSN2>-0",
"s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing Answer("CAPI/BRI1/<MSN2>-0", "") in
new stack
== BRI1: Answering for 703241494
-- Executing Wait("CAPI/BRI1/<MSN2>-0", "1") in
new stack
Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our
native format has changed to unknown
Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our
native format has changed to unknown
[snipped tons more of these]
Dec 6 02:30:48 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our
native format has changed to unknown
-- Executing VoiceMail("CAPI/BRI1/<MSN2>",
"u1003") in new stack
Dec 6 02:30:48 WARNING[28889]: channel.c:2313 set_format: Unable to
find a codec translation path from unknown to gsm
Dec 6 02:30:48 WARNING[28889]: file.c:820 ast_streamfile: Unable to
open vm-theperson (format unknown): No such file or directory
== BRI1: CAPI Hangingup
> CAPI INFO 0x3490: Normal call clearing
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Sorry Patrick, I was mistaken here. The Diva Server for Linux drives currently only support Little Endian machines. Unfortunately the PPC based chipsets use Big Endian. There is a discussion about this here: http://www.cs.umass.edu/~verts/cs32/endian.html Thanks David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Patrick Sent: 08 December 2005 21:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on PPC & chan_capi issue On Thu, 2005-12-08 at 08:47 +0000, David Waugh wrote:> Hello Patrick, > > I have an Eicon Diva PRI-30M card and use the Eicon Linux drivers with chan_capi_cm. > I am able to do ISDN to SIP calls with this. > > Have you tried using the Eicon drivers instead, rather than zaptel and zib pri. > > Instruction for doing this can be found here: > http://www.voip-info.org/wiki/view/Asterisk+Eicon+Diva+CAPI+ISDN > > I hope this helpUnfortunately not. The config I described below works fine on an Intel server. The box showing the issue is a PPC box. The author of chan_capi-cm doesn't think there is anything wrong with chan_capi-cm on PPC and the way it behaves in the logfiles below. So the problem seems to be Asterisk & PPC related but I have no idea how to track down the issue or solve it. Regards, Patrick> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Patrick > Sent: 06 December 2005 01:40 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Asterisk on PPC & chan_capi issue > > > Hi all, > > I have a PPC box (IBM RS6000 43P-150, bigendian afaik) which runs Fedora > Core 5 Test1 and zaptel, libpri and asterisk 1.2.0. I also installed > chan_capi (0.6.1) so I can use my Eicon Diva Server BRI card. Asterisk > was compiled with DEBUG=-g and DEBUG_THREADS = -DDUMP_SCHEDULER > -DDEBUG_SCHEDULER-DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS. Next I > did make clean, make valgrind, make install. Asterisk runs as user/group > asterisk/asterisk. > > SIP <--> SIP calls are fine, Calls from SIP out to the PSTN via > CAPI/ISDN are fine too. ISDN/CAPI --> SIP calls don't work. Example > output of the issue is below. Anyone have an idea how I fix this? > > Thanks and regards, > Patrick > > > chan_capi registers fine: > ********************************************************************** > [chan_capi.so] => (Common ISDN API for Asterisk) > == This box has 1 capi controller(s). > == Reading config for BRI1 > -- ast_capi_pvt BRI1-pseudo-D (<MSN1>,<MSN2>,capi-in,0,2) (1,4,128) > -- ast_capi_pvt BRI1 (<MSN1>,<MSN2>,capi-in,0,2) (1,4,128) > -- ast_capi_pvt BRI1 (<MSN1>,<MSN2>,capi-in,0,2) (1,4,128) > -- listening on contr1 CIPmask = 0x1fff03ff > == Registered channel type 'CAPI' (Common ISDN API Driver ($Revision: > 1.115 $) ) > == Registered application 'capiCommand' > == Registered custom function VANITYNUMBER > > Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2): > ********************************************************************** > == BRI1: Incoming call '<my GSM>' -> '<MSN2>' > > -- Executing Macro("CAPI/BRI1/<MSN2>-0", "stdexten|1003|SIP/1003") > in new stack > -- Executing Dial("CAPI/BRI1/<MSN2>-0", "SIP/1003|10|TtwW") in new > stack > Dec 6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No > translator path exists for channel type SIP (native 65535) to 0 > Dec 6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to > create channel of type 'SIP' (cause 0 - Unknown) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing Goto("CAPI/BRI1/<MSN2>-0", "s-CHANUNAVAIL|1") in new > stack > -- Goto (macro-stdexten,s-CHANUNAVAIL,1) > -- Executing Goto("CAPI/BRI1/<MSN2>-0", "s-NOANSWER|1") in new stack > -- Goto (macro-stdexten,s-NOANSWER,1) > -- Executing Answer("CAPI/BRI1/<MSN2>-0", "") in new stack > == BRI1: Answering for 703241494 > -- Executing Wait("CAPI/BRI1/<MSN2>-0", "1") in new stack > Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping > incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our > native format has changed to unknown > Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping > incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our > native format has changed to unknown > > [snipped tons more of these] > > Dec 6 02:30:48 NOTICE[28889]: channel.c:1893 ast_read: Dropping > incompatible voice frame on CAPI/BRI1/<MSN2>-0 of format alaw since our > native format has changed to unknown > -- Executing VoiceMail("CAPI/BRI1/<MSN2>", "u1003") in new stack > Dec 6 02:30:48 WARNING[28889]: channel.c:2313 set_format: Unable to > find a codec translation path from unknown to gsm > Dec 6 02:30:48 WARNING[28889]: file.c:820 ast_streamfile: Unable to > open vm-theperson (format unknown): No such file or directory > == BRI1: CAPI Hangingup > > CAPI INFO 0x3490: Normal call clearing > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users