Hi everybody,
I am a newbie and trying to make calls with SIPX. I installed and configured
sipx. I am using an external DNS server and just have A entries for my domain.
That's why i am using Fully Qualified host name (pingtel.europe.nortel.com).
/etc/hosts
# Do not remove the following line,
Hi,
I have a scenario where
Phone A - xxAccess modem - IPPBX
Phone B - xxAccessi modem - IPPBX
I like to make a scenario where call from from xxAccess modem will be
treated by IPPBX.
Could we achieve it by any setting in sipx.
Is there any mac binding concept in sipx.
Do i need to configure my M
I am already anxiously looking at JIRA...if you need any help
please let me know.
Wow about those more than a dozen connections!
M. Ranganathan ha scritto:
> On Mon, Jan 26, 2009 at 4:27 PM, Paolo Prandini wrote:
>> Before considering sipx we checked other solutions, but SBCs
>> are not meant to
On Mon, Jan 26, 2009 at 4:27 PM, Paolo Prandini wrote:
> Before considering sipx we checked other solutions, but SBCs
> are not meant to be used as B2BUA and sip proxies. In fact
> we don't need nor want media relay and that's the main
> purpose of SBCs, isn't it? I think sipx could be quite the
>
Scott Lawrence wrote:
> On Mon, 2009-01-26 at 09:05 -0500, Becker, Jesse wrote:
>> I am trying to rebuild my server to the latest nightly build, however,
>> I am unable to do so as yum cannot located a dependency:
>>
>> Error: Missing Dependency: sipxconfig-conference is needed by package
>> sipxec
On Mon, 2009-01-26 at 10:02 -0500, Gilmour, Scott wrote:
> HI am using the sipXecs Server with the Siemens Optipoint 410 advance
> phones.
>
> I was able to make it so it would display the Name and the extension.
> But I was wondering if there is an easy way to make it so I can see
> the actual ip
Before considering sipx we checked other solutions, but SBCs
are not meant to be used as B2BUA and sip proxies. In fact
we don't need nor want media relay and that's the main
purpose of SBCs, isn't it? I think sipx could be quite the
solution we're looking for, apart from writing our own.
> Overa
Overall, sipXbridge was not meant as a full scale SBC. In your instance, you
might look at OSBC or something like that which will give you (I think) what
you are looking for. I don't recall sipxbridge being meant to be used in a
solution with more than a couple of dozen connections, at least in
I agree. We won't use call transfer anyway.
> Paolo Prandini wrote:
>> Yes I understand.
>> My use case is quite special, I know.
>> We are a small ISP and deliver also VoIP services, that we
>> buy from bigger operators. The problem is that our services
>> are quite reliable, but bigger operators
Damian wrote:
> Scott Lawrence wrote:
> > On Mon, 2009-01-26 at 10:11 -0500, Paul Mossman wrote:
> >> Hi all,
> >>
> >> Can anyone think of a good reason to keep the Polycom "SIP
> Settings
> >> in DHCP" screen around?
> >>
> >> It is cumbersome to use with sipXecs, and I don't think there's a
>
Scott Lawrence wrote:
> On Mon, 2009-01-26 at 10:11 -0500, Paul Mossman wrote:
>> Hi all,
>>
>> Can anyone think of a good reason to keep the Polycom "SIP Settings in
>> DHCP" screen around?
>>
>> It is cumbersome to use with sipXecs, and I don't think there's a good
>> use case for doing so anyway
Paolo Prandini wrote:
> Yes I understand.
> My use case is quite special, I know.
> We are a small ISP and deliver also VoIP services, that we
> buy from bigger operators. The problem is that our services
> are quite reliable, but bigger operators'ones are not...so
> customers complain with us of s
They are using user speed dials with blf subscriptions, sorry I thought they
were using the buddy lists.
Thanks
Chris
--Original Message--
From: Paul Mossman
To: McCoy, Chris
Subject: RE: [sipx-users] Polycom Buddy list problem
Sent: Jan 26, 2009 12:35
Chris wrote:
> Yes ihave the "mapp
no change after turning it off.
Mit freundlichen Grüßen
Jakub Ginal
Hey everyone,
I'm using Linksys SPA962 phones with a Mediant 1000 gateway running on
SipX 3.10.2. The problem I'm facing is that MoH only works with
internal calls (i.e. calls between Linksys phones), but not PSTN calls
(i.e. calls from a Linksys phone to the outside world). I believe that
I
On Mon, 2009-01-26 at 10:11 -0500, Paul Mossman wrote:
> Hi all,
>
> Can anyone think of a good reason to keep the Polycom "SIP Settings in
> DHCP" screen around?
>
> It is cumbersome to use with sipXecs, and I don't think there's a good
> use case for doing so anyway.
>
> Basically, it can be
On Mon, 2009-01-26 at 10:02 -0500, Gilmour, Scott wrote:
> HI am using the sipXecs Server with the Siemens Optipoint 410 advance
> phones.
>
> I was able to make it so it would display the Name and the extension.
> But I was wondering if there is an easy way to make it so I can see
> the actual i
Yes I understand.
My use case is quite special, I know.
We are a small ISP and deliver also VoIP services, that we
buy from bigger operators. The problem is that our services
are quite reliable, but bigger operators'ones are not...so
customers complain with us of service interruptions.
