Hi
i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.
Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.
h323.conf:
[general]
port = 1720
bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP
address for this machine
allow=all
extension.conf:
exten = 3298,1,Answer
exten = 3298,2,Dial(SIP/user@193.y.y.y)
If a make a call to callamanager CISCO that forward to 3298 i read in
asterisk console:
Log:
Verbosity is at least 20
-- Executing Answer("H323/ip$172.z.z.z:4836/14", "") in
new stack
-- Executing Dial("H323/ip$172.z.z.z:4836/14",
"SIP/user@193.y.y.y") in new stack
-- Called user@193.y.y.y
-- SIP/user@193.y.y.y-40455d68 is ringing
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to ulaw
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec ........
.......
translation path from g729 to slin
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to ulaw
Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
find a codec translation path from g729 to slin
Dec 15 14:45:13 WARNING[19794]: translate.c:116
ast_translator_build_path: No translator path from alaw to unknown
Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies:
Cannot build a path from g729 to slin
Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to
transmit frame type 64, while native formats is 256 (read/write 4/64)
Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to
transmit frame type 256, while native formats is 4 (read/write = 4/4)
Dec 15 14:45:13 WARNING[19794]: translate.c:116
ast_translator_build_path: No translator path from alaw to unknown
Dec 15 14:45:13 WARNING[19794]: channel.c:2752
ast_channel_make_compatible: No path to translate from
H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8)
Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to
drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible
with SIP/193.x.x.x-40455d68
== Spawn extension (default, 3298, 2) exited non-zero on
'H323/ip$172.z.z.z:4836/14'
Why? where am i wrong?
probably you haven't g729 installed in asterisk, use g711 instead, put this in h323.conf and in callmanager place asterisdk gateway in region that will use g711... disallow=all allow=alaw alternatively you can find g729 codecs binaries here: http://kvin.lv/pub/Linux/Asterisk/ nik600 wrote:> Hi > > i am trying to do the same thing: > receive a call from a cisco callmanager and forward it to a SIP user. > > Asterisk is compiled with h323 support, and is configured as a gateway > in the cisco callmanager. > > h323.conf: > [general] > port = 1720 > bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP > address for this machine > allow=all > > extension.conf: > exten = 3298,1,Answer > exten = 3298,2,Dial(SIP/user@193.y.y.y) > > If a make a call to callamanager CISCO that forward to 3298 i read in > asterisk console: > > Log: > > Verbosity is at least 20 > -- Executing Answer("H323/ip$172.z.z.z:4836/14", "") in new stack > -- Executing Dial("H323/ip$172.z.z.z:4836/14", > "SIP/user@193.y.y.y") in new stack > -- Called user@193.y.y.y > -- SIP/user@193.y.y.y-40455d68 is ringing > Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to > find a codec translation path from g729 to ulaw > Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to > find a codec ........ > ....... > translation path from g729 to slin > Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to > find a codec translation path from g729 to ulaw > Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to > find a codec translation path from g729 to slin > Dec 15 14:45:13 WARNING[19794]: translate.c:116 > ast_translator_build_path: No translator path from alaw to unknown > Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies: > Cannot build a path from g729 to slin > Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to > transmit frame type 64, while native formats is 256 (read/write > 4/64) > Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to > transmit frame type 256, while native formats is 4 (read/write = 4/4) > Dec 15 14:45:13 WARNING[19794]: translate.c:116 > ast_translator_build_path: No translator path from alaw to unknown > Dec 15 14:45:13 WARNING[19794]: channel.c:2752 > ast_channel_make_compatible: No path to translate from > H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8) > Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to > drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible > with SIP/193.x.x.x-40455d68 > == Spawn extension (default, 3298, 2) exited non-zero on > 'H323/ip$172.z.z.z:4836/14' > > Why? where am i wrong? > _______________________________________________ > --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
nik600 wrote:> Hi > > i am trying to do the same thing: > receive a call from a cisco callmanager and forward it to a SIP user. > > Asterisk is compiled with h323 support, and is configured as a gateway > in the cisco callmanager.The incoming call is in the g.729 format, you should be able to fix this in cisco call manager. If not, make sure that the SIP target can accept a g.729 call. Failing that buy a license for the codec.> > h323.conf: > [general] > port = 1720 > bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP > address for this machine > allow=all > > extension.conf: > exten = 3298,1,Answer > exten = 3298,2,Dial(SIP/user@193.y.y.y) > > If a make a call to callamanager CISCO that forward to 3298 i read in > asterisk console: > > Log: > > Verbosity is at least 20 > -- Executing Answer("H323/ip$172.z.z.z:4836/14", "") in new stack > -- Executing Dial("H323/ip$172.z.z.z:4836/14", > "SIP/user@193.y.y.y") in new stack > -- Called user@193.y.y.y > -- SIP/user@193.y.y.y-40455d68 is ringing > Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to > find a codec translation path from g729 to ulaw > Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to > find a codec ........ > ....... > translation path from g729 to slin > Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to > find a codec translation path from g729 to ulaw > Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to > find a codec translation path from g729 to slin > Dec 15 14:45:13 WARNING[19794]: translate.c:116 > ast_translator_build_path: No translator path from alaw to unknown > Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies: > Cannot build a path from g729 to slin > Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to > transmit frame type 64, while native formats is 256 (read/write > 4/64) > Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to > transmit frame type 256, while native formats is 4 (read/write = 4/4) > Dec 15 14:45:13 WARNING[19794]: translate.c:116 > ast_translator_build_path: No translator path from alaw to unknown > Dec 15 14:45:13 WARNING[19794]: channel.c:2752 > ast_channel_make_compatible: No path to translate from > H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8) > Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to > drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible > with SIP/193.x.x.x-40455d68 > == Spawn extension (default, 3298, 2) exited non-zero on > 'H323/ip$172.z.z.z:4836/14' > > Why? where am i wrong? > _______________________________________________ > --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 > >