Hello,
I can vouch for this. When running calls through GSM gateways the timing
varies substantially depending on who is being called and where they
happen to be. The call reaches the gateway and is sent out immediately
but mileage varies once it starts moving across the mobile operators
netwo
On Mon, 2008-08-04 at 17:09 +0300, otto wrote:
> How to connect Asterisk (1.4.21 we try also 1.6.0 beta-9) to sipx ?
>
>
> We follow description in
> http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_sipX_with_the_Asterisk_PBX
>
> at the bottom “Use Case 4: Use Asterisk as a PSTN gatew
Hi All,
I'm a newbie of SipX. After installation of Sipx 3.11.4 on Fedora Core 6, I
cannot access the Welcome Page with Firefox. The error message is as follows.
"Unable to connect
Firefox can't establish a connection to the server at sipx.xxx.xxx:8443"
I've already generate the certificate
On Mon, 2008-08-04 at 16:32 -0700, Max Clark wrote:
> Correct,
>
> I a) want to prevent the call from reaching the cell phone voicemail,
> and I b) want to give the answering person the option of either
> accepting or denying the call.
This is very tricky, and cannot always be made to work consi
Correct,
I a) want to prevent the call from reaching the cell phone voicemail,
and I b) want to give the answering person the option of either
accepting or denying the call.
-Max
On Mon, Aug 4, 2008 at 3:20 PM, Picher, Michael
<[EMAIL PROTECTED]> wrote:
> Why don't they just answer or not answer
Why don't they just answer or not answer the phone? Why would I need
to press 1 if I want to accept the call?
Are you trying to prevent it from going to the cell phone's voicemail?
If so just set the call forward timing such that the call gets pulled
back before the cell phone VM picks up.
Mik
On Sat, 2008-08-02 at 21:26 -0300, Tony Graziano wrote:
> Where do I look for error messages or reasons why this fails? I have no
> failed jobs and there are no phonebooks involved. I simply see the
> directory.xml file updated (date and time) but never a presence entry
> shows up in the file(s).
On Mon, 2008-08-04 at 10:23 +, Panagiotis Bekiaris wrote:
> I have installed version SIPx 3.10.2.
> I have tried to keep the configuration pretty simple. 1 polycom sipphone and
> 1
> GW. The GW is propertly configured and works 100%.
> I can call the voicemail from the phome but there is no
On Mon, 2008-08-04 at 17:09 +0300, otto wrote:
> We follow description in
> http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_sipX_with_the_Asterisk_PBX
>
> at the bottom “Use Case 4: Use Asterisk as a PSTN gateway ”
It would help if you get a trace of Asterisk's attempts to register:
On Mon, 2008-08-04 at 09:05 -0700, Max Clark wrote:
> No - specifically I mean that when the call is forwarded to a cell
> phone, that the person answering the cell phone would press 1 to
> accept the forwarded call.
I can't see how to implement this -- there's no way to get information
from the c
On Mon, 2008-08-04 at 13:32 -0700, Max Clark wrote:
> From a station we can dial out to the PSTN via our Cisco 5350 gateway,
> however from the PSTN we cannot dial into the main line (extension
> 100) and when we dial into a user's DID the station rings but you
> cannot answer the call, nor does th
on sip tracing on sipx, I just type "tshark > tmp.txt" tshark/wireshark
seems to be already installed(which kicks butt). On my machine, I just use
wireshark.
I will get a screenshot once I get into the office later today
later
Dean
On Mon, Aug 4, 2008 at 11:22 PM, Melcon Moraes <[EMAIL PROTECT
Hello,
We are attempting to migrate from an Asterisk (Trixbox) based system
to a SipX server running 3.10.2 (using the iso from sipfoundry). All
of our internal phones register and properly work (for the most part).
By this I mean we can can internally between phones, phones can reach
voicemail (b
Hi guys,
I'm having some issues concerning Agent Activity Summary report. The
number of calls and minutes handled by each agent are wrong, as you
can see on the attached screenshots. The file all_queue_activity.png
shows the real number of calls at the moment I took ths screenshots.
Those screens
Vikas Sharma wrote:
>
> hi all
> while install sipxecs jain-sip in needed >=1.2.74
>
> --> Processing Dependency: jain-sip >= 1.2.74 for package: sipxconfig
> ---> Package sipxconfig-snmp.i386 0:3.11.5-013221 set to be updated
> --> Finished Dependency Resolution
> Error: Missing Dependency: jain
Max Clark wrote:
> Hello,
>
> I need to configure a hunt group so that it rings a series of internal
> phones simultaneously and if there is no answer forward the call to an
> external number. How do I accomplish this?
>
> Thanks,
> Max
Max,
In your other thread regarding the "Follow-me", you
This was resolved by unchecking "use voicemail" in the hunt group configuration.
-Max
On Mon, Aug 4, 2008 at 9:05 AM, Max Clark <[EMAIL PROTECTED]> wrote:
> No - specifically I mean that when the call is forwarded to a cell
> phone, that the person answering the cell phone would press 1 to
> acce
No - specifically I mean that when the call is forwarded to a cell
phone, that the person answering the cell phone would press 1 to
accept the forwarded call.
