Hello users,
i have recently purchased Authentica x100p Fxo card for asterisk 1.4
i have following settings
# /etc/zaptel.conf
fxsks=1
loadzone=in
defaultzone=in
# /etc/asterisk/zapata.conf
[channels]
context=from-pstn
usecallerid=no
hidecallerid=yes
immediate=no
signalling=fxs_ks
Anyone know how to use regcontext et regexten parameter from sip.conf
and can give an example ?
Sure... let's say I have a phone with the following configuration in
sip.conf:
[myphone]
type=friend
context=inside
host=dynamic ; phone will register w/ Asterisk
secret=mysecret
Chris Bagnall wrote:
First things first. You are running /very/ old versions of firmware -
particularly on the 300 and 320. Upgrade them. I've been running
7.3.14 for some time without a problem, though it appears that 7.3.23 is
now out.
I concur about upgrading the software, but
On Aug 10, 2009, at 9:52 AM, harry R wrote:
Anyone know how to use regcontext et regexten parameter from
sip.conf
and can give an example ?
Sure... let's say I have a phone with the following configuration in
sip.conf:
[myphone]
type=friend
context=inside
host=dynamic ; phone
Hi all,
We've been having a very frustrating time with our Asterisk install (well,
okay, actually Switchvox). We have an open ticket with Digium/Switchvox but I
was wondering if anyone here might have some helpful tips.
We're basically getting glitching on the line, and in the error logs it's
/etc/asterisk/extensions.conf
/etc/asterisk/extensions.ael
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of harry R
Sent: Monday, August 10, 2009 1:22 PM
To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial
Hello,
i have a problem with the context parameter in the sip.conf. i'm using
a german sip provider (sipgate.de) and everything worked fine in
asterisk 1.4, but on 1.6.1 i got the following error message:
NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to
extension
Try prefix your extension in extensions.conf with _, e.g.
exten = _123,1,...
--
Sent from mobile device
On Aug 10, 2009, at 6:55 AM, Patrick Plattes patr...@erdbeere.net
wrote:
Hello,
i have a problem with the context parameter in the sip.conf. i'm using
a german sip provider
Thanks for the fast reply, but it does not help :-(.
Bye, Patrick
On Mon, Aug 10, 2009 at 1:01 PM, Alex Balashovabalas...@evaristesys.com wrote:
Try prefix your extension in extensions.conf with _, e.g.
exten = _123,1,...
--
Sent from mobile device
Patrick Plattes wrote:
extensons.conf:
[testing]
exten = 8001187e0,1,Dial(SIP/263)
What does dialplan show testing output?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Word of advice: When you try SIP clients, focus on how the far-end is
hearing you, not whether you can hear them. In my experience, that's
where 90% of the deal-breakers lie with the iPhone.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
What does dialplan show testing output?
[ Context 'testing' created by 'pbx_config' ]
'261' = 1. Noop(261) [SIP]
'262' = 1. Noop(262) [SIP]
'263' = 1. Noop(263)
Try putting exten = 8001187e0,1,Dial(SIP/263) in the [default]-context.
I have the same issue. Apparently your SIP-provider send calls to your
Asterisk-box from multiple IP's so that Asterisk cannot match the
inbound call on source IP and therefore sends it to the default-context.
Jonas.
On
Underscore won't help as that's for pattern matching.
Under the sip conf, have you tried adding 'fromuser=8001187e0' to the
[8001187e0] bit?
I have this in my Sipgate setup and it works. Worth a try.
Cheers
Andy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
jonas kellens wrote:
I have the same issue. Apparently your SIP-provider send calls to your
Asterisk-box from multiple IP's so that Asterisk cannot match the
inbound call on source IP and therefore sends it to the default-context.
I'd second this suggestion.
Doug
--
Ben Franklin quote:
Hi Andrew,
it didn't help. Which version of Asterisk do you use?
Thanks
On Mon, Aug 10, 2009 at 1:55 PM, Andrew Thomasa...@datavox.co.uk wrote:
Underscore won't help as that's for pattern matching.
Under the sip conf, have you tried adding 'fromuser=8001187e0' to the
[8001187e0] bit?
I
Hi Jonas,
that works fine, but I think its just a work arround and not a real
fix :-). For the moment it is okay and I'll try to fix the error next
days.
