I see that the number of auto attendants may be configured, but how do I get my
incoming calls directed to a different auto attendants based on the time of day
the call is received.
For example:
if I work M-F 9-12 1-5 .. I would like a caller to reach a different
message at 10am vs 12
I have registration issue using VOIP routers (Billion7404VGPM and 7401VGPMR3,
Dlink VOIP phone DPH-C160S).
I can register with Billion 7404VGPM-old firmware, Cisco ATA 186, Xlite.
My setup:
Public IP
SipX 4.0
NAT Traversal
No NAT at SipX
These routers work with Asterisk.
In wireshark log, the
As addition if not available already...
Variables in Templates
==
Some variables should be available for use in templates so that
information can be reused. In the given case, it shoud be possible to
specify the setting of Caller ID based on data available at a different
sour
Milosz
1) Different Subnets - No, we have been putting all of our VOIP devices on the
same VLAN - so as far as layer 3 is concerned, they do not see any routing.
2) We are using SRV records.
3) I will install/test with audiocodes 5.2 firmware - hopefully tonight.
A potentially dumb question...
Two things to ask first:
1. Can assume you have your internal dialplan set for 4 digits?
2. Are you aware you are on a non-production version and that upgrading
to a stable release might not be possible?
That being said you should make sure you have a current AudioCodes
firmware version, can you
On Thu, 2009-05-14 at 12:37 -0700, Jason Jason wrote:
> Any assistance would be greatly appreciated J
>
Without a snapshot of a specific failure instance, there isn't much we
can do.
Dale
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
Lis
On Thu, 2009-05-14 at 12:37 -0700, Jason Jason wrote:
> Sipxecs Version: 3.11.12-014990 2009-03-22T14:01:49 ecs-centos5
>
> Gateway: Audiocodes Mediant 600
>
> Softphones: Eyebeam 1.5.19.5 Build 52344
>
> Hardphones: Nortel 6830
>
> Issue: Call Park Issue
>
>
>
> After setting up Call
Sipxecs Version: 3.11.12-014990 2009-03-22T14:01:49 ecs-centos5
Gateway: Audiocodes Mediant 600
Softphones: Eyebeam 1.5.19.5 Build 52344
Hardphones: Nortel 6830
Issue: Call Park Issue
After setting up Call Park, giving it extension , we are having many issues
including:
· Softphones
I have fixed the issue. Dale and Tony, you were both correct. It was both a
bandwidth.com issue and outdated firmware/bootrom software.
Tony, the update for the phone software went very smooth, thank you. Sometimes
that wiki can be a little daunting considering 50-75% of the materials appear
You can use a Dymanic IP address with your ITSP. At least I do with mine -
VOIP.MS You will need to open a pinhole in your router, and port
forwarding in the router. I have a working configuration like that today
that I would be happy to share. Stun isn't require, but if your ISP isn't
consist
You can do this as follows:
For outgoing traffic you will not have any problem, but with remote workers or
incoming calls from ITSP they will need to find the sipx host.
1. DNS:
a. If you have a dynamic ip address (DHCP lease) on your public interface
you will need Dynamic DNS and re
a few things:
(1) are the audiocodes and the sipx gear/phones on different subnets?
if so, make sure the sip alg is off on the device doing the routing
between them.
(2) is your dns resolution method set to A-record? if so, your proxy
name should be the fqdn of your sipx server, not your sip dom
On Thu, 2009-05-14 at 05:33 -0700, Bryan Hiller wrote:
> I am having an issue with 4.0.0 and call waiting.
>
> issue: When I am on the line with someone and another caller is calls
> in, the call I am on automatically drops.
This is almost certainly not a sipX problem.
If it is a problem with y
I don't think there will be any issue with running 3.1.3 firmware, even though
sipx doesn't officially support it yet, because the differences ebtween 3.1.2
and 3.1.3 refer to a phone that you are not using, so you should be good there.
I did post a link to bootrom 4.1.2, but should be revision
On Wed, 2009-05-13 at 11:18 +0200, Borginger Rikard wrote:
> Ok, sorry, I thought this was a common problem with a standard solution! =)
>
> Here are my output and logs..
>
[...]
Here is the interesting part (and the reason that sipXconfig is not
starting):
> "2009-05-12T14:13:26.437000Z":3:JAV
Your firmware and botrom are VERY old. Upgrade to bootrom 4.2 and firmware
3.1.2.
>>> Bryan Hiller 05/14/09 10:41 AM >>>
I know I have 1 trunk, but I am not sure Call waiting is disabled at
bandwidth I will check
Bootrom: 3.2.2.0019
Firmware: 2.2.2.013
Looks like I need to update... how
It's all in the wiki.