We would lik
I am of course available to do any kind of testing, if needed.
Thanks everybody
M. Ranganathan ha scritto:
> Scott Lawrence wrote:
>> On Mon, 2009-01-26 at 15:00 +0100, Paolo Prandini wrote:
>>
>>> I managed to use the sipXbridge, very nice indeed.
>>> I am however in the following case:
>>> a)
Scott Lawrence wrote:
> On Mon, 2009-01-26 at 15:00 +0100, Paolo Prandini wrote:
>
>> I managed to use the sipXbridge, very nice indeed.
>> I am however in the following case:
>> a) sipx server has a public ip, no NAT
>> b) sip trunk has a public ip, no NAT
>> c) users are on the internet with s
Scott,
Is there a workaround? I am trying to test the Cisco Plus
patches that Sen and I are working on.
Thanks,
Jes
-Original Message-
From: Scott Lawrence [mailto:scott.lawre...@nortel.com]
Sent: Monday, January 26, 2009 10:57 AM
To: Becker, Jesse
Cc: sipx-users@list.sipfoundry
On Mon, 2009-01-26 at 09:05 -0500, Becker, Jesse wrote:
> I am trying to rebuild my server to the latest nightly build, however,
> I am unable to do so as yum cannot located a dependency:
>
> Error: Missing Dependency: sipxconfig-conference is needed by package
> sipxecs
>
>
>
> Is there a wa
Ok, I must admit I was thinking of forcing things a little bit.
I understood exactly the use case, but nowhere is written
that it can work only as described, and that it must absolutely
remain in the media path; I read instead that it uses the siprelay
that is perfectly able not to stay in the medi
On Mon, 2009-01-26 at 15:00 +0100, Paolo Prandini wrote:
> I managed to use the sipXbridge, very nice indeed.
> I am however in the following case:
> a) sipx server has a public ip, no NAT
> b) sip trunk has a public ip, no NAT
> c) users are on the internet with static public IP, no NAT
> In this
On Mon, Jan 26, 2009 at 9:00 AM, Paolo Prandini wrote:
> I managed to use the sipXbridge, very nice indeed.
> I am however in the following case:
> a) sipx server has a public ip, no NAT
> b) sip trunk has a public ip, no NAT
> c) users are on the internet with static public IP, no NAT
> In this c
Scott,
Fyi, the PBX itself will not listen on multiple IP addresses...
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gilmour,
Scott
Sent: Monday, January 26, 2009 9:57 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-u
On 2009-01-26, at 16:22, Ginal, Jakub wrote:
> Hi All!
>
> I have a sipxpbx running on CentOs5 and trying to get an ACD working
> with SNOM phones.
> Test scenario consists 2 SNOM 320 phones and 1 softphone Express
> Talk from NCH Software.
> I have no prpblems with the cals between the phone
Hi All!
I have a sipxpbx running on CentOs5 and trying to get an ACD working with SNOM
phones.
Test scenario consists 2 SNOM 320 phones and 1 softphone Express Talk from NCH
Software.
I have no prpblems with the cals between the phones, all is ok.
Then i added an ACD queue with number 801, the s
Hi all,
Can anyone think of a good reason to keep the Polycom "SIP Settings in
DHCP" screen around?
It is cumbersome to use with sipXecs, and I don't think there's a good
use case for doing so anyway.
Basically, it can be used to configure the Polycom to get the SIP server
address from DHCP. It
HI am using the sipXecs Server with the Siemens Optipoint 410 advance phones.
I was able to make it so it would display the Name and the extension. But I
was wondering if there is an easy way to make it so I can see the actual ip
address?
I can see the IP address when I bootup or by going into t
Hi,
Is there an easy way to change the ip address of your SIPX Server and to add a
second interface to the sipx-server?
Thanks
Scott
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Un
Hi all,
panic over, the problem turned out to be a setting that accidentally got
changed on the Mediant Gateway. It was connecting alright, but not
automatically ansering the call. I'd hear constant ringing, the poor
bloke on the other end answered the call and would hear nothing at all.
Sort
I am trying to rebuild my server to the latest nightly build, however, I
am unable to do so as yum cannot located a dependency:
Error: Missing Dependency: sipxconfig-conference is needed by package
sipxecs
Is there a way around this? Is there something wrong with the
repository?
Thanks,
I managed to use the sipXbridge, very nice indeed.
I am however in the following case:
a) sipx server has a public ip, no NAT
b) sip trunk has a public ip, no NAT
c) users are on the internet with static public IP, no NAT
In this case I expect RTP to be between sip trunk and
users without media ser
I found the solution.
If ip_nat_sip and ip_conntrack_sip are loaded, any
application is then unable to send udp 5060 packets!
Thanks for your help anyway
Paolo
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I think it would help you if you stated how many digits the sipx system
you've setup recognizes as an "internal" call. Even though you say it is
internal since it hits another PBX on site, it's NOT internal to your
sipx system, since it passes out a gateway.
How many digits is sipx set for (the
Hi all,
I'm still playing around trying to set up call plans and I've hit a
couple of snags.
I have my Sip-X server connecting to a Mediant 1000 gateway, which
currently has a single analogue line plugged in for testing. The
analogue line is off a Siemens HiCom switch, one of several on site, a
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