Thanks,
Patrick Plattes
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i faced the same problem with callcentric.. when i register i had to add the
extension .. like this
egister = 1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID
which caused my context to go to the default context and never use the one i
already setup..
so removing the extension in the
V1.6.1.0
[9290740]
type = peer
username = 9290740
fromuser = 9290740
secret = you-wish!
host = sipgate.co.uk
fromdomain = sipgate.co.uk
insecure = port,invite
context = inbound
caninvite = no
canreinvite = no
nat = yes
disallow = all
allow = ulaw
allow = alaw
dtmfmode = info
qualify = 5000
That
Venkateshwarlu Kakkireni wrote:
Can I mute a connected channel? Also, can I play MOH on a specific channel
without transferring it to a different MOH extension or MeetMe? I am pretty
new to the dialplans your help would be very much appreciated... Thanks in
advance...
No, Asterisk does not
I am probably just being stupid again, but...
I have some non-SIP phones which are set up for doing transfers by DTMF,
by simply adding T or t to the appropriate Dial options. This works
quite well in general.
They can also do non-directed call pickup with *8. However, after a call
pickup they
Hello Team
As you are all aware, digium has removed agentcallbacklogin as from 1.6.
Is anyone knows any work around to have say 20seats (SIP Clients), 100
agents call center for which user will have to login to the queue
dynamically from any extension and yet populate queue information with
own's
Hooman Peiro escribió:
Hello everyone,
Hey
I can not get the name of the recoding file of agents calls. I set
agents.conf as following:
; Insert into CDR userfield a name of the the created recording
; By default it's turned off.
createlink=yes
;
as you can see I set createlink=yes so
On Mon, Aug 10, 2009 at 4:53 AM, Alex Balashovabalas...@evaristesys.com wrote:
Word of advice: When you try SIP clients, focus on how the far-end is
hearing you, not whether you can hear them. In my experience, that's
where 90% of the deal-breakers lie with the iPhone.
Absolutely right! When
On Fri, Aug 7, 2009 at 7:25 PM, Pascal Bruno tipas...@gmail.com wrote:
Where you able to compile DAHDI in a virtual environment? How about skype
for asterisk? Has anyone tried that in a virtual environment? Seems like
to register the license, digium tool is looking for a connection on eth0,
On Thu, 06 Aug 2009 21:28:01 +0530 wrote
On 6 Aug 2009, at 16:32, kumarshantanu wrote:
Hello Everybody,
Hi.
I have a genuine problem in Asterisk setup.
Ok.
I have three inbound trunks in my asterisk box, everything is
What kind of trunks.
These are sip trunks
I have 6 Cisco 7940g phones and I would like to add them to my Asterisk
2.6.2 box. My SNOM 320 work just fine but I cannot get the Cisco's to
register. They pull the TFTP P0S3-08-9-00 just fine. I change the NAT to no
but it still does not register. Please advise.
Assuming you are configuring your Asterisk using the configuration
files -- if you want the caller ID on phone calls between users to be
the same as the caller id on calls made with the trunk lines, set the
external caller id information for the users in sip.conf (i.e.
callerid=9995551212) or to
Anybody tried one with Asterisk yet ? Views ?
Best Regards,
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On Aug 6, 2009, at 9:43 PM, randulo wrote:
Hi,
I've tried two SIP clients so far and both have unusable outgoing
audio quality. Skype app sounds fine, and recording the same mic
sounds fine, so I can only assume there is an issue with the clients
themselves.
Both clients allow you to
2009/8/11 Chuck Coleman p...@2cci.com
I have 6 Cisco 7940g phones and I would like to add them to my Asterisk
2.6.2 box. My SNOM 320 work just fine but I cannot get the Cisco’s to
register. They pull the TFTP P0S3-08-9-00 just fine. I change the NAT to *
no* but it still does not register.
I want to set sflphone as extension on asterisk. I have a sip
account/DID with vitelity.net. Not sure what to put in the wizard:
alias ???
hostname ??? is this the asterisk server hostname, or the hostname
where my sflphone is sitting on the lan (it's a home network)
username ??? is this
I want to adjust the timeout to send CANCEL after sending out INVITE. How to do
it?
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
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Does anyone have a working patch for the following issue on Asterisk
1.4.26 or an earlier version of 1.6 than 1.6.2? It looks like it got
committed somewhere after 1.6.1 was branched and is only available
natively in Asterisk 1.6.2.x.
https://issues.asterisk.org/view.php?id=12382
I have a
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