I can't seem to attach my pcap file here.
--- Begin Message ---
Good day,
I am having a problem with my ITSP at the moment, we are still trying to
identify the error, when I point the calls at 5060, the calls seems to
go through, but no audio is available both end. But when I have them
point the s
Good day,
I am having a problem with my ITSP at the moment, we are still trying to
identify the error, when I point the calls at 5060, the calls seems to
go through, but no audio is available both end. But when I have them
point the signalling back to me at 5080, I receive 500, Internal Server
Err
On Wed, 2 Dec 2009 20:47:12 -0600, Christopher Jim Capcom Coleman wrote:
> You haven't given us a lot of information to work with.
Not sure what else I needed to say until someone asked some questions I guess
:).
The fax calls would be coming into the mediant 2K, I need to route them to a
fax
You haven't given us a lot of information to work with.
In my current setup, I use a Patton 4960 with multiple DIDs. Two numbers are
given out as fax numbers. The Patton 4960 recieves all calls and based on
the numbers my sipX server routes those two numbers to a Grandstream HT502
Analog Telephone
Also, the soon to launch SipX forum also uses these message-ids.
So this message would have shown as a reply to an existing thread and not a new
thread as you intended.
-M
>>> On 12/2/2009 at 05:33 PM, in message
>>> <1259793183.3845.35.ca...@khone.us.nortel.com>, "Dale Worley"
>>> wrote
Hi Tony
Thanks for you explanation. You said you've learned..., I remember I didn't
post hunt group chain problem here, it's interesting for me to know how did
you know? My manager talked with you?
Yes, although we will probably give up sipXecs in Australian office, we
still have one sipXecs in p
Hunt groups have to be crafted carefully. I've learned you can't have the
same line in more than once,etc. I create very simple hunt groups and they
always work.
We never use asterisk anywhere so that never bites us.
We NEVER use an itsp who operate their trunks on an asterisk platform.
Call pi
On Tue, 2009-12-01 at 12:32 -0500, Burden, Mike wrote:
> I changed the subject line and removed any additional recipients (I just
> did the "reply" to quickly get the list address.)
> Did something of the original message still come through?
Almost all mail readers, when you select "reply", will a
Thank you Josh
It's a shame that we probably have to give up sipXecs, I worked on it almost
one month. sipXecs does give us so many troubles, like call pickup never
worked, hunt group chain doesn't really work We fixed most of the
problems or just simplify our requirements, but at last ... M
I am currently in Sydney Australia and trying to implement sipXecs, I think
it's easy to configure a dial plan, you'd better ask some specific detailed
questions, not such general questions, if you don't know dial plan, you need
to find a phone and read.
On Thu, Dec 3, 2009 at 12:45 AM, ch...@velo
Don't know if it can help you, but please take a look at this:
http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration
That says:
"Typically ITSPs do not handle certain types of SIP requests such as
*REFER*which is used in Call Transfer operations. To implement call
transfer,
On Tue, 2009-12-01 at 17:48 +0800, Wen Jun wrote:
> Hi, I have no idea why there are 2 SIP INVITE from SIPX to called
> party in a call. Is there any specific reason for that ?
Retransmissions can cause a duplicated INVITE. (You can verify this by
seeing that the "branch" value in the top-most "V
The source file meta/port-registry should be an up-to-date list of what
ports are used.
See the attached file or
http://sipxecs.sipfoundry.org/rep/sipXecs/main/meta/port-registry
Dale
This list is the authoritative record of what ports are reserved by sipXecs
components.
All ports are assumed
When you login to sipxconfig it will display at the bottom.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk C
Sorry, I am using Vitality.net as my sip trunk.
I am not sure which version of the software I am using, how do I tell?
Dan White
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, December 02, 2009 3:55 PM
To: Dan White
Subject: Re: [sipx-users] Problem with call for
I setup a user when a call comes in it gets routed to an internal ext
say 204, if it no one picks up, it suppose to forward to an external
number, but this just doesn't seem to work. Sometimes it rings on the
other side, but you can't hear anything. Is this a known issue?