-Max
On Mon, Aug 4, 2008 at 7:48 AM, Nikolay Kondratyev <[EMAIL PROTECTED]> wrote:
> Max,
> do you mean that calling user should press a b
Tony Graziano wrote:
> Using Polycom 650's with bootrom 4.1 and sip 3.03c.
>
> Built fresh centos 5 system from sipx iso and restored data, reissued
> certificates, all else seems to be OK after rebooting.
>
> When a user or superadmin creates a personal speedial, the phone
> specific directory x
I'm facing some similar issues around ACD queues vs Presence Server.
I have 1 queue with 8 agents and 'sometimes' the agents logged in shown in
Presence Server page, doesn't show on ACD Agent Stats. That way, it's like
I have no agents available. I know that 'sometimes' is way too vague, but I
th
Max,
do you mean that calling user should press a button to agree with call
transfer? May be personal auto attendant will suite you...
HTH,
Nikolay.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:sipx-users-
> [EMAIL PROTECTED] On Behalf Of Max Clark
> Sent: Monday, August 04, 200
Hello,
I want Exchange 2007 to provide answer machine for attendants or hunt groups
therefore i Can't customize the root.vxml.in file in order to forward the
voicemessages to exchange's accounts !!
Is there somebody to share his experience ?
Harry
PS: MWI with Geomant is OK but MS could prov
I found a work around and it works perfectly I can have users using both
sipx and exchange voicemail system without any problems.
it's all in the forwarding and I will post how to do this. Very easy!
- Original Message -
From: Akshata <[EMAIL PROTECTED]>
To: "Jermaine Pinder" <[EMA
Hello,
I need to configure a hunt group so that it rings a series of internal
phones simultaneously and if there is no answer forward the call to an
external number. How do I accomplish this?
Thanks,
Max
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Hello,
Is it possible to configure the follow-me functionality in a way that
the user has to press a number on the phone to accept the call?
Thanks,
Max
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The only way I've been able to get this to work is to set a call forward no
response to the user's extension prefaced with the prefix number.
Kyle
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sipx-
How to connect Asterisk (1.4.21 we try also 1.6.0 beta-9) to sipx ?
We follow description in
http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_sipX_with_the_Aste
risk_PBX
at the bottom “Use Case 4: Use Asterisk as a PSTN gateway ”
can't register asterisk to sipx
asterisk sa
On Mon, 2008-08-04 at 17:50 +0530, Akshata wrote:
> Jermaine Pinder wrote:
> > How can i run two voice mail systems?
> >
> > Is this possible?
> >
> >
> This is possible except that you have to select the priority at the
> time. It includes an issue
> http://track.sipfoundry.org/browse/XECS-
The BW field is not in the firmware template... it's one of the items
in the -directory.xml. If that file is empty the phone won't have
any blfs...
From: Tony Graziano [mailto:[EMAIL PROTECTED]
Sent: Monday, August 04, 2008 8:47 AM
To: Picher, Michael; sipx-users@list.sipfoundry.org
Subject:
hi all
while install sipxecs jain-sip in needed >=1.2.74
--> Processing Dependency: jain-sip >= 1.2.74 for package: sipxconfig
---> Package sipxconfig-snmp.i386 0:3.11.5-013221 set to be updated
--> Finished Dependency Resolution
Error: Missing Dependency: jain-sip >= 1.2.74 is needed by package
s
No, you only need to activate if you make changes to the ACD service.
One thing that is typically recommended if you have multi-line phones is
to create another extension that is used for the ACD only and make it a
second line on the phone. This way when a user calls out on their phone
they do
I reactivated a few times. do I also have to reactivate when agents sign in
and out? I guess I will try again tomorrw the reactivating. I keep getting
no agents, but I have signed them in, so I am not sure what is wrong. any
logs somewhere to look at?
thx,
dean
On Mon, Aug 4, 2008 at 6:42 PM,
Can't find "buddy watch" in the 2.0 firmware template for this in
sipxconfig. I don't think it is a firmware issue. It only happens if I
reboot a phone after the upgrade that all BLF's disappear. Any phone not
rebooted since the upgrade still has the BLF's on their display, but
their -directory.xml
Jermaine Pinder wrote:
How can i run two voice mail systems?
Is this possible?
This is possible except that you have to select the priority at the time. It includes an issue http://track.sipfoundry.org/browse/XECS-1010
I want some of my users to use sipx voice mail system and other
How can i run two voice mail systems?
Is this possible?
I want some of my users to use sipx voice mail system and others to use
exchange 2007 vm.
I have followed all the documentation but I get the exchange mailbox auto
attendant when i try to leave a voice mail for users on the exchange side.
Silly question but did you activate the Dial plan after you added the
gateway to the dial plans?
Mike
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:sipx-users-
> [EMAIL PROTECTED] On Behalf Of Panagiotis Bekiaris
> Sent: Monday, August 04, 2008 6:23 AM
> To: sipx-users@list.sipfo
Any time you make an ACD change you have to re-activate the ACD.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Hiller
Sent: Monday, August 04, 2008 1:43 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] ACD just rings and rings
I setup and ACD and added 2000
Hello
I have installed version SIPx 3.10.2.
I have tried to keep the configuration pretty simple. 1 polycom sipphone and 1
GW. The GW is propertly configured and works 100%.
I can call the voicemail from the phome but there is no way that I can make an
external call through the GW. I have tried
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