AstUnicall for Asterisk 1.4.18 is now available in downloads.
- users now can set the calling party category request from the dial plan and from the unicall.conf file (more info in the included unicall.conf)
- added a README file with instructions to install.
- added a sample unicall.conf with comments about how to use the UC_CATEGORY dial plan variable and the new category parameter for unicall.conf
- removed menuselect.makeopts ( Clod Patry, thanks for the heads up )
- fixed a bug with the usage count of the chan_unicall.so module
The driver for Asterisk 1.6 is also about to be released someday the next week (yeah right …)
Hy Moy,
I downloaded this last realease. Untar the files on /usr/src and started to install them. I compiled all packages, but on Asterisk I got ´no make file´. No way to compile Asterisik. So, did I miss something?
BTW, the Astunicall 1.4.16-1 worked like a charm. Keep the good job!
Luc
Hum … sorry if I ask something too obvious, but did you run the ./configure script? ./configure finishes just fine just no Makefile result after running it?
I will test in a fresh box tonight and see if I have the same issue.
Thanks!
Da, my mistake, will update the tar in a couple of minutes. I got too excited when packaging and I just removed the Makefile from all the projects, Asterisk is a special case, it requires the Makefile from the very beginning.
Luc. Is fixed now. Thanks for the heads up
Hy Moy,
I downloaded this last realease, compiled without any problem, but when asterisk starts, it dies whitjh:
/usr/sbin/asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_start
any clues?
Hello nb71,
Please re-install the version of libunicall. I think you may have an old libunicall version where uc_start was not defined, tho I am not sure if old versions define it or not.
execute objdump -T /usr/lib/libunicall.so | grep uc_start
That should show if the libunicall you have installed has the uc_start routine defined. If not, then is not the proper libunicall version, the one I included in astunicall has it defined for sure.
moy,
you are right, I’ve just found an old unicall wich was left from a previous install.
I’m sorry an Thanks!
Moy:
Thanks by this wonderful job.
Thanks a lot.
Moy:
I haved installed a server with AstUnicall for Asterisk 1.4.18 y Digium card Tarjeta TE120P with a echo in some calls in and out.. I migrate for a TE122P with echo cancelation, echo goes away, but now the calls the voice dropped. In Asterisk list suggest me to to install ZAPTEL branches…
My question can i go direct to recompile ZAPTEL or am going to have problems with UNICALL?.
Thanks
Ruben
There should be no problem, I have worked with latest zaptel branch during my development and has not caused issues.
Moy
`
I installed the AstUnicall… replaced the Zaptel with the Zaptel branches…compiled everything….But when i start the Asterisk i cant see the UC show channels. No erros in the compiled.
Ruben
Moy
I all ready installed Zaptel 1.4.10 with Unicall. I work ok.
I reinstalled with a TE122B with echo cancelation. I still have echo.
Digium Support told me that maybe echo cancelation(VPMADT032) doesnt work with Unicall.
What to you think?
Moy
Really the problem is not with UNICALL.
With the TE122B with echo cancelation, we remove in zapata.conf
echocancelwhenbridged=yes and the echo goes away.
Thanks
Moy
If i modified in the unicall.conf the echotraining=no takes effect in voice?
or if a modified rxgain or txgain take efect in the voice?
My question is because i all ready didìt in the zapata.conf.
moy thanls for your excellent work, i’ve got a silly question but maybe you can help or guide me. Is it possible to use app_rfax and app_tfax with unicall?
is even possible to get faxing via E1?
can you point me to some docs about it?
thanks
Hello Seaq, I have not tried, so I cannot give an educated opinion about those fax applications. I suppose they analyze the audio, then my guess is that it should work, since they should be agnostic of how the audio gets in an out. If you ever try them, please report feedback here so all we can know
hi moy, i’m testing still without success, but as soon as i’ve got somethin i’ll let you now.