Dan White
___
Staffan,
Please shoot me a sanitized config or just strip out pw info and I can
adjust the rest for the Wiki.
Thanks,
Mike
From: Staffan Kerker [mailto:ietf-li...@kerker.se]
Sent: Wednesday, December 02, 2009 12:36 PM
To: Dylan Ebner
Cc: Mike Ketchum; Picher, Michael;
Upgrading to the PAE enabled kernel was painless except for the fact
that the sipxecs.repo installed by the 4.0.4 iso is incorrect. In fact,
any upgrade will fail if a repository not found error.
The base directory pointed to is:
http://mirror.centos.org/centos/5.2/os/$basearch/
Changed this to
> Before you do that, go to the Systems page a Send Profiles to all the
> services - if there's anything left over in the configuration, that
> should clear it.
>
I did that, and it seems to have solved the problem. Thanks!
Jeff
___
sipx-users mailing
Thanks Staffan. I was thinking that I possibly needed an additional module. I
do not have any DSPs installed today. This is most certainly my problem. The
problem with the T track is most of those images need 256 MB of memory, where
my 3663 only has 196. Once I get a DSP I will look for a T tra
Hi
Are you sure you have DSPs installed in the router (PVDM2-8/16/32/64)? DSPs are
needed to convert TDM voice to IP voice. Make sure you have the DSPs installed
and the ISDN interface (se 0/0/0:23) should pop up once you set the pri-group
timeslots command on the controller.
Also, T-track IO
I think Staffan recommend something in the T track... and not in
mainline.
From: Dylan Ebner [mailto:dylan.eb...@crlmed.com]
Sent: Wednesday, December 02, 2009 12:29 PM
To: Mike Ketchum
Cc: Picher, Michael; Staffan Kerker; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] Cisco router
I have tried using 12.2 Mainline IP Enterprise and 12.4 Voice to Voice
Gateway. What seems to be missing is the ability to set the voice-card command
and the pri-group timeslots command in the controller. Without this I cannot
get a serial interface at 0/0/0:23. The closest I got was with the v
What IOS are you using?
Dylan Ebner wrote:
Thanks
everyone for the posts. This has
been most helpful. I have found several examples online when searching
“cisco
pstn gateway”, most of the config examples are for Asterisk, but the
basics should be the same. The pro
I figured that was it, although I wish it were not the case, I really didn't
want to have to take down the system and rebuild it... it being a production
system and all. Which is why I had to try it that way first to minimize
downtime.
The only thing that I could find was that once it was using
The only time that 3-31000 and tcp/udp are assured are when you are
utilizing sipXbridge to handle the calls.
If there is no NAT involved you are in a more typical routed network scenario.
For this type of scenario you should probably consider the following ports at a
minimum:
5060
A while back, one of the list users told me that I could either forward fax
calls using
the mediant gateway or using sipx.
What I need is to identify fax calls coming into the mediant, then forwarding
those on to
a fax server. Should I be looking at using the mediant or sipx to do this?
Thank
I've tried with 4.0.x to change between the two and always end up having
to rebuild the system... You can export your users and import them to
take some of the pain away.
Maybe there's a step I'm missing but I just can't make things work
right. May have to do with the user accounts.
Mike
Thanks everyone for the posts. This has been most helpful. I have found several
examples online when searching "cisco pstn gateway", most of the config
examples are for Asterisk, but the basics should be the same. The problem I am
running into is none of the examples are going into detail about
I think you can say it is an internal firewall.
We have 4 SIPX servers located in our datacenter which is managed by some other
company. That is why we have a firewall between our SIPXecs cluster and
internal network.
