I recently configured T.38 on Asterisk 10.4.2. When I send the fax to
Asterisk, it gives the errors as listed below; WARNING: app_fax.c:442 transmit_audio: channel
'SIP/192.168.1.69-00000000' refused to negotiate T.38
WARNING: app_fax.c:174 span_message: WARNING T.30 ECM carrier not
found As in sip.conf the configuration is listed below; t38pt_udptl = yes,fec,maxdatagram=400
faxdetect = t38 And the rest are the standard configuration. Please advise to resolve this issue.
i am trying to install the just from git cloned app_swift version.
Compiling works fine. Install as well. But if i try to load the module
at Asterisk it fails with. Command 'module load app_swift.so ' failed.
[Jun 20 11:29:51] WARNING: loader.c:460 load_dynamic_module:
Error loading module 'app_swift.so':
/usr/lib/asterisk/modules/app_swift.so: undefined symbol: swift_port_close
[Jun 20 11:29:51] WARNING: loader.c:850 load_resource: Module
'app_swift.so' could not be loaded. My System Informations: server*CLI> core show version
Asterisk 188.8.131.52 built by root @ server on a x86_64 running Linux on
2012-06-20 08:55:14 UTC root@server:~# uname -r
Just a note to let everyone know I've finally finished up the new BETA release of app_swift (now v3.0.1 b1).
This release introduces some pretty major changes to app_swift such as: - The entire code-base has now been unified and the build system auto detects which Asterisk version you're using (yay! one branch!) - Auto-detection and support for both the Cepstral 5.0 and 6.0 engines - Support for Asterisk 1.4, 1.6, 1.8, 1.8 Certified, and 10 - Asterisk 1.2 support has been dropped.
I have only been able to do some basic testing with…
Has anybody here successfully integrated IMAP voicemail with Exchange 2010? If so - could you point me in the right direction to configure this properly?
I'm running Asterisk 1.8.11-cert2.
searchcontexts=yes (*I have tried adding imapport=993 and imapflags=ssl, but just adds a 'security problem' error message to the logfiles). my voicemail users are in Realtime, and I have a column populated and set for imapuser (you can see in the log errors below that we are successfully getting…
Just thought to let you know of a weird issue in Asterisk 1.8.? + Dahdi
2.6.? (and 2.5.?). When you specify any cadence in an app (Dial, Queue) then caller id does
not work. For instance with the default cadences (everything commented out in
chan_dahdi.conf) : Dial(DAHDI/54) caller id works Dial(DAHDI/54r1) caller id does not work (even for r1) I just found this issue did not have time to investigate further. Can
anyone else verify that this is true for tonezones other than 13 (gr) which
I am using? Cheers, Panos
notice in the console output beneath that there is a resultid 6 but it
can not be cleared :
[May 5 11:46:27] -- Executing [s@sub:3] MYSQL("SIP/vart-00000336",
"Connect connid localhost dialplan host Asterisk") in new stack
[May 5 11:46:27] -- Executing [s@sub:4] MYSQL("SIP/vart-00000336",
"Query resultid 4 DELETE FROM pickuptbl WHERE pickmark LIKE
"%SIP/vart2-00000336%"") in new stack
[May 5 11:46:27] -- Executing [s@sub:5] MYSQL("SIP/vart-00000336",
"Clear 6") in new stack
[May 5 11:46:27] WARNING: app_mysql.c:194 find_identifier:
Identifier 6, identifier_type 2 not found in identifier list
[May 5 11:46:27] WARNING: app_mysql.c:510…
I hope for a hint on this issue. I had a voicemail running on ast release 1.6.2 latest which i upgraded
to 1.8.11 latest release.
during this process I did add a couple of fields like minsecs and maxsecs. I do now get empty emails where the attached voicefile only contains the
the message length written in the email is ok.
If I go to the voicemailbox during the recording then I can se the files
grow to the filesize i would expect, looks like everything is ok until then.…
In Asterisk, how do I catch the audio stream in real-time for another
application? The audio stream can be in the form of samples or RTP Cells , just need
real-time. Because of that I have the applications myself that need to analysis the
audio in asterisk. I have read the Asterisk Definitive and find that the "app_jack" may be my
solution. But, I still need to hear that anyone have the opinions ? I'm the new one for asterisk and it's my first time to send mail for mailing
list. Please let me…
I'm giving a try to FollowMe app. I've configured a few profiles and
everything but caller id worked fine. The problem is that caller id is
replaced by "asterisk" and called parties don't see the real caller id
(regardless the PSTN destinations in follow me profiles, those cases are
perfectly understandable because caller id is replaced by trunk or line
used to send the call out). Is it possible to forward or pass-through the
original caller id when using FollowMe app?
Thanks for your help. Paul