That works fine now in my system. Thanks so much !
While I observed a weird issue. The subscribers in A site registered in
Primary server could make call and get voice from B site, but the subs in A
site registered in Secondary server could not get voice from B site. The A
site (192.168.2.x) is
Which sites have siptrunking role enabled?
You have specified in subnets what is local on both systems?
Is your ipsec connection blacking or filtering any traffic?
On Thu, May 20, 2010 at 3:57 AM, Wen Jun jun.wen.s...@gmail.com wrote:
That works fine now in my system. Thanks so much !
I tested without siptrunking role, no change, still 4.2 to 4.0.4 does not
work.
On Wed, May 19, 2010 at 10:33 PM, Black, Dave dave.bl...@davcom.ca wrote:
One other thing Tony. I don't have the Sip Trunking role configured on
both the 4.2 and 4.0.4 system, so no internal SBC. Can you test
I enabled both of sites having sip trunking role. I've also specified the
subnets of 192.168.0.0/16 in the Internet Calling of both sites.
In my IPSec connection, I enabled the SIP/RTP over UDP/TCP in both end
points of IPSec tunnels. The strange issue is the call from subs of A site
in Primary
Hi to all..
I'm a little bit stuck here and I need some help.
I have set up the sipX at my office to which are 3 phones connected (on the
same subnet). The phones are Snom 300, Grandstream GXP2000 and Yealink SIP
T-20. All 3 phones register to sipX, and they can make calls to each other.
Then I
I typically allow all traffic between networks and disable any sip alg and
filters first. Have you tried that?
Is your internal subnet really a /16? it sounds like the ipsec connection is
also filtering something.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax:
If your remote phone iis not on a vpn connection you have 2 items to check.
Your remote router needs to have spi and alg functions turned off. Look at
the registration in sipx for it, there should be 2 ip addresses, the public
internet one, then the private one. If they are both not there it
I'm on 4.2.
The thing is that I don't want the sipx to handle RTP. Can this be done? I
have deleted the default sipxbridge.
Another thing..
I'm using dyndns as I have dynamic public ip. I have put
mydomain.dyndns.orgon the router which is pointing to internal ip of
sipx. and the FQDN of sipx
is
SOMETHING has to handle the fact that you have NAT between your remote user
and your sipx installation. That's what Session Border Controllers do. I
doubt your Microtik can do this.
You either need to get another SBC or allow sipx to be the sbc. The ONLY way
a remote user can go peer to peer in
Isn't the issue here that remote survivability wouldn't work? I currently have
my dial plans set to handle dropping prefixes - but had considered/dreaded
putting that on the gateway to enable outbound dialing in the event the phones
couldn't reach the sipx server.
-Original Message-
I don't see why survivability is affected whether it is all done at the
proxy or partly at the proxy and gateway. The same data would be sent.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security
If the gateway is expecting a prefix to be already stripped (via sipx) - and
sipx is unreachable. Wouldn't the prefix then be sent to the gateway (i.e. I
dial 9,xxx-xxx-) - which it wouldn't be able to make an outbound call?
Or, as usual, am I missing something?
-Original
In my example, the prefix (if needed) is being added at' the gateway. It
was being stripped from the phone, because calling it back' may not need
the prefix depending on your provider or gateway. So no matter what gateway
is being used the same 10 digits are being sent, each gateway adds what it
Yeah - that was me changing topic mid-stream...
I'm with you one the +1 redial issue - I was asking more in relation to remote
survivability in general. My apologies - I did, as usual, miss something
(staying on topic isn't one of my strong suits...)
From: Tony Graziano
Tony, your instructions worked perfectly. Implemented the changes last
night and all dialing works as needed, including the +1 missed calls
from the handset. Thanks!
Ly Tran
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, May 19, 2010 10:44 AM
To: Tran, Ly V.
Tony, thank you for your answer.
I understand the problem of NAT. The NAT is symmetric, so I thought I could
translate ports to internal IP addresses. (That's what I have done with 3CX
PBX and it worked). I can see the public IP address of the remote phone in
my registration in Sipx PBX:
I finally did my XMPP tests with a third server to trick Bria.
In short (the good news): it all worked!!
The longer version (including the bad news):
When I set up a third HA server with FQDN==sip-domain with only XMPP
enabled then everything works.
- Online Status is updated
- GroupChat works
Well, not exactly.
sipXbridge with a properly configured firewall will anchor the media. It
will manage the RTP and the media relay.
RTP is plainly defined (default udp 3-31000) and the symettric port nat
which sipxmediarelay employs when supporting remote users handles the first
call to be
I am going completely crazy. I am new to VoIP but I have spent a lot of time
researching and going through the documentations.