We do not have any other firewalls in our internal network. We are using Pa
I have a 4.0.2 box that I replaced 3.10.2 with. I ran two boxes and then
switched the 4.0.2 live. I believe this may have caused a problem with
authentication realms and my remote users. The original box was using the
straight domain as the sip domain, while the 4.0.2 box used a FQDN until it
On Wed, 2009-12-02 at 08:13 -0600, Robert B wrote:
> Ranga,
>
> At the end of the rope here... I am thoroughly confounded by this.
>
> I created a new sipXecs install and did nothing other than run
> sipxecs-setup, added one line, one user, and one SIP trunk (which I
> know works).
>
> I did not
Ranga,
At the end of the rope here... I am thoroughly confounded by this.
I created a new sipXecs install and did *nothing* other than run
sipxecs-setup, added one line, one user, and one SIP trunk (which I know
works).
I did not mess with any port settings at all.
Exact same issue.
-- Rob
Hi Ranga, hi all,
I tried to do what you told me.
1. In sipxbridge1 configuration I modified caller id, so that From is:
"1...@sipx1". User 3...@sipx1 calls 3...@sipx2, 3...@sipx2 makes blind
transfer to 3...@sipx1. sipxbridge2 converts refer into invite, with from:
"1...@sipx1". The call is not c
Hi,
Does anyone have a configuration dial plan for australia at all.
I have tried to get a couple of dial plans to work for australia phone
standards to no avail.
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sip
Hi,
If I remember well, the installation is quite simple: one has to type
"yum install kernel-PAE" in a shell and it will install all the
dependencies and make the PAE kernel default in GRUB. If Jim could
verify it, we can put together a page about it.
Why I'm asking for verification is, that we've
If you install the kernel after the iso install, please consider a
step-by-step how-to for the wiki.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 43
Chiris,
Thanks!
jim
> Hi,
> We've been using a PAE kernel for a while and seen no problems so far
> which we could relate to it.
> BR,
> Chris
>
> -Original Message-
> From: jnolen [mailto:jnol...@mindspring.com]
> Sent: Tuesday, December 01, 2009 9:36 PM
> To: sipx-users
> Subject: [s
Right, internal phone to phone registered to the system from outside
(through sipXbridge). Get this message. Happened to me on a separate
system as well (my home test system, at which time I blew it away and
installed 4.1.x).
Haven't tried via an alias.
Dial plan is pretty simple so it doesn't
On Wed, 2009-12-02 at 05:22 -0500, Picher, Michael wrote:
> Sorry... I tend to lump it all together. The remote phones are
> registered to 5060 and using sipxbridge-1 as SBC for NAT traversal.
They are not using sipXbridge for NAT traversal - that's not how it
works.
> For
> some reason the
Let me make sure I understand this.
Local phone dials remote user:
Local user 200 dials remote user 250 and you get this error message? Call
the remote user (250) dial local user (200)?
Can the local dial it via an alias? Cal external users dial it via DID or
AA?
Ton
Sounds to me like a bug on the Aastra side. I assume you are ok Bria to
Bria or Bria to X-Lite?
Mike
-Original Message-
From: Robert B [mailto:d...@spudland.com]
Sent: Tuesday, December 01, 2009 2:48 PM
To: Picher, Michael
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] On-
Staffan,
Perhaps a sanitized config file would help all involved. I can post it
to the Wiki also.
Thanks,
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Staffan
Kerker
Sent: Tuesday, December 01, 2009 5:22
Sorry... I tend to lump it all together. The remote phones are
registered to 5060 and using sipxbridge-1 as SBC for NAT traversal. For
some reason the PBX thinks when trying to call the externally registered
phones back that it's trying to dial back out to an ITSP.
Mike
-Original Message--
Hi,
We've been using a PAE kernel for a while and seen no problems so far
which we could relate to it.
BR,
Chris
-Original Message-
From: jnolen [mailto:jnol...@mindspring.com]
Sent: Tuesday, December 01, 2009 9:36 PM
To: sipx-users
Subject: [sipx-users] PAE Kernel
I have a customer who
45 matches
Mail list logo
