TCE400P diagnostic messages

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Hello, I'm running Asterisk C.3.7.2 and DAHDI 2.4.1.2
No changes in hardware since last April and asterisk version since last summer. Since last Friday one of the call centers I support has been complaining of lagged audio, dropped calls, and "two calls at once". Now I run 5 call centers off a single ABE installation and have never had only a single site have isolated problems like this. The only piece of the equation which is different from the other sites is the use of Digium TCE400P transcoder cards. Since last Friday, these two diagnostic messages came over the…

Asterisk Users 3.5 years ago 1 Answer

CDR Analyzer/Queue stats reporting

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I've been tasked with finding and implementing a CDR/Queue analyzer to provide information to management about the call center's performance. My Google-fu seems to be returning a lot of things that are more or less abandoned projects. Does anyone have any recommended solutions for a CentOS 6 / Asterisk 10 "vanilla" server? Not opposed to something commercial, provided it actually works and isn't a disaster to set up. Thank you, Noah Engelberth
MetaLINK Technologies
System Administration
nengelberth@team-meta.net
419-636-0999 ext 100 The message does not contain any threats
AVG for MS Exchange Server (2012.0.1913…

Asterisk Users 3.5 years ago 0 Answers

AstLinux 1.0.2 Release

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The AstLinux team is happy to announce the release of version 1.0.2. This release features several security updates. All current users are encouraged to upgrade as soon as possible. Please see the documentation at http://doc.astlinux.org for upgrade details or the official release pages. Updates:
Asterisk (1.8.9.2)
DAHDI (2.5.0.2)
Rhino(0.99.5b1)
Wanpipe (3.5.24)
The Sangoma BRI/Hybrid cards (A500 + B700) are now supported via DAHDI Security Fixes:
PHP(5.3.10)
OpenSSL(0.9.8t) New Features:
A "Test SMTP Mail Relay" feature was added to verify msmtp configuration See the change log on either of these release…

Asterisk Users 3.5 years ago 0 Answers

dahdi timing

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Hi, We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are: ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks WARNING[22024] app_meetme.c: Unable to write frame to channel
Right now, dahdi in our setup uses the software timer (with res_timing_dahdi.so which gives much better results compared to res_timing_timerfd.so). On an idle system (Centos 6, Asterisk 1.8.7, dahdi 2.5), dahdi_test results are pretty good at about %99.99. However, when loaded, the numbers fluctuate between %99.90 and %99.99 which seem to cause the…

Asterisk Users 3.5 years ago 0 Answers

Correct call duration when transfer a call

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Hi.
I am new to asterisk.
I have an ivr application with asterisk and voiceglue. I make a call from
asterisk (say to A) and when callee press a button voiceglue transfer the
callee to another number (say to B). When I look cdr records the billsec
between A and B always 0 and billsec with A shows the billsec for A and B.
I am confused. Is there a reliable way to get the real call durations?
Best regards.

Asterisk Users 3.5 years ago 0 Answers

runtime codec selection

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Hi All, I don't know it's possible or not. I want to do run time codec selection by
asterisk. I have an account in sip.conf and I have active only g729 on it [2209]
.......
disallow=all
allow=g729
....... When try to dial that number when I want to use codec GSM. Is this possbile to change code after making call to exten ? *extensions.conf information is below:-* exten => _2xxx,1,Answer()
same => n,GotoIf($["${EXTEN}" = "2209"]?setchannel:gowithoutit)
same => n(setchannel),Set(foo=${CHANNEL(audioreadformat)})
; same => n,Set(CHANNEL(audioreadformat)=gsm) it's not update
anything might be readonly property, that's…

Asterisk Users 3.5 years ago 0 Answers