Hi it seems to be working now!!!
i’m using agx-ast-addons
http://sourceforge.net/projects/agx-ast-addons/
Still i’ve got more testing to do, but at least i’ve received 3 test faxes at this point !!
nice! , thanks for the feedback!
Hi moy i’m having a kernel panic error with astunicall-1418. I’m downgrading right now to check with 1416. I’m thinking about testing with your zaptel patches..
anyway i’m having kernel panic not syncing errors. one with libata, the other with smp core….
almost all of them are related with a disabling echo canceller on chan_unicall message at the asterisk log.
I’m using centos5 with 2.6.18-53.1.14.el5 kerrnel
seaq: yay, don’t know what to say to you. Not my fault
.. that sounds like a bug in zaptel and probably you should report it. If you have time/resources to test Asterisk 1.6 with my chan_zap R2 stuff, I’d be glad to see your feedback in the bugtracker.
yeap that’s what i’ve thought … but with the downgrade it worked without problems!
so it seems to be an issue with 1.4.18.
I’ll try to test the new chanzap stuff!!
Hi Moy,, Im having a little trouble with this unicall. I installed Unicall with the following running on CentOs 5:
asterisk 1.14.18
zaptel 1.4.9.2
spandsp 0.0.4
libmfcr2-0.0.3
libsupertone-0.0.2
libunicall-0.0.3
its working, but sometimes during the call, the call drops, some times after 3,4 minutes, some other times after 1 minute,, it varies. I talked with the people of Telmex to verify what was going on, they says its a problem with the signaling. they sent me their traces where show (still dont understand the trace) this is the last part where the call dropped
telmex trace
4 MINUTES
09510 0052670 CLEAR FORWARD S 0000 1001 H’40 BL3DM-3586
09540 0000030 R 0000 1001 H’40 BL3DM-3586
09760 0000220 R 0000 1001 H’35 BL3DM-3586
09760 0000000 IDLE S 0000 1001 H’35 BL3DM-3586
RECORDING STOPPED PERMANENTLY BY OPERATOR
END
this is what ive got on my unicall.conf
[channels]
language=sp
hidecallerid=no
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=no
rxgain=0
txgain=0
callgroup=1
pickupgroup=1
immediate=no
musiconhold=default
loglevel=0
protocolclass=mfcr2
protocolvariant=mx,30,4
protocolend=co
group=1
loadzone=mx
defaultzone=mx
context=entradae1
channel=1-15
i have doubts about these parameters
protocolvariant
protocolend
is there a table with values to put?,, or how do i now what values to put in this parameters?
i will really appreciate your help, i have with this for over 1 week, and its getting anoying getting the calls drop. or if you have a document with this kind information about the unicall configuration, or table with parameters. that will help a lot.
thanks for your time,
ron
I suggest you to read this: http://www.moythreads.com/unicall/mfcr2-asterisk-unicall-0.2-english.pdf
I never had any issues with Telmex R2 signaling, tho its possible now they are sending metering pulses, if so, the library needs to be adjusted. Ask them if they send metering pulses or billing pulses.
