Hi there,
I would like to setup up my Asterisk to do this:
receptionist answers the call, caller says he wants to leave a
voicemail message for Ashleigh, receptionist transfers the call to
Ashleigh's voicemail
I guess It has something to do with dynamic features, or probably
blind transfer to spe
Asterisk 1.4.32 (Also 1.4.26, 1.4.33)
Broadvox ITSP (xxx.xxx.xxx.xxx)
Linksys 2102(yyy.yyy.yyy.yyy)
Both peers :
canreinvite=yes
t38pt_udptl = yes
I'm having some trouble getting a T.38 fax call established with
Broadvox. During negotiation, Asterisk sends a SIP re-invite (T38
switchover)
Hi all,
I need to achieve the following function:
user 1 call to user 2, In the process they calling, if user 2 press *3 keys,
then the call hangup and playback voice file.
My setting as following:
* features.conf**
[featuremap]
textkey1 => *3
[
Hi Everyone,
I want to know a bit about the guts of the current AsterisNOW system. I know
that FreePBX is embraced as the main GUI but is just an install of CentOS
5.4 + (Asterisk/FreePBX from Yum repos)?
- Or is there anymore to this? Maybe some security tools?
- Or is Asterisk built from the so
hi, list
i want to know what is the best OS for install Asterisk 1.6.2.9,
which should work properly on working system.
i want to use CentOS5.2 or CentOS 5.4. Which is better and stable?
Thanks for your help.
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Thanks for your supporting,
have a nice day.
Sucan
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This is a very good question. I faced the same problem some time ago, and by
goggling found out that somebody had actually programmed a patch for this
purpose, but it never got approved to go into the main branch of Asterisk.
If you google, you'll probably found out details on it.
I am, however, f
Is is possible with a Polycom phone to update the LCD with the
callee's name after dialing them?
When you dial ext 103 now, it says 'To:103'...would be nice if could
have 'To:Dan Marino'
This is the case even when you have a contact for ext 103.
None of the phones I have ever tested do this, Pol
> Well, I¹ve tried this, and something just isn¹t right.
Look here:
> Event: Hangup
> Channel: SIP/ShoreTel-1-0004
> Cause: 17
> Cause-txt: User busy
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Greetings list,this question is rather a pain in my side.. i have been trying
to figure it out.. it could be simple.i have a customer with a callcenter .. we
developed a CRM "Customer Relations Management" with an SIP dialers built
in.the question is the following.. is it possible to force the
Well, I¹ve tried this, and something just isn¹t right. Here¹s the context
from "dialplan show," so I know it¹s loaded anyway:
[ Context 'PressTwo' created by 'pbx_config' ]
'*' =>1. Goto(accept|s|1) [pbx_config]
'1' =>1. ForkCDR(v,s(fullcmd=${Dat
At 8:08 AM on 28 Jun 2010, Jerry Geis wrote:
> It seems that for local channels (asterisk 1.4.33) the variable
> Variable: SIPADDHEADER="Alert-Info: Ring Answer"
> (call polycom phones and ring then auto answer)
>
> Is ignored, Is this just an oversite or is there some reason?
>
> It works fin
On 06/28/2010 03:27 PM, Carlos Chavez wrote:
> We just recently upgraded a server from Zaptel to DAHDI and Asterisk
> 1.4.30 to 1.6.2.9 and now we are getting this message before the server
> reboots every few minutes:
>
> Message from syslogd@ at Mon Jun 28 15:17:48 2010 ...
> pbx kernel: D
Hello there
You should have a look at features.conf
Regards Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 28. juni 2010 21:39
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users] Ast
On Mon, Jun 28, 2010 at 4:02 PM, mohamed daif wrote:
> i want to use asterisk as a sip server without installing any hardware in
> this machine
> the question is
> how can i configure the external getaways with asterisk
> how can i configure the costumer who is i provide calls to hem
> what is
We just recently upgraded a server from Zaptel to DAHDI and Asterisk
1.4.30 to 1.6.2.9 and now we are getting this message before the server
reboots every few minutes:
Message from syslogd@ at Mon Jun 28 15:17:48 2010 ...
pbx kernel: Dazed and confused, but trying to continue
Jun 28 15:17:
hi
i want to use asterisk as a sip server without installing any hardware in
this machine
the question is
how can i configure the external getaways with asterisk
how can i configure the costumer who is i provide calls to hem
what is the billing software can i use to calculate the the calls and
Hi,
One of the big features of 1.6 was described as multi-tenant parking.
Basically, parking people in different "lots" so the sales dept. could only
pick up their calls, and tech support theirs and no mix up was possible.
I can only find the original announcement and others asking the same
Yes
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mohamed daif
Sent: Monday, June 28, 2010 2:49 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip server
Hi,
Can i use "asterisk" as " sip server "
Hi,
Can i use "asterisk" as " sip server " for manage call Transmission between
gateways
Best Regards
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Let me know if you need any further info !!
On Mon, Jun 28, 2010 at 9:15 PM, G M wrote:
> One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0.
