Re: [asterisk-users] mysql call stored procedure
MYSQL(Nextresult resultid ${connid}) after MySQL(Fetch fetchid ${resultid}
pass) helped to resolve this.
you should always get all results sp produce otherwise mysql returns error.
On Tue, May 17, 2011 at 4:36 PM, Borin katerin.bo...@gmail.com wrote:
Hi Guys,
I am getting an error when executing another mysql query in dialplan after
calling stored procedure.
If calling the procedure from mysql cli it gives a result like:
mysql call call_control(78236721,1000,1233);
+--+
| pass |
+--+
|1 |
+--+
So I need asterisk to recognize this pass and take some actions based on
what the pass value is.
Dialplan looks like this:
MYSQL(Connect connid ${DBDefaultHost} ${DBuser} ${DBpass} ${DBname})
MySQL(Query resultid ${connid} CALL call_control(78236721,1000,1233))
MySQL(Fetch fetchid ${resultid} pass)
MYSQL(clear ${resultid})
MySQL(Query resultid ${connid} SELECT/INSERT whatever from table)
So, it gives me this pass value correct, but if I execute some other query
INSERT or SELECT after clearing the result, it gives me an error
[May 17 16:16:13] WARNING[19572]: app_addon_sql_mysql.c:374 aMYSQL_query:
aMYSQL_query: mysql_query failed. Error: Commands out of sync; you can't run
this command now
The error disappears if I reconnect to mysql after calling the stored
procedure but it seams not right to me to connect to mysql 2 times for 1
call.
Did anyone have the same issue?
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[asterisk-users] mysql call stored procedure
Hi Guys,
I am getting an error when executing another mysql query in dialplan after
calling stored procedure.
If calling the procedure from mysql cli it gives a result like:
mysql call call_control(78236721,1000,1233);
+--+
| pass |
+--+
|1 |
+--+
So I need asterisk to recognize this pass and take some actions based on
what the pass value is.
Dialplan looks like this:
MYSQL(Connect connid ${DBDefaultHost} ${DBuser} ${DBpass} ${DBname})
MySQL(Query resultid ${connid} CALL call_control(78236721,1000,1233))
MySQL(Fetch fetchid ${resultid} pass)
MYSQL(clear ${resultid})
MySQL(Query resultid ${connid} SELECT/INSERT whatever from table)
So, it gives me this pass value correct, but if I execute some other query
INSERT or SELECT after clearing the result, it gives me an error
[May 17 16:16:13] WARNING[19572]: app_addon_sql_mysql.c:374 aMYSQL_query:
aMYSQL_query: mysql_query failed. Error: Commands out of sync; you can't run
this command now
The error disappears if I reconnect to mysql after calling the stored
procedure but it seams not right to me to connect to mysql 2 times for 1
call.
Did anyone have the same issue?
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Re: [asterisk-users] extensions.lua with luasql.mysql.
Hi
try this pls
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18671
it did help to me
On Thu, Feb 24, 2011 at 9:31 PM, Rodrigo Lang rodrigoferreiral...@gmail.com
wrote:
Hi to all!
I'm trying to create a context for integration with extensions.lua and
libsql.mysql, but I'm not getting to run. When I reload the module
pbx_lua.so the following error appears:
[Feb 24 16:59:29] ERROR[30749]: pbx_lua.c:1249 exec: Error executing lua
extension: error loading module 'luasql.mysql' from file
'/usr/lib/lua/5.1/luasql/mysql.so':
/usr/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_getfield
stack traceback:
[C]: ?
[C]: in function 'require'
[string extensions.lua]:205: in function [string
extensions.lua]:204
I tested my script with a file.lua and works ok and the extensions.lua
works fine too. My extensions.lua:
extensions = {
luatest = {
[302] = function()
require(luasql.mysql)
app.Answer()
app.Log(NOTICE, Trying to connect in MySQL)
app.Wait(2)
env = assert(luasql.mysql())
sql = assert
(env:connect(asterisk_teste,root,*,localhost,3306))
sel = sql:execute('SELECT * FROM cdr;')
sel:fetch(Fetcharray)
app.Noop(Fetcharray[1])
end;
h = function()
app.Hangup()
end;
};
}
Does anyone know what is happening?
Thansk in advance,
--
Rodrigo Lang
Opening your mind - Just another Open Source
sitehttp://openingyourmind.wordpress.com/
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Re: [asterisk-users] lua -asterisk manual
Hi again
Could anybody pls share some thoughts about dialplan in lua? I mean some say
it works faster...I have tested my dialplan with pbx_config
(extensions.conf) , then with ael. Dialplan is not very complex (just some
selects in mysql, then based on select some if, then...etc) I think it is
just easier to use some script language to program it.
Does anyone know any drawbacks for using lua?
Does it work stable with asterisk?
And for example why lua and not just ael? Ael seams also being convenient.
On Fri, Feb 18, 2011 at 3:51 PM, Borin katerin.bo...@gmail.com wrote:
Pls could you share some lua config which contains mysql quires
On Fri, Feb 18, 2011 at 3:38 PM, Faisal Hanif fai...@vopium.com wrote:
The only specific you need to do in extensions.lua is create a table to
put your extensions in like,
Extension{
}
Else all will be LUA code and all asterisk applications can be called as
app.application_name.