Hello moy, I am trying to compile asterisk and when I write the make instruction I get this:
[LD] chan_mgcp.o -> chan_mgcp.so
[LD] gentone.c -> gentone
./gentone busy 480 620
Wavelength 1 (in samples): 16.66667
Minimum samples (1): 50 (3.000000.3 wavelengths)
Wavelength 1 (in samples): 12.90323
Minimum samples (1): 400 (31.000000.3 wavelengths)
Need 400 samples
Wrote busy.h
./gentone ringtone 440 480
Wavelength 1 (in samples): 18.18182
Minimum samples (1): 200 (11.000000.3 wavelengths)
Wavelength 1 (in samples): 16.66667
Minimum samples (1): 50 (3.000000.3 wavelengths)
Need 200 samples
Wrote ringtone.h
[CC] chan_oss.c -> chan_oss.o
[LD] chan_oss.o -> chan_oss.so
[CC] chan_phone.c -> chan_phone.o
In file included from /usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/cdr.h:49,
from /usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/channel.h:115,
from chan_phone.c:57:
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:336: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_mallocâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:336: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_mallocâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:359: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_callocâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:359: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_callocâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:395: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_reallocâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:395: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_reallocâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:422: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_strdupâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:422: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_strdupâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:451: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_strndupâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/utils.h:451: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â_ast_strndupâ
In file included from chan_phone.c:57:
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/channel.h:1025: warning: no previous prototype for âast_cause2strâ
/usr/src/astunicall-1.4.18-0.2/asterisk-1.4.18/include/asterisk/channel.h: In function âast_cause2strâ:
and the error keeps going up to this:
DialTone.h:73: warning: (near initialization for âDialToneâ)
DialTone.h:73: warning: excess elements in scalar initializer
DialTone.h:73: warning: (near initialization for âDialToneâ)
DialTone.h:73: warning: excess elements in scalar initializer
DialTone.h:73: warning: (near initialization for âDialToneâ)
DialTone.h:73: warning: excess elements in scalar initializer
DialTone.h:73: warning: (near initialization for âDialToneâ)
DialTone.h:73: warning: excess elements in scalar initializer
DialTone.h:73: warning: (near initialization for âDialToneâ)
DialTone.h:73: warning: excess elements in scalar initializer
make[1]: *** wait: No child processes. Stop.
make[1]: *** Waiting for unfinished jobs….
make[1]: *** wait: No child processes. Stop.
make: *** [channels] Error 2
and the compilation stops. What do you think its happening?
Thanks a lot.
Hello Moy,
I successfully installed the package for asterisk and unicall 1.4.19 and worked perfectly. I performed successful tests with a R2 Link and everything is Ok.
However, when I started to test other functions of the PBX, I realized that the Queue Application was not working properly. When calls are set to go configured queues, I have a warning message “WARNING[10105]: app_queue.c:3939 queue_exec: Unable to join queue ’0800′”, and the call is hang up.
Then, after executing the command “show queues”, I had a unknown status of the agents (Invalid).
I then manually reload app_queue, and it started to work properly.
However, when I restart asterisk it goes back to the same condition.
Do you think it has something to do with Real-time?? If so, I won’t use realtime, is there a way to uninstall it???
Best Regards and thanks in advance,
Marco Cordeiro – Brazil
Sorry, version is 1.4.18 no .19 as mentioned before!!!
ggarcia: honestly, I don’t have a clue.
mhcordeiro: that’s kind of a generic question that you should redirect to asterisk-users mailing list, it has nothing to do with Unicall itself
Thanks moy, I figured it out, i dont know exactly what it was but upgraded to the lates version of fedora and it compiled just fine. It was an asterisk problem, not a unicall problem.
Another question, im in mexico and im using a E1 from telmex, i keep getting Unknown for callerid, I called them and they said that I need to set it up in my pbx because they always send the callerid signaling. How can I set up callerid?
Thanks.
protocolvariant=mx,10,4,7
the second comma separated parameter (10) is the amount of callerid digits you expect
Hola:
En la forma que está el comprimido,
¿Habría que recompilar completamente el asterisk?
O
¿Sería como pasar un patch sin modificar el asterisk original?
Mis preguntas es teniendo en cuenta a los Asterisks pre-instalado en un AsteriskNOW, Elastix, Trixbox, etc..
en este mundo gabachizado no vas a llegar muy lejos en cuanto a tecnología se refiere sin inglés, por ello este sitio está en inglés, y porque gente de brasil u otros paises no hispano-parlantes nos visitan. Si puedes escribir inglés por favor en un futuro escribe en inglés en este blog, si no, pues ni hablar asi está bien
La respuesta corta a tu pregunta es, si, recompilar completamente el Asterisk. Es la forma mas sencilla de hacerlo. De otro modo, requieres conocimientos de C y ganas de investigar, puesto que no he escrito ningun tutorial de como hacerlo con Asterisk pre-instalados.