Devices>Device Files>ADD FILES... Polycom Soundpoint
Give it a name (no spaces), SAVE.
Open the NAME you just created. Upload the files below to the bootrom and sip
application locations it prvides.
After the uploads are done (you do have to download them first), ACTVIVAT
I know I have 1 trunk, but I am not sure Call waiting is disabled at
bandwidth I will check
Bootrom: 3.2.2.0019
Firmware: 2.2.2.013
Looks like I need to update... how do I do this with sipx, or do I have to
manually download the file to the tftp server?
Bryan
Still need to know bootrom and firmware on the phone. It might just be
relevant, or at least make sure the bootrom is 4.2 and the firmware is 3.1.2.
When you are on a call, how in the world is bandwidth.com sending you another
call? It should be ringing busy. They shouldn't be doing that. Have t
Bootrom/Firmware on your phone:
Is your phone remote or local to sipXecs?
Is the call you are on an outside call when you get the second call?
How many siptrunks do you have?
Who is your provider? When you say bandwidth do you mean bandwidth.com?
What is your call waiting setting for the phone is
On Thu, May 14, 2009 at 6:42 AM, Alexey Lubimov wrote:
> Is it possible to use gateway sipnet.ru with sipX?
>
> Any solutions, howto to configure sipX 4.0?
Absolutely no idea (my Russian is a bit rusty) but presumably they are
an ITSP and if so it ought to work. Any idea what kind of
infrastruct
On Thu, 2009-05-14 at 12:53 +0530, VG wrote:
> Hi,
>
> I like to know do we need a Public IP must for both in ITSP Calling
> and Remote registration?
>
> Without Public ip can both work on basis of stun..
Yes, you can use a dynamic ip address and stun. It is significantly
less reliable that
> Thanks Tony.
>
> But i want to know simply that if i configure sipx on local
> ip like sip 192.168.x.x and there is a dsl router comes in
> picture with dynamic ip given by my isp then sipx will work
> for itsp registration in that case.
Any time you deploy sipX in a private network behind a
On Thu, 2009-05-14 at 15:16 +0530, VG wrote:
> Hi,
>
> I am getting Trying and then 408 request timeout in response to registration
>
> I have attached the sipXproxy.log and sipregistrar.log for your kind
> reference.
You don't have all your DNS records set up correctly. These log entries
from
It would be good to provide some more detail with something like this.
Bootrom/Firmware on your phone:
Is your phone remote or local to sipXecs?
Is the call you are on an outside call when you get the second call?
How many siptrunks do you have?
Who is your provider? When you say bandwidth do you
The 1st caller is not dropped, They can hear my voice, but I can not hear
theirs.This happens until the 2nd caller gets voicemail and then the call
resumes.
Thanks
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.
I am having an issue with 4.0.0 and call waiting.
issue: When I am on the line with someone and another caller is calls in, the
call I am on automatically drops.
SetuP:
Public IP
4.0.0
Bandwidth with 1 sip trunk
1 polycom sounpoint 301
1 user ext 301
All calls routed to 301
Any help would de
Is it possible to use gateway sipnet.ru with sipX?
Any solutions, howto to configure sipX 4.0?
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.o
If your firewall has a dynamic ip address then you have to also use a DNS host
that supports SRV, NAPTR and Dynamic DNS, wouldn't you?
Any comments from anyone else?
>>> VG 05/14/09 5:40 AM >>>
Thanks Tony.
But i want to know simply that if i configure sipx on local ip like
sip 192.168.x.x
a
Hi,
I am getting Trying and then 408 request timeout in response to registration
I have attached the sipXproxy.log and sipregistrar.log for your kind reference.
Also specify Where Can i found config.defs file for sipx..
Regards,
VG
log.tar.gz
Description: GNU Zip compressed data
_
Thanks Tony.
But i want to know simply that if i configure sipx on local ip like
sip 192.168.x.x
and there is a dsl router comes in picture with dynamic ip given by my
isp then sipx will work for itsp registration in that case.
I like to know that itsp calling module will first ask to stun for it
This is two different questions, and the answer to one actually depends on the
other.
If your ITSP required registration, and can send you calls on a port other than
5060, then I think a dynamic isp address will work for both. If your ITSP
requires a static IP address, then that does limit you o
Hi,
I like to know do we need a Public IP must for both in ITSP Calling
and Remote registration?
Without Public ip can both work on basis of stun.
Regards,
VG
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.
33 matches
Mail list logo
