Hi,
Which free or non-free (as beer) Sugarcrm plugin would you recommend
to add click to dial feature with asterisk ?
I can see a quite long list of such plugins but not all of them seem
up-to-date (judging by comparing with latest Sugarcrm version number).
Regards.
--
On 04/16/2012 08:06 AM, Olivier wrote:
Hi,
Which free or non-free (as beer) Sugarcrm plugin would you recommend
to add click to dial feature with asterisk ?
I can see a quite long list of such plugins but not all of them seem
up-to-date (judging by comparing with latest Sugarcrm version
-Original Message-
From: Niccolò Belli [mailto:darkba...@linuxsystems.it]
Sent: Monday, 16 April 2012 4:21 a.m.
To: asterisk-users@lists.digium.com
Cc: siva...@paradise.net.nz
Subject: Re: [asterisk-users] Pickup calls coming from queues
Il 20/01/2012 20:32, Alec Davis ha
Patch:
https://issues.asterisk.org/jira/secure/attachment/42605/local_remote_hint2.diff
https://issues.asterisk.org/jira/browse/ASTERISK-16735?focusedCommentId=191720page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-191720
It's already merged in asterisk 10.4-rc1,
I suspected it, but it didn't work at first. I fear I didn't understand
what the context refers to in Pickup(extension[@context]).
I will make an example: phone-100 wants to pick up a ringing phone-200
(call comes from my-sip-provider).
This is my sip.conf
[phone-100]
context=context-100
Hi,
If you are using IAX and a later version (I know it works in 1.8.x) you
can use IAXVAR.
The following URL has a post which has a good example.
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022313.html
Kind Regards
Stuart Elvish
On 04/16/2012 08:16 AM, Steve Edwards wrote:
On
Thanks Danny. I test it with blind transfer and hey, you're right, the
caller ID passed successfully, but the attended transfer doesn't.
What version did you refer to by saying 10.x ? Asterisk? Shoudn't
current version of asterisk is 1.x and should move to 2.x instead of a
big jump to 10.x ?
Hi Anam,
Hope this helps explain Asterisk version numbering:
http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/
Easy to get confused!.
Cheers,
AJ.
- Original Message -
From: Satria Anamarta anam.satri...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial
Hi
greats thanks that work very good
Olivier
Le 16 avril 2012 12:47, Stuart Elvish - IP Exchange Systems
asterisk.li...@ipesys.com a écrit :
Hi,
If you are using IAX and a later version (I know it works in 1.8.x) you
can use IAXVAR.
The following URL has a post which has a good example.
On Sat, Apr 07, 2012 at 08:20:56PM +, Noah Engelberth wrote:
In order to reconnect to asterisk (asterisk -r), you need root
permissions.
In order to use 'asterisk -r' you need write access to the unix-domain
socket /var/run/asterisk/asterisk.ctl . Root has it. Others may have it
as well.
Hi out there
We have a strange Problem here with invites.
We observe this SIP conversation.
C3 PBX - Asterisk
Case 1. Sequence Numer always increasing:
= Invite
= 401 Unauthenticated
= Invite+auth with sequence number previous Invite.
= 100 Trying etc. Works OK.
Case 2. Sequence Number
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
We use a obfuscation software to encrypt/mangle both SIP/RTP which sits
before asterisk. What happens is sometimes we don't get any voice. after
some rtp set debug we found out that when received ip of the rtp stream
is router's public ip, everything
Hi Arthur, read the article and understand,thanks :)
Btw, is there any patch for this problem without need to upgrade to
version 10.x ?
On 4/16/12, Arthur Stanfield a...@dmcip.com wrote:
Hi Anam,
Hope this helps explain Asterisk version numbering:
Hello all.
I want to know this issue is bug or not.
My Asterisk version is 1.6.2.6.
I used nat=yes on sip.conf.
##
Issue 1. SDP session handring by Asterisk
##
I used 2 clients , A and B. 2 UAC under another NAT.
No - if someone figures out a way, let me know since my receptionist doesn't
like blind transfers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria
Anamarta
Sent: Monday, April 16, 2012 7:45 AM
To:
Niccolo:
I've reopened the issue and placed some comments on the issue requesting
more information. In the future, if you need an issue reopened, you can
contact a bug marshal in #asterisk-bugs.
Thanks,
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL
Hi all,
This post is in case someone else has this problem.
The cause of the issue turned out to be one of the site technicians
having the same extension registering from his laptop as the ATA we
were testing. His laptop wasn't always connected to the voice network
and the soft-phone wasn't
It's not a bug - decrementing the CSeq header field value is directly in
violation of RFC 3261. From section 22.2:
When a UAC resubmits a request with its credentials after receiving a
401 (Unauthorized) or 407 (Proxy Authentication Required) response,
it MUST increment the CSeq header
Greetings All,
I have a compatibilty problem between asterisk 1.4 and 1.6.2
In my 1.4 asterisk I have a custom application that users call and make
recordings which recording I save to a file with the caller Id.
Below is the config file which works perfectly in 1.4
[timo]
exten =
Hi,
on asterisk 1.6.2.22
exten = s,1,Set(CHANNEL(language)=fr)
exten = s,n,Dial(Local/${myEXTEN}@context/n,,)
Transfer, Voicemail, demo, aso are played in english even despite the
fact that language=fr in sip.conf.
What's wrong? Bug?
Thanks for any hint
--
Daniel
--
We found this morning we had no SIP connection to another site. sip show
registry on the main site gave no authentication. sip show peers on
the other site showed the peer unspecified.
The odd part about this: doing sip reload on the main site made it all
work. Nothing else was changed.
On 04/14/2012 07:33 AM, Niccolò Belli wrote:
Il 04/04/2012 07:45, Anton Kvashenkin ha scritto:
Check it out, thank you.
You're welcome.
New packages against dahdi-linux-2.6.0, dahdi-tools-2.6.0, libpri
1.4.12+svn20120409 and spandsp-0.0.6~pre20:
I have experienced this issue with a provider with Asterisk 1.2, 1.6 1.8.
I never got to the root cause of the problem however it used to occur
quite frequently, now it appear to occur once every month or two -
haven't seen it occur for a while now but then I have been incrementally
updating
On 04/16/2012 08:36 AM, Billy Kaye wrote:
In my 1.4 asterisk I have a custom application that users call and make
recordings which recording I save to a file with the caller Id.
Below is the config file which works perfectly in 1.4
I am not going to say that your application doesn't work
I applied the patch to my 1.8.11.0 build and observed the same error as
shown in you t38_send.log.
I have maintained a private patch file for this functionality and
reverted to it when I too observed the INTERNAL_OBJ: user_data is NULL
message.
Do you have directmedia=no in your SIP
Hi,
Il 16/04/2012 22:50, Larry Moore ha scritto:
Do you have directmedia=no in your SIP configuration?
Yes I have.
Niccolò
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Thanks Dale,
Am not sure why it was working in 1.4 but for some reason it was ( Note : My
Asterisk is running bundled with Elastix).
But any your suggestion worked very fine.
Now am having one problem how can define those extensions only with in
different contexts, the problem I see is since am
Perhaps your problem may be that Asterisk doesn't like to send T.38 to a
peer other than the one it negotiates the SIP connection with.
If I recall correctly you mentioned a while back that eutelia made a
change which broke your outgoing T.38 functionality, did you ever find
out what the
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