I already know it for the next occasion. I wrote the commentary thinking about you (Hispanic loudspeaker) and not in the community.
Thanks for its correction.
Hi moy,
I’m having a weird problem, it might be caused by timing issues somewhere, we’re testing the platform so we have someone calling out hanging up when the call connectes and start it over again, out of 30 calls 4 failed, what i found is that the ones that connected correctly had MFC/R2 UniCall/1 0101 -> [1/ACCEPTED/Answer /Accepted Paid] whereas the ones that failed had MFC/R2 UniCall/1
Weird that was cut off,
I got a protocol analyzer to try and see what could be wrong.
this is a non working call
0032.658 A-3 B Adr cmp B
0032.808 II-1 F Sub w/o p
0032.878 B-6 B Sub LFC
0045.827 1001 F ClrForwd
0045.867 1001 Idle
this is a working call
0021.178 A-3 B Adr cmp B
0021.328 II-1 F Sub w/o p
0021.398 B-6 B Sub LFC
0026.477 0101 B Answered
0030.947 1001 F ClrForwd
0030.967 1001 Idle
on the non working call ringing stopped and audio got connected then the call was hung up. i see that astersik/unicall never sent 0101 to the remote R2 so probably that end timed out and sent the 1001 back to asterisk/unicall
hello Moy.
i downloaded and built astunicall-1.4.18-0.2 on Debian Etch (AMD64). when i got to
building asterisk. i got “relocation R_X86_64_32 against `a local symbol’ can not be used when making a shared object; recompile with -fPIC”. so i went ahead and put “-fPIC” in CFLAG. it seems to build fine afterward. but when i start Asterisk. i got segfault right when it load chan_unicall.so. any thoughts?
I would need the full logs to be able to tell.
jsolares: “I would need the full logs to be able to tell” was meant for you.
edwin.Iam: I just discussed this bug in my blog at http://www.moythreads.com/wordpress/2008/05/25/a-tale-of-two-bugs/ , the first bug I mention is the bug you are facing, you can find the answer there. It’s just a typo in a Unicall header that is causing the crash.
thanks Moy. i changed the uc_statet2str to uc_state2str in unicall.h, that seems to work.
now the phone company i connect to uses Uruguay R2 variant. i’ve noticed there’s no Uruguay in mfcr2.c. would that compatible with ITU? if not. is there any pointers on how do i go about customizing it?
I have no experience with Uruguay variant, try ITU, if that does not work let me know and probably we can work together to add Uruguay support.
Hello!