Regards,
Faisal Hanif
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
*Sent:* Friday, February 18, 2011 4:33 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] lua -asterisk manual
Please could someone advise good manual for using lua for asterisk
dialplan. There is not much docu about it.
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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
thanks a lot. that was a problem. On Fri, Feb 18, 2011 at 8:44 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Friday 18 February 2011 05:29:56 Borin wrote: Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Make sure res_ael_share.so is loaded first. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Regards, Kate -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lua -asterisk manual
Please could someone advise good manual for using lua for asterisk dialplan. There is not much docu about it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif fai...@vopium.com wrote: Did you checked if you extension.ael doesn’t have syntax error? I think there is no error. I loaded the standard ael first (provided by asterisk) then my test config, got the same result. Did you upgraded anything after last compile? No. I just took ver 1.6.2.16.1 , compiled with ael support got this error. then decided to check with ver 1.8.2. Error remained the same. Or Try a clean recompile Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin *Sent:* Friday, February 18, 2011 4:30 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2 Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Regards, Kate -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] lua -asterisk manual
Pls could you share some lua config which contains mysql quires
On Fri, Feb 18, 2011 at 3:38 PM, Faisal Hanif fai...@vopium.com wrote:
The only specific you need to do in extensions.lua is create a table to put
your extensions in like,
Extension{
}
Else all will be LUA code and all asterisk applications can be called as
app.application_name.
Regards,
Faisal Hanif
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin
*Sent:* Friday, February 18, 2011 4:33 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] lua -asterisk manual
Please could someone advise good manual for using lua for asterisk
dialplan. There is not much docu about it.
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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
Linux version 2.6.26-2-686 (Debian 2.6.26-24) (da...@debian.org) (gcc version 4.1.3 20080704 (prerelease) (Debian 4.1.2-25)) On Fri, Feb 18, 2011 at 4:09 PM, Faisal Hanif fai...@vopium.com wrote: Are you on CentOS? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin *Sent:* Friday, February 18, 2011 7:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2 On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif fai...@vopium.com wrote: Did you checked if you extension.ael doesn’t have syntax error? I think there is no error. I loaded the standard ael first (provided by asterisk) then my test config, got the same result. Did you upgraded anything after last compile? No. I just took ver 1.6.2.16.1 , compiled with ael support got this error. then decided to check with ver 1.8.2. Error remained the same. Or Try a clean recompile Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin *Sent:* Friday, February 18, 2011 4:30 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2 Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Regards, Kate -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
The error appears only if I load module. There is no warning during installation, so module pbx_ael.so is compiled and placed in modules dir of asterisk On Fri, Feb 18, 2011 at 4:15 PM, Borin katerin.bo...@gmail.com wrote: Linux version 2.6.26-2-686 (Debian 2.6.26-24) (da...@debian.org) (gcc version 4.1.3 20080704 (prerelease) (Debian 4.1.2-25)) On Fri, Feb 18, 2011 at 4:09 PM, Faisal Hanif fai...@vopium.com wrote: Are you on CentOS? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin *Sent:* Friday, February 18, 2011 7:46 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2 On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif fai...@vopium.com wrote: Did you checked if you extension.ael doesn’t have syntax error? I think there is no error. I loaded the standard ael first (provided by asterisk) then my test config, got the same result. Did you upgraded anything after last compile? No. I just took ver 1.6.2.16.1 , compiled with ael support got this error. then decided to check with ver 1.8.2. Error remained the same. Or Try a clean recompile Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Borin *Sent:* Friday, February 18, 2011 4:30 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2 Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Regards, Kate -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound call leg CALLID
Solved it with returning from Dial to Macro
exten = _X.,1,NoOp(SIPID1 = ${SIPCALLID})
exten = _X.,n,Dial(SIP/${EXTEN}@some_IP,120,M(to))
exten = h,1,NoOp
[macro-to]
exten = s,1,NoOp(${DIALSTATUS},1)
exten = s,n,Set(SIPID2=${SIPCALLID})
exten = s,n,Set(CDR(sipid2)=${SIPID2})
exten = s,n,Set(CDR(userfield)=${SIPID2})
SIPID2 will provide Call ID for outbound call leg. I also tryed
${BRIDGEPEER} and ${BRIDGEPVTCALLID} but they don't show anything in my
case.
On Tue, Feb 15, 2011 at 3:15 PM, Borin katerin.bo...@gmail.com wrote:
Hello everyone
Is there a possibility to catch an outbound callleg ID for the follovong
scenario: some carrier - --(asterisk1) asterisk2 ?
I can get inbound callid for asterisk1 with a ${SIPCALLID} in
extensions.conf or to look it up in cdrs field (are the same). But how about
outbound? I have all calls just forwarded through asterisk1, not answered
and for every call I have 1 cdr record with only inbound call leg callid.
I can see all the call legs with rasterisk -x sip show channels, but I
would like to get the data in cdrs as well. Does anyone have any idea? I
have asterisk 1.6.2 and I don't want to fork cdrs, just another field for
the came call.