I understand and believe that SipX can be a superior product to Asterisk but
I cannot understand the following:
Many SIP Trunking Companies are offering IP-Based
Use the bandwidth.com template as the provider and change the settings
appropriately.
On Thu, May 20, 2010 at 12:43 PM, Modulus Node - Information
i...@modulusnode.com wrote:
I am going completely crazy. I am new to VoIP but I have spent a lot of
time researching and going through the
Hi there:
I'm running sipxecs 4.2 with a mixture of polycom models (301, 430,
501). I'm upgrading the bootrom to 4.x but some polycoms pick up the
upgrade and others do not.
I format the file system to do a clean install but it still doesn't
work.
Any ideas?
Thanks...
Notice of
301 and 501 are at end of life and probably won't take the bootrom versions
you are pushing to them.
Check polycoms support area where they list what versions are available on
their phones.
On Thu, May 20, 2010 at 1:00 PM, IT Services itservi...@apiwellness.orgwrote:
Hi there:
I'm running
You can't get rid of this one Paul. Multiple AA's with different
dialing plan's would not be that uncommon. IMO
It's not uncommon to have multiple Auto-attendants on a
system. I have one system with three distinct auto
attendants now. Additionally, is there really anything to
Paul - that would be a huge improvement over the current setup. Managing
multiple auto attendants is currently much more cumbersome. Moving AA out of
Dial Plans solves both issues (simplifying AA and simplifying Dial Plans).
-Original Message-
From:
Slightly off topic, would this method make it easier to implement a
simple way for end users to record the greetings for these attendants?
The current method is very cumbersome and difficult to manage.
Should I open an improvement request? I thought at one point there was a
ticket open for
Thanks all for the input. I'll take a stab at this and see if I can get
it to work. We will not be using physical faxmodems so I'm looking at
the softmodems. Can the iaxmodems work here or go with the t38modem?
Like Mr. Kitchin, I understand the part about the phantom user and have
done that
Hello all,
We currently have sipxbridge running on port 5080 and have
successfully been sending outbound calls to our ITSP for several
months. We recently started receiving inbound DIDs from the ITSP to
the sipxbridge on port 5080. The ITSP is sending the calls based on
our internal numbering
I don't get it, the DID functions keep it from being complicated unless you
complicate it. How it normally would work...
1. put the DID number in the user alias (18045551212, as in the phone
number)
2. Have the ITSP send the invite sip:18045551...@y.y.y.y.15:5080 normally.
On Thu, May 20, 2010
If the user's extension is 204, why can't I send a sip invite to
sip:2...@y.y.y.y.15:5080 and have it work? Inside our firewall invites
to sip:2...@y.y.y.y.15:5060 do what they are supposed to.
-Max
On Thu, May 20, 2010 at 11:57 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I don't get
Your firewall is sending port 5080 to 5060 instead of 5080 to 5080?
On Thu, May 20, 2010 at 3:10 PM, Max Clark max.cl...@gmail.com wrote:
If the user's extension is 204, why can't I send a sip invite to
sip:2...@y.y.y.y.15:5080 and have it work? Inside our firewall invites
to
No 5080 goes to the sipbridge, the sipbridge gets the call and doesn't
process it correctly. 5060 is sipx. There's no port translation
occurring.
On Thu, May 20, 2010 at 12:17 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Your firewall is sending port 5080 to 5060 instead of 5080 to
It is hard to tell from the 4 lines you sent in.
INVITE is different than the TO. sipx cares about the actual INVITE. If the
invite is formatted properly (sip:u...@ip-or-domain:5080) when coming from
the ITSP it should work. You are tailing the proxy, which shows the call
coming in on port 5080,
How do I generate a call trace with sipx?
On Thu, May 20, 2010 at 12:37 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
It is hard to tell from the 4 lines you sent in.
INVITE is different than the TO. sipx cares about the actual INVITE. If the
invite is formatted properly
http://wiki.sipfoundry.org/display/xecsuserV4r0/Display+SIP+message+flow+using+Sipviewer
On Thu, May 20, 2010 at 3:39 PM, Max Clark max.cl...@gmail.com wrote:
How do I generate a call trace with sipx?
On Thu, May 20, 2010 at 12:37 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
It is
Who is the itsp?
On Thu, May 20, 2010 at 3:40 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
http://wiki.sipfoundry.org/display/xecsuserV4r0/Display+SIP+message+flow+using+Sipviewer
On Thu, May 20, 2010 at 3:39 PM, Max Clark max.cl...@gmail.com wrote:
How do I generate a call
No. Setup the did as the alias in sipx, forward from sipx to the asterisk
fax user. For outbound, try having asterisk send through sipx...