In the past I successfuly compiled and use your “Unicall 0.0.5 & spandsp 0.0.4 with Asterisk 1.4.9″ pack. Now I tried this 1.4.18 version. I compile and install the included zaptel (1.4.9.2) and spandsp (0.0.4) with no problems, but when I try to compile libmfcr2-0.0.3, it bombs out with error:
# make
make all-am
make[1]: Entering directory `/home/teste/asterisk_unicall/astunicall-1.4.18-0.2/unicall-0.0.5pre1/libmfcr2-0.0.3′
if /bin/sh ./libtool –tag=CC –mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I/usr/include/libxml2 -g -O2 -MT mfcr2.lo -MD -MP -MF “.deps/mfcr2.Tpo” -c -o mfcr2.lo mfcr2.c; \
then mv -f “.deps/mfcr2.Tpo” “.deps/mfcr2.Plo”; else rm -f “.deps/mfcr2.Tpo”; exit 1; fi
gcc -DHAVE_CONFIG_H -I. -I. -I. -I/usr/include/libxml2 -g -O2 -MT mfcr2.lo -MD -MP -MF .deps/mfcr2.Tpo -c mfcr2.c -fPIC -DPIC -o .libs/mfcr2.o
mfcr2.c:109: error: syntax error before “delete_context”
mfcr2.c:109: warning: data definition has no type or storage class
mfcr2.c:191: error: initializer element is not constant
mfcr2.c:191: error: (near initialization for `protocol_descriptor.xx_call_control’)
mfcr2.c:192: warning: initialization from incompatible pointer type
mfcr2.c:193: warning: initialization from incompatible pointer type
mfcr2.c:194: warning: initialization from incompatible pointer type
mfcr2.c:195: warning: initialization from incompatible pointer type
mfcr2.c:196: warning: initialization from incompatible pointer type
mfcr2.c:197: warning: initialization from incompatible pointer type
mfcr2.c:198: warning: initialization from incompatible pointer type
mfcr2.c:199: warning: initialization from incompatible pointer type
mfcr2.c:200: warning: initialization from incompatible pointer type
mfcr2.c:201: warning: initialization from incompatible pointer type
mfcr2.c:203: warning: excess elements in struct initializer
mfcr2.c:203: warning: (near initialization for `protocol_descriptor’)
mfcr2.c:4212: error: `delete_context’ redeclared as different kind of symbol
mfcr2.c:109: error: previous declaration of `delete_context’
mfcr2.c:4212: warning: `delete_context’ was declared `extern’ and later `static’
make[1]: *** [mfcr2.lo] Error 1
make[1]: Leaving directory `/home/teste/asterisk_unicall/astunicall-1.4.18-0.2/unicall-0.0.5pre1/libmfcr2-0.0.3′
make: *** [all] Error 2
Any help will be greatly appreciated…
I think you did not compile and installed the new libunicall included in the package before compiling libmfcr2. Try with that.
Hello Moy
My name is Carlos. I had a TE110P configured with Asterisk 1.2 working (E1 from Telmex) for about two years using a patch from zarzamora.com. Now I had to change the server and installed your chan unicall 1.4.18 with Trixbox 2.6.07 which basically comes with Asterisk 1.4.18 and centos 5 in a Dell Optiplex 755. Everything was smooth and didn’t have any problems (except a kernel upgrade to match zaptel). I have the server working with no problems at all. Thanks a lot for this great work. I have also installed a2billing for those insterested…. thanks
Hi Moy.
After some time we decided to upgrade our dell 2950 from asterisk 1.2 to astunicall 1.4.18, everything works well unitl our users begin to complain about cuts on calls.
After making some research we notice that even when the calls are between two local phones the calls sufers cuts after five or six minutes.
I know that is not a unicall specific problem but I just dont know what to do because i do not have this problem before the upgrade.
Any Ideas ? Some else had this problem before ?
Regards.
You should better try asterisk-users mailing list and provide a better technical description about your issue, codecs involved, if it happens just on IAX/SIP etc etc
Thanks moy , i did reinstall the asterisk in another box with centos 4.1 and seems like everything is working better.
thanks for the code.
Regards
Hi Moy ,
Since i moved the asterisk to another box everything is working fine. Just some times i geting this kind of errors :
[Sep 4 17:21:23] NOTICE[4776]: chan_unicall.c:2516 handle_uc_event: Unicall/1 event Dialing
[Sep 4 17:21:23] NOTICE[4776]: chan_unicall.c:1912 unicall_exception: Exception on 17, channel 1
[Sep 4 17:21:40] NOTICE[4776]: chan_unicall.c:2516 handle_uc_event: Unicall/1 event Protocol failure
[Sep 4 17:21:40] ERROR[4776]: chan_unicall.c:2520 handle_uc_event: Unicall/1 protocol error. Cause 32769
[Sep 4 17:21:40] WARNING[4776]: app_dial.c:743 wait_for_answer: Unable to forward voice or dtmf
This is normal situation ? carrier issue or unicall issue ?
Thanks in advance.
