I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
The system itself is happy and phone calls (between two parties) seem fine.
Unfortunately, when a caller listens to a Playback recording, there
seems to be moments of stutter - perhaps 1 second of stutter for every
10
On Wed, 2012-07-11 at 15:08 +0100, Ishfaq Malik wrote:
Hi
I'm using asterisk 1.8.7
My dialplan for an inbound number is along the lines of
[default]
exten = idenfier,1,Goto(specific-context,s,1)
[specific-context]
exten = s,1,NoOp()
exten = s,2,Dial(SIP/some-extenion,20)
I have
Hey Ioan,
thanks for your answer.
It helped a little bit but I have no idea what exactly could work wrong.
My new situation:
*CLI originate SIP/123456789101112 application MusicOnHold
== Using SIP RTP CoS mark 5
-- Got SIP response 482 Loop Detected back from 192.168.0.102:5060
[Jul
Hi List!
I have a Problem with Telecom Hungary, if I set a callforwarding on the
Snom, to an external number (mobile).
Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0, snom-7.7.30
When I call the Snom (Extension 68), it responds with 302 moved
temporarily, and Asterisk try to
Mebe your operator doesnt like the CallerID(num) set as NULL just remove
the callerid(num) statement and let the standard callerId get set by
network.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten = _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten =
Unfortunately not, I already tried different forms callerid(num). Always
the same error.
I came across this entry in asterisk changelogs - maybe an update of
asterisk will help.
* Asterisk 1.4.36-rc1 Released.
2010-08-20 16:46 + [r283048-283123] Richard Mudgett rmudg...@digium.com
I forgot to ask:
Do I have to load res_fax or app_fax to use the T38 gateway capability?
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Hello,
Can someone give me an understanding about E1 with ISUP on CCS 7
signalling? Is it possible with asterisk + digium card and how
Regards,
Ashish
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you need to either use chan_ss7 or libss7.
Also look for mailing list archives of asterisk-ss7
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
You can use asterisk 1.6+ and libss7 for this functionality. Any
Digium or Sangoma card working ok on this setup. Currently i am using
it on both of them.
On Wed, Jul 18, 2012 at 5:14 PM, Ashish Agarwal ashisha...@gmail.com wrote:
Hello,
Can someone give me an understanding about E1 with ISUP
- Original Message -
From: Alejandro Recarey a...@recarey.org
To: Asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 6:30:26 AM
Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
Hi all, and thanks for taking the time to read
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 3:13:13 AM
Subject: Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY
calls
On 07/18/2012 09:43 PM, Matthew Jordan wrote:
- Original Message -
From: Alejandro Recarey a...@recarey.org
To: Asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 6:30:26 AM
Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
Hi
On Wed, 2012-07-18 at 09:16 -0500, Matthew Jordan wrote:
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 3:13:13 AM
Subject: Re:
On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote:
Hi
I'm having a problem with the entirety of a call being recorded in the
following scenario
I'm using asterisk 1.8.7.0
Person A (asterisk peer) calls Person B (not on asterisk, real world
number via a SIP trunk)
Mixmonitor is
On 07/18/2012 06:30 AM, Alejandro Recarey wrote:
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten =
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf
indicate v34 is supported, but when I enable it I get the message
res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored. Is
v34
On 07/18/2012 10:06 AM, Eric Wieling wrote:
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is
supported, but when I enable it I get the message res_fax_digium.c:1624
dgm_fax_new:
Thank you. While you are at it, ask them to document where the audio / data
from fax set g711cap| t38cap on is saved to. 8-)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday,
Hi,
I'm trying to set my system to set a caller id using the diaplan when
calling an internal extension. In other words, when I dial Joe Smith's
extension I want my own phone to show Joe Smith 555. I have sort of
managed that in the sense that my phone shows Joe Smith's caller id based on
his
Remove the ,i to start with. Do you have the various rpid related options in
sip.conf set?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, July 18, 2012 12:08 PM
To: 'Asterisk Users
I’m trying to set my system to set a caller id using the diaplan when
calling an internal extension. In other words, when I dial Joe
Smith’s extension I want my own phone to show “Joe Smith 555”. I
have sort of managed that in the sense that my phone shows Joe
Smith’s caller id based on his
On 07/18/2012 10:51 AM, Eric Wieling wrote:
Thank you. While you are at it, ask them to document where the audio / data from
fax set g711cap| t38cap on is saved to. 8-)
That is documented in the CLI help for the commands themselves; the
capture files are placed into subdirectories of the
I m trying to set my system to set a caller id using the diaplan when
calling an internal extension. In other words, when I dial Joe Smith s
extension I want my own phone to show Joe Smith 555 . I have sort of
managed that in the sense that my phone shows Joe Smith s caller id
based on
exten = 124,n,Set(CONNECTEDLINE(all,i)=Name 555-555-) instead of a
separate name and number priority.
An example of my line is:
Set(CONNECTEDLINE(all)=${cid.name} ${ARG1})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety,
Why would you NOT want the connectedline info sent immediately?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 18, 2012 12:24 PM
To: Asterisk Users Mailing List -
On 7/18/2012 2:27 AM, Jeremy Kister wrote:
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
.. ok, if the system weren't Solaris - let's say it was Debian Linux,
what would be on the list of things to check for ?
--
Jeremy Kister
http://jeremy.kister.net./
--
exten = 123,1,Verbose(1,Test)
exten = 123,n,Set(CONNECTEDLINE(number,i)=555-555-)
exten = 123,n,Set(rclidname=TestingB 123-444-)
This line is just setting an ordinary channel variable.
What do you think is supposed to use this value?
exten =
Hi Guys,
asterisk drive me crazy!
Now I have tried to use FreePBX but it require MySQL which I can not
install du to a conflict with PostgreSQL.
Does someone know, how to configure FreePBX to use PostgreSQL?
Or does someone know another Asterisk Web-Frontend, without Database?
It is realy
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