On Thu, May 20, 2010 at 2:42 PM, Tran, Ly V. lt...@rrtgi.com wrote:
Thanks all for the input. I’ll take a stab at this and see if I can get
it to work.
http://dpaste.com/197178/
Call sent to extension at ip address (sipXbridge):
Line 6: INVITE sip:2...@207.171.12.15:5080 SIP/2.0
sipXbridge immediately sends it to the default destination, doesn't
even try to invite the extension:
Line 48 63: To: sip:opera...@cthought.com
On Thu, May 20, 2010
Based on experience, do you guys prefer a Xeon or Pentium processor to run
sipXecs? Would a dual Xeon 2.8 with 4 GB RAM be considered a strong system
or should we go with a different blend?
Thanks,
Ken Fulmer
___
sipx-users mailing list
Um, go to advanced setting in sipxbridge and remover OPERATOR from the
calls destination, restart services. then be happy.
What you failed to mention is that your operator is getting all the
calls. So... remove the operator from sipxbridge to receive all the
calls.
On Thu, May 20, 2010 at 3:56
Depends on the roles, expected load.
On Thu, May 20, 2010 at 3:58 PM, Ken Fulmer
kenful...@icstechnologysolutions.com wrote:
Based on experience, do you guys prefer a Xeon or Pentium processor to run
sipXecs? Would a dual Xeon 2.8 with 4 GB RAM be considered a strong system
or should we go
Devices, SBC's, sipXbridge_1, SIP (advanced)
Incoming calls destination (Default: operator)
Determines where to send inbound calls. If empty, inbound calls are
directly routed to the specified number in the inbound request and
have to be redirected by aliases or dial plan rules.
I read this as a fall back and not an override. Thank you for the catch.
Max
On Thu, May 20, 2010 at 1:08 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Devices, SBC's, sipXbridge_1, SIP (advanced)
Incoming calls destination (Default: operator)
Determines where to send
Is this resolved then?
On Thu, May 20, 2010 at 4:23 PM, Max Clark max.cl...@gmail.com wrote:
I read this as a fall back and not an override. Thank you for the catch.
Max
On Thu, May 20, 2010 at 1:08 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Devices, SBC's, sipXbridge_1, SIP
That would work fine in my situation. Now know what the details are
behind the change it makes sense. I always thought AA and VM in the dial
plan were a little strange...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On
Yes - working like a charm thanks!
On Thu, May 20, 2010 at 1:24 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Is this resolved then?
On Thu, May 20, 2010 at 4:23 PM, Max Clark max.cl...@gmail.com wrote:
I read this as a fall back and not an override. Thank you for the catch.
Max
from sipx nothing more is needed. whether there is an acl or trusted
peer thing in asterisk is another thing.
back when multitech's DID fax appliance was an FXS device, I could
send to the a patton gateway via 1...@5.6.7.8) and it would then
route it to the appropriate port and it would work
21:41:14 SIP_SI [EP IF_SIPX-00cfda78 SES 0xa72c68] Stack: 415
Unsupported Media Type
Doh! Could it be that it's trying to negotiate G.722?
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive:
On Thu, May 20, 2010 at 5:10 PM, Jim Canfield jcanfi...@emstar.com wrote:
21:41:14 SIP_SI [EP IF_SIPX-00cfda78 SES 0xa72c68] Stack: 415
Unsupported Media Type
Doh! Could it be that it's trying to negotiate G.722?
It's fixed. Turns out i was testing t.38 faxing on the Patton which
Way to go homer.
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 Contract Customers:
Mine would be going directly to a rightfax sip compliant fax server. How does
it know what port to hit it on? How do I tell sipx to allow outbound calls from
rightfax to route through it?
Sent via BlackBerry from T-Mobile
-Original Message-
From: Tony Graziano
You can specify it in the forward if it needs a port specified (ie., :5060).
I've only sent out via gateway so can't answer that directly. I would
normally send it via sip to gateway over ip, then the gateway would assume a
user identity to send it out. That was a LONG time ago. I'm sure there's
Gotcha. So I guess my question doesn't even need to involve faxing. Per your
instructions, I know how to route certain numbers to another sip server. I have
to figure out how tell sipx to allow the same sip server to send calls outbound
routed through it from the other server.
Sent via
Right. Essentially how * might register a user and dial.
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
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Fulmer
[kenful...@icstechnologysolutions.com]
Based on experience, do you guys prefer a Xeon or Pentium processor to run
sipXecs? Would a dual Xeon
This happens on all calls to any extensions voicemail
This is referred to as Call Pilot style voicemail I think. In 4.0.4 The
user recorded voicemail greeting was played and then immediately
following that the user would get the automated voice when you are
finished, press 1 for more options.
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