Regards, Kate
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[asterisk-users] outbound call leg CALLID
Hello everyone
Is there a possibility to catch an outbound callleg ID for the follovong
scenario: some carrier - --(asterisk1) asterisk2 ?
I can get inbound callid for asterisk1 with a ${SIPCALLID} in
extensions.conf or to look it up in cdrs field (are the same). But how about
outbound? I have all calls just forwarded through asterisk1, not answered
and for every call I have 1 cdr record with only inbound call leg callid.
I can see all the call legs with rasterisk -x sip show channels, but I
would like to get the data in cdrs as well. Does anyone have any idea? I
have asterisk 1.6.2 and I don't want to fork cdrs, just another field for
the came call.
Regards, Kate
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[asterisk-users] how to fake asterisk register?
Hello, I am wondering how I can make asterisk think that the user is registered on it..Scheme is the following: user registerkamailio (put data in location table)--asterisk All users are registered on kamailio and I want to duplicate that info on asterisk. If a call comes to kamailio it will be forwarded to asterisk and then to another user (registered on kamailio and I want register info be on asterisk too). So I gather the info from location table on kamailio and put it in ASTDB, so register info is there. But asterisk still thinks that the user is not registered. in astdb is the following about the user: /SIP/Registry/1002200 : 10.1.2.82:1024 :3600:1002200:sip:1002...@10.1.2.82:1024;line=xd932iq7 Where asterisk stores the all info about registered users and how can I fake it? Any help will be appreciated :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REFER method
Hello, Is there a possibility for asterisk to work out REFER messages in the dialplan? like INVITES. I need it to distinguish forwarded calls from all the other calls. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf payload 100
Probably has anyone idea how dtmf payload type could be changed in Asterisk say to 100? On Wed, Mar 10, 2010 at 2:53 PM, Katerina Borin katerin.bo...@gmail.comwrote: Hello, I encountered the dtmf problem between my asterisk box (1.4.23) and suppliers gateway (unknown vendor). I have dtmf mode set to rfc2833 and it alway worked till supplier has changed something. Now I receive from him dtmf payload 100. With the second supplier which sends dtmf with payload type 101 everything works. in cli I get this message as dtmf is entered rtp.c:1287 ast_rtp_read: Unknown RTP codec 100 received from 'suppliers IP' Is there any way to get asterisk understand dtmf payload type 100? Regards, Katerina -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf payload 100
Hello, I encountered the dtmf problem between my asterisk box (1.4.23) and suppliers gateway (unknown vendor). I have dtmf mode set to rfc2833 and it alway worked till supplier has changed something. Now I receive from him dtmf payload 100. With the second supplier which sends dtmf with payload type 101 everything works. in cli I get this message as dtmf is entered rtp.c:1287 ast_rtp_read: Unknown RTP codec 100 received from 'suppliers IP' Is there any way to get asterisk understand dtmf payload type 100? Regards, Katerina -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transcoding with TC400P
Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [7.590966] Zaptel Version: 1.4.12.1 [7.590966] Zaptel Echo Canceller: MG2 [7.610963] zttranscode: Loaded. [7.618969] wctc4xxp: tc400b0: Attached to device at :02:07.0. [7.618969] firmware: requesting zaptel-fw-tc400m.bin [8.86] ACPI: PCI Interrupt :02:07.0[A] - GSI 19 (level, low) - IRQ 19 [ 11.400359] wctc4xxp: tc400b0: (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) [ 11.400359] wctc4xxp: tc400b0: Installed a Wildcard TC: Wildcard TC400P+TC400M [ 11.400359] zttranscode: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) [ 11.400359] zttranscode: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) [ 11.572355] ACPI: PCI Interrupt :00:1f.5[B] - GSI 17 (level, low) - IRQ 17 in asterisk cli: Connected to Asterisk 1.4.23.1 currently running on katerin (pid = 3168) Verbosity was 3 and is now 5 katerin*CLI transcoder show 0/0 encoders/decoders of 92 channels are in use. I am trying to do 2 experiments: 1) asterisk1 ---g729-- asterisk2 (with transcoding card) Playfile in ulaw format 2) asterisk2 ---ulaw-- asterisk2 (with transcoding card) Playfile in g729 format In the first case I get all calls proceeding and in asteris2 cli Connected to Asterisk 1.4.23.1 currently running on katerin (pid = 3168) Verbosity was 3 and is now 5 katerin*CLI transcoder show 90/0 encoders/decoders of 92 channels are in use. But it all does not work for the second case, in asterisk2 cli I get messages like [Feb 19 15:18:32] ERROR[3121]: codec_dahdi.c:244 zap_translate: Unable to attach to transcoder: Input/output error [Feb 19 15:18:32] WARNING[3121]: translate.c:294 ast_translator_build_path: Failed to build translator step from 8 to 2 [Feb 19 15:18:32] WARNING[3121]: chan_sip.c:3751 sip_write: Asked to transmit frame type 256, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x100 (g729)(256) Btw call does go through but transcoding is done by processor not by TC400P. Did anyone encounter such problem? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
