Planned Service Outage For Community Services On August 2, 2012

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On August 2, 2012 during a window from 10:00AM to 11:00AM (Central Daylight Time, GMT-5), the core routers that provide connectivity through to all Asterisk community services will be swapped out. The actual service interruption may only last a few minutes, but could last longer. These services could be unavailable during most, if not all, of this time window. Once the move is complete, the services will be available again, with no user-visible changes. The services affected include: bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.asterisk.org

Asterisk Users 3 years ago 0 Answers

question on meetme

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I have a meetme running that is taking audio from a PC running asterisk
(console) as input
to my server that is then feeding it using meetme to two other asterisk
PC's going out the console.
All running 1.4.43 I have noticed that when the meetme first starts if I change the input
audio (new song) it changes very fast
at the output to keep up. Over time, like one hour later, if I change
songs it can be like 5 seconds before
the output changes. How can I keep that change…

Asterisk Users 3.2 years ago 0 Answers

GoogleVoice woes

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I have two GV numbers. Both are configured to send calls to my Asterisk
1.8.13.0 box using the Google chat interface. At one time I had both
working with Asterisk. Now, for whatever reason, one of them has stopped
sending incoming calls to my asterisk box and instead just rolls to GV
voicemail. The other number continues to work fine. One is associated
with my wife's google account and the other is mine. I've compared our
account settings in Google and can't find any differences. Running "jabber
show connections" shows connections to each…

Asterisk Users 3.2 years ago 4 Answers

destroying time

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Hello, Please forgive me if I'm repeating this post.
I have searched and looked for similar question but have not seen a similar
one. When I do call from client to Asterisk 1.8, I do follows. 1. REGISTER
2. INVITE
3. call established
4. Bye
5. call ended After the call ended, my client receive the message like this. "Really destroying SIP dialog '54c09c884a61e39140a2358a5adb7d96@192.168.1.2'
Method: REGISTER"
And at least after 30sec, my client disconnect from Asterisk.
I have to do REGISTER again. I think this message means that "Asterisk will…

Asterisk Users 3.2 years ago 2 Answers

Administering phones

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I'm new to asterisk, currently working through Asterisk, The Definitive Guide. I have a couple Grandstream 200 phones and a Polycom 501 to play with. The Grandstreams were very easy to configure, but the more capable Polycom almost drove me crazy. There is so much conflicting information on the web about necessary config files and names of those files. Anyway, all my phones are now working on a basic level, but it occurs to me that once spend a lot of time learning Asterisk, I might then have to spend an equal amount of time figuring out how to configure…

Asterisk Users 3.2 years ago 1 Answer

CDRs on multiple servers.

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"Owais Ahmad" writes: > Hello guys,
>
> I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. cdr_adaptive_odbc handles multiple servers. Just define several with
[foo] and [bar] and it Just Works.
/Benny

Asterisk Users 3.2 years ago 0 Answers

Music instead of Ring Ring

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Hello, How can I achieve to play a music file instead of typical ring ring
(something like MusicOnHold)... I have the following dialplan, the time when the user calls the context is
executed and the system calls both the user and I hear a Ringing sound. [inc-call]
exten => s,1,Dial(DAHDI/i1/USER2&DAHDI/i1/USER1,20,A(sound-file)) Please suggest

Asterisk Users 3.2 years ago 0 Answers

axfer with simple CDR

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hi, i read a lot about CDR problems
this document is the best description of CDRs problem in Asterisk
http://svn.digium.com/svn/asterisk/team/murf/RFCs/CDRfix2.rfc.docx i found but
i cant still answer my question is it possible with simple CDR fully describe axfer? (axfer is asterisk
native, not phone function)
scenario
A - customer
B - secretary
C - consultant1
D - consultant2 A -> B
B axfer C
C axfer D i need to know
time B with C (consultation)
time A with C
time C with…

Asterisk Users 3.2 years ago 3 Answers

Which combination of codecs are required?

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Hi; In Voicemail.conf  If I  am using format=h263|gsm ,and i want to store only audio , then it is not storing.In log it shows that video is deposite less then 5 second. If i want to store video and audio both then it will store properly. If am using   format=gsm|h263 ,then my Xlite  softphone will go to haung. I just want to store audio and video both or some time only audio . 1)Plz guide me which combination of codec will be usefull. 2)Is there is any serial number signifance in format,ie one time if i use as format=h263|gsm and second time i…

Asterisk Users 3.2 years ago 0 Answers

Common/Reasonable Assumption on DID/Channel over-subscription

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Hello All, just throwing this out there. What are people generally using these days
when designing their services, esp. those that require a user to call a DID
to access their system, similar to calling card services. There was a time
when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of
channels bought in SMB with IP-PBX. I believe this would have changed today and assuming a service is pretty
popular, the ALOCs are longer due to cheaper rates and convenience of
calling. Does anyone have…

Asterisk Users 3.2 years ago 1 Answer