>
> Today, when they downloaded , the CDR from the carrier site for 26th June 2010
> , they see 50% calls are NEVER dialed by Dialer but it ap
It would need to be set in the gsm gateway and in the corresponding
section in sip.conf which connects to that gsm gateway. Everything else
should be left as rfc.
It may help or it might not.
The gateway might also have some settings you can change to improve the
detection. The Patton unit I h
Thanks Gareth, when you say that I can choose INBAND for DTMF MODE in
the GSM Gateway, that implies that the DTMF MODE of the Asterisk
extension registered for the GSM Gateway has to be set to INBAND too
or can it remain in RFC2238 ???
Because I have all my Asterisk extensions and IP telephones se
Alejandro Cabrera Obed wrote:
> Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for
> mobile phone calls coming from a GSM Gateway.
>
> All the components are set up in DTMFMODE = RFC2238, and so when the
> caller from mobile touches the IP phone LAN extension, the call is
> succ
Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for
mobile phone calls coming from a GSM Gateway.
All the components are set up in DTMFMODE = RFC2238, and so when the
caller from mobile touches the IP phone LAN extension, the call is
succesfully established. Everything is OK exce
One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0.
Today, when they downloaded , the CDR from the carrier site for 26th June 2010
, they see 50% calls are NEVER dialed by Dialer but it appears in CDR.
Amazingly, all the call durations are of 29-30 secs.
When we checked the status of
On 28 Jun 2010, at 15:36, Jonas Kellens wrote:
>>> Does this mean I have a "patched" asterisk ? (I ask this because some
>>> applications require a non-patched asterisk version)
>> Yes.
> What is then the "unpatched" version of Asterisk 1.4.30 ??
The one you have before you apply the patch?..
--
>> Does this mean I have a "patched" asterisk ? (I ask this because some
>> applications require a non-patched asterisk version)
>>
> Yes.
>
What is then the "unpatched" version of Asterisk 1.4.30 ??
Jonas.
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-
On Mon, Jun 28, 2010 at 10:00 AM, Jonas Kellens
wrote:
> I'm using asterisk 1.4.30.
>
> I've found this patch for app_queue.c :
> https://issues.asterisk.org/view.php?id=11700
>
> Can I easily implement this by issuing : wget
> 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug
Hello.
I'm using asterisk 1.4.30.
I've found this patch for app_queue.c :
https://issues.asterisk.org/view.php?id=11700
Can I easily implement this by issuing : */wget
'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug'
-O - | patch -p0/* ??
Does this mean I have a "pat
Paul Belanger schrieb:
> On Mon, Jun 28, 2010 at 5:15 AM, John Taylor wrote:
>
>> Any idea what may be happening?
>>
>>
> acknowledged
> https://issues.asterisk.org/view.php?id=16287
>
>
>
hello,
i´ve reported the same bug i´ve found out later with this issue:
https://issues.asterisk.o
Thanks Mike!
We are using one Aastra phone with expansion module and the remaining 27
phones are from Yealink (new phones that came out), currently Aastra phone
used to freeze while paging, but now we replaced the aastra to Yealink and
will see if this solves the problem.
Sandesh
On Fri, Jun 25,
On 28 Jun 2010, at 13:08, Jerry Geis wrote:
> It works fine with I call the SIP phone directly - however -
> when I first call the Local channel - then Dial the SIP phone
> the SIPADDHEADER doesnt seem to do anything.
Are you adding the header before or after you dial the local channel?
S
--
__
It seems that for local channels (asterisk 1.4.33) the variable
Variable: SIPADDHEADER="Alert-Info: Ring Answer"
(call polycom phones and ring then auto answer)
Is ignored, Is this just an oversite or is there some reason?
It works fine with I call the SIP phone directly - however -
when I first
On Mon, Jun 28, 2010 at 5:15 AM, John Taylor wrote:
> Any idea what may be happening?
>
acknowledged
https://issues.asterisk.org/view.php?id=16287
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Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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I have an Asterisk server on our LAN that serves our office VOIP
phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are
ulaw/alaw
We use attended transfer extensively. If our trunk is ulaw/alaw they work fine.
If the trunk is ilbc we have problems
1- incoming PSTN call routed via voipfo
On Mon, Jun 28, 2010 at 09:16:37AM +0200, Arjan Kroon | Mobillion wrote:
> Hi,
>
> A question.
> We are using TE420 cards.
> Normally we configure for each truncs one group.
> group=1
> channel => 1-15,17-31
> group=2
> channel => 32-46,48-62
> group=3
> channel => 63-77,79-93
> group=4
> channel
Hi,
A question.
We are using TE420 cards.
Normally we configure for each truncs one group.
group=1
channel => 1-15,17-31
group=2
channel => 32-46,48-62
group=3
channel => 63-77,79-93
group=4
channel => 94-108,110-124
My question now, is it possible to join more groups to one group?
Example:
Group
Hi,
A question.
We are using TE420 cards.
Normally we configure for each truncs one ring-group.
group=1
channel => 1-15,17-31
group=2
channel => 32-46,48-62
group=3
channel => 63-77,79-93
group=4
channel => 94-108,110-124
My question now, is it possible to join more ring-groups to one ring-group?
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