Displaying 20 results from an estimated 7000 matches similar to: "Pereserving the callerid value when presentation set to witheld over sip"
2010 Sep 15
3
Skip Busy Agents/Channels from Queue
Is there a way skip / ignore the member whose status is busy in the Queue.
I have two channel member in queue and i have set the peer limit 2 for these
members.
I want to skip those member who are currently on the call (answered to
calls) and now their status is busy, if Queue see the busy status caller
will not enter in the Queue and go to the next priority.
[test-queue]
strategy = rrmemory
2010 Sep 24
2
best format for playback/generation
Greetings fellow listers,
I have an application where I have
approximately 300 files that I playback individually or in blocks to
simulate "text-to-speech" in a "less mechanical" voice than normal Allison
files provide. These files are presently in GSM format and sound pretty
good when I play them on my computer speakers or on my in-house
2009 Jul 20
0
No subject
external AGI command.
Best regards,
Marco Signorini.
--
Marco Signorini
http://www.ethermania.com
http://www.ingegnitech.com
Roberto Piola wrote:
> we're using a Damocles Mini
> (http://www.hw-group.com/products/damocles/damocles_mini_en.html). of
> course, the damocles will have to drive a high-power relay.
>
> the damocles can be driven via snmp, so you'll have to
2009 Jul 20
0
No subject
used Kamalio to "supplement" the features that Asterisk either doesn't
provide or doesn't provide in as nice a form as the OP desired - can't
really speak beyond this as I am not one of them.
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2009 Jul 20
0
No subject
at least once a week I receive such an attack coming from a different ip.
I will read the articles. Thanks again to everyone.
Regards,
Rodrigo Lang.
2010/6/29 Kenny Watson <kwatson at geniusgroupltd.com>
> Hi, you can use fail2ban
>
2010 Apr 28
6
Asterisk 1.4.30 is slow sending STDIN to AGI script
I have upgraded Asterisk from 1.4.22 to 1.4.30 and I have noticed I am
getting a lot of errors like this on the console :-
ERROR[23912]: utils.c:968 ast_carefulwrite: write() returned error:
Broken pipe
I have tracked it down to a perl AGI script which performs our own CDR
recording. It is called before the start of the call, once answered and
again when the call is hungup.
It works fine when
2010 May 20
10
Which issue is keeping you from updrading to 1.6.2 ?
Hi,
I'm evaluating what could keep me from upgrading production systems to
1.6.2.
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Have you met other issues I should include include in my checklist ?
Regards
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2007 Mar 15
0
Re: busy/hangup/answer detection in PRI E1 channels (Vidura Senadeera)
> Hi Gareth Blades & Doug,
>
> Thanks so much for for the feedback. I have searched on lot of documents
> but couldn't able to find clear answer regarding it.
>
> I hope you guys replies are very much help all in aterisk community.
>
>
> Thanks & Regards,
>
> Vidura Senadeera,
>
> Network Engineer,
>
> Debug Solutions
>
> Sri Lanka .
2006 Mar 16
1
setting callerid not working if no callerid on incoming number
If we get an incoming call I can edit the callerID provided to add the
leading '90' and set the name so that sales calls can be identified
according to the number called.
If however the callerID is unavailable then setting the callerID name or
number fails (it shows as unavailable on the phone).
This is the call log from such an incoming call without callerID.
== Spawn extension (voip,
2013 Sep 27
2
Is this SDP payload Asterisk created valid?
We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails.
We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network.
After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488
2015 May 06
1
Recommendations for IMAP Voicemail
imapsync seems a very interesting tool.
Thanks for sharing this.
Now I'm still curious to learn if moving from local file storage to cloud
IMAP storage is still resilient to short network outages (without IMAP
replication).
Regards
2015-05-06 10:18 GMT+02:00 Gareth Blades <mailinglist+asterisk at dns99.co.uk>:
> On 05/05/15 17:52, Olivier wrote:
>
>> 2. From personal
2010 Jul 19
2
Multiple sip.conf files?
Hey, all. I'm trying to do some fun with auto-provisioning of Polycom
phones, and one thing that would make life easier for me would be if I
could have a per-phone sip.conf file. If not, no biggie -- but if there's
a way to do an include (as per extensions.conf) or something, that would
be great. I've gone through docs, and an older version of "Asterisk: the
Future of
2005 May 25
1
Default caller ID
Hi,
I've been looking at the problem of the default caller ID. When a call
comes in with no CID or witheld it's always set to 'asterisk' which is
what the phone displays. I've been looking for an option to change that.
The only place I can find is DEFAULT_CALLERID in chan_sip.c. This is
set by the 'callerid' option in the sip.conf.
However the documentation
2013 Sep 25
1
Generating a different countries ringtone on a per call basis
We can use the Dial() command with the 'r' option in order to generate
the UK ringtone (as we are UK based the default is UK).
How do we generate a USA ringtone for example?
I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us'
(and calling Progress() beforehand) and although this works for
Playtones() the Dial command still continues to play the UK ringtone.
2010 Aug 11
6
asterisk on Vmware
Hello,
Is it possible to install Asterisk on Vmware(centos) from source. Is there
any difference or disadvantage for this compared to asterisk running on
physical machine.
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2013 Sep 17
1
RTP not being switched between both SIP endpoints
We have a system where calls are coming in from telcos via an opensips
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation
being performed so I would expect asterisk to issue a reinvite after the
call is answered and switch the audio however it is not happening.
Here is the sip peer information for the call
2014 Mar 03
0
Asterisk 12.1.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 12.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New
2014 Mar 03
0
Asterisk 12.1.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 12.1.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New
2004 Oct 05
1
QOS on each interface
I have a firewall with 3 interfaces DMZ, INTERNET, LAN.
Does anyone have an example script to do QOS on multiple intefaces using
htb?
Gareth Segree
mailto:Gareth.Segree@gleanerjm.com <mailto:Gareth.Segree@gleanerjm.com>
Technical Support Analyst
The Gleaner Company Ltd.
7 North Street
Kingston
Tel: 922-3400
2008 Nov 18
0
RES: Symbols output
Sorry, the code is incomplete.
You get a better result this way...
postscript('Circle.eps',paper='special',width=4,height=4)
par(mar=c(0,0,0,0))
plot.new()
points(0.5,0.5,pch=21,cex=50,bg='gray')
dev.off()
-----Mensagem original-----
De: Rodrigo Aluizio [mailto:r.aluizio em gmail.com]
Enviada em: ter?a-feira, 18 de novembro de 2008 19:28
Para: 'Gareth Campbell'