Hi Guys,
In this moment, I have installed Elastix-1.1-8-11ago2008.iso (with Asterisk 1.4.19) + astunicall-1.4.19-0.1.tar.gz
Asterisk is integrated with unicall successfully:
elastix*CLI> UC show channels
Channel Extension Context Status Language MusicOnHold
1 ivr-2 Idle en
2 ivr-2 Idle en
3 ivr-2 Idle en
4 ivr-2 Idle en
….
29 ivr-2 Idle en
30 ivr-2 Idle en
31 ivr-2 Idle en
elastix*CLI>
E1 channels are Ok:
# zttool
â Current Alarms: No alarms. â
â Sync Source: Internally clocked â
â IRQ Misses: 3 â
â Bipolar Viol: 0 â
â Tx/Rx Levels: 0/ 0 â
â Total/Conf/Act: 31/ 31/ 30 â
â 1111111111222222222233 ââââââââ â
â 1234567890123456789012345678901 â Back â â faces âââââââââââââââââââââââ
â TxA 111111111111111-111111111111111 ââââââââ
â TxB 000000000000000-000000000000000
â TxC 000000000000000-000000000000000
â TxD 111111111111111-111111111111111
â â
â RxA 111111111111111-111111111111111
â RxB 000000000000000-000000000000000
â RxC 000000000000000-000000000000000
â RxD 111111111111111-111111111111111
Communication between extensions, loca time, conference… and another services are ok. But, outbound and inbound calls do not work
Inside the file /var/log/asterisk/full, I can see a few errrors:
[Mar 2 10:44:24] ERROR[3201] chan_unicall.c: Unicall/13 protocol error. Cause 32771
[Mar 2 10:44:46] ERROR[3201] chan_unicall.c: Unicall/14 protocol error. Cause 32771
[Mar 2 10:45:07] ERROR[3201] chan_unicall.c: Unicall/15 protocol error. Cause 32771
[Mar 2 10:45:08] ERROR[3201] chan_unicall.c: Unicall/17 protocol error. Cause 32771
[Mar 2 00:58:12] ERROR[3095] app_cbmysql.c: Failed to connect to mysql database meetme on localhost.
[Mar 2 00:58:13] ERROR[3095] res_config_mysql.c: MySQL RealTime: Failed to connect database server asteriskrealtime on 127.0.0.1 (err 1049). Check debug for more info.
[Mar 2 00:58:13] ERROR[3095] chan_misdn.c: Unable to initialize mISDN
[Mar 2 05:37:28] ERROR[3107] app_cbmysql.c: Failed to connect to mysql database meetme on localhost.
[Mar 2 05:37:29] ERROR[3107] res_config_mysql.c: MySQL RealTime: Failed to connect database server asteriskrealtime on 127.0.0.1 (err 1049). Check debug for more info.
[Mar 2 05:37:29] ERROR[3107] chan_misdn.c: Unable to initialize mISDN
[Mar 2 05:47:41] ERROR[3168] app_cbmysql.c: Failed to connect to mysql database meetme on localhost.
[Mar 2 05:47:42] ERROR[3168] res_config_mysql.c: MySQL RealTime: Failed to connect database server asteriskrealtime on 127.0.0.1 (err 1049). Check debug for more info.
[Mar 2 05:47:42] ERROR[3168] chan_misdn.c: Unable to initialize mISDN
*********************************************************
[Mar 1 23:22:57] ERROR[4630] config.c: *********** YOU SHOULD REALLY READ THIS ERROR ***********
[Mar 1 23:22:57] ERROR[4630] config.c: Future versions of Asterisk will treat a #include of a file that does not exist as an error, and will fail to load that configuration file. Please ensure that the file ‘manager_additional.conf’ exists, even if it is empty.
Any idea? Thanks,
I told you to use the asterisk-r2 list
Sorry!.
Ok, Posted in asterisk-r2 list now!