Hi Scott,
Thank you for your reply.
Here's the new siptrace.
Thanks in advance.
Rhon
On Mon, Apr 19, 2010 at 7:52 PM, Scott Lawrence xmlsc...@gmail.com wrote:
On Mon, 2010-04-19 at 10:35 +0800, Rhon wrote:
Hi Scott,
Attached is the siptrace for your reference. Tried to interpret it,
Does Sipxecs provide t38?
I try find out others solution to doing my FAX in Audiocodes gateway
mediant 1000.
Can comfirm this NET SatisFAXtion Server can support but it's expensive.
Any Idea?
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Hi All,
I would like to extend the SipX voicemail to users on
another IP telephone system.
Not to dig up old threads, what were the issues with ACD in 4.0.4 that
are of concern?
Dave B.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: April 19, 2010 5:37 PM
To:
Agents not staying logged in or appear logged in through GUI but not
receiving calls and the ACD does not like it when you transfer calls out
of queue. There may be others, but those are my major 2.
Check the tracker.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
sipXecs provides no faxing capabilities at this time.
There is a feature request for it already in the tracker.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Winson
(Elabram)
Sent: Tuesday, April 20,
sipXecs does not support the typical type of phone system signaling that
would support this.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Abdul Mayat
Sent: Tuesday, April 20, 2010 6:12 AM
To:
Tony wrote:
Subject: [sipx-users] CDR missing external calls for user portal
I am noticing that in 4.2, users cannot see any calls in the
CDR with the exception of internal calls. If they are a
member of the admin group, they can see them.
Is it just me?
--
==
Tony
On 4/19/2010 at 08:44 PM, in message
e010ecb9aab57a418d87234287233e04efc...@saca-exchange.sacaexchange.local,
Nathan Nieblas nathan.nieb...@sacatech.com wrote:
Ran into some firmware compatibility issues after upgrading from 4.0 to
4.2
Firmware 8.3.5 on 79xx would only ring internal
http://wiki.sipfoundry.org/display/xecsdev/Centralized+Voicemail
What is that for then? AFAIK sipX can provide other systems with
voicemail capabilities provided it is configured properly. I haven't
tried this before but I'm sure you could make it work.
Josh Patten
Assistant Network
In my experience people are VERY turned off by the free ones, such as
those provided by festival and eSpeak. I even went to all the trouble of
installing . We are planning to purchase cepstral voices for both
english and spanish. Though they aren't the best voices available, they
are much less
The better way to ask the question is:
Does voip.ms support static ip (in lieu of registration) on port 5080?
It is perhaps best to ask voip.ms directly.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
On 4/20/10 9:30 AM, Tony Graziano wrote:
The better way to ask the question is:
Does voip.ms support static ip (in lieu of registration) on port 5080?
It is perhaps best to ask voip.ms directly.
yes, it does. inbound, outbound, everything works fine, but when I do
that, I can't transfer
Cisco is not well known for sip adherence. Unless you are a cisco guru, IMO,
they are not worth it. I would consider moving to polycom as a more sip
compatible platform. If you enjoy bruised foreheads though...
Cisco's black and blue for a reason, eh?
Tony Graziano,
Wrong. You want to send to them on port 5060, you want them to send calls to
you on 5080.
When you say inbound and outbound it makes me think you think the two
paths are related. They are not.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email:
ROFL
In my opinion if Cisco is releasing bad SIP firmware then they shouldn't
even offer SIP firmware and stick with SCCP much like Shoretel is a
walled garden with MGCP. Would save a lot of headache with users trying
to fit a square peg in a round hole. Don't get me wrong, the Cisco phone
On 4/20/10 9:37 AM, Tony Graziano wrote:
Wrong. You want to send to them on port 5060, you want them to send calls to
you on 5080.
When you say inbound and outbound it makes me think you think the two
paths are related. They are not.
just trying to give as much info since this seems to
So, would it be possible to terminate g.729 to media services if the license
is installed under FreeSwitch?
Thanks,
Ken
From: Eric Varsanyi [mailto:sip...@eljv.com]
Sent: Monday, April 19, 2010 6:43 PM
To: Picher, Michael
Cc: Ken Fulmer; sipx-users@list.sipfoundry.org
Subject: Re:
those are computer voices similar to ATT's natural voices right? How do they
compare to an actual voice over of a real person? I've heard demos of the ATT
and at time it sounded robotic.
On Apr 20, 2010, at 9:29 AM, Josh Patten wrote:
In my experience people are VERY turned off by the free
The only way you're going to get truly human sounding voices is when a
human records them. Any text to speech voice is going to require some
tweaking to sound human. The ATT voices may have been a bad example,
I'm sure there are better ones out there. Download a demo of cepstral
and play with
Thanks.
On a side note, are you going to implement Cepstral to use for incoming
operator greetings as well? Or for that part still advisable to purchase real
voice over?
I'm contemplating whether to use a speech synthesis or a real live recording.
On Apr 20, 2010, at 9:47 AM, Josh Patten
When you register on port 5080, they should send to port 5080, and I think
the voip.ms template predefines that. If it does not, set the
ITSP Registrar Port
to 5080, and leave everything else at default with a siptrunk using
authentication.
If voip.ms has a portal that allows you so too what
On 4/20/10 9:53 AM, Tony Graziano wrote:
When you register on port 5080, they should send to port 5080, and I
think the voip.ms http://voip.ms template predefines that. If it
does not, set the
can't see voip.ms template. I get 'internal error' every time I look at it.
--
Michael Scheidell,
On 4/20/10 9:43 AM, Ken Fulmer wrote:
So, would it be possible to terminate g.729 to media services if the
license is installed under FreeSwitch?
do we need ONE license for every concurrent (non G.729) connection?
I would love to force our remotes to use G.729, and according to this,
if
Use a windows pc with IE?
When you choose your sbc route the page should refresh with a template list.
That's when your system (tapestry) is throwing up. I've noticed in BIG RED
LETTERS when logging in, that chrome, opera and others give weird behavior.
Perhaps its more like something with the
Depends on how much money you want to spend and how often the prompts
will change. If the prompts change all the time, go with a TTS voice. If
the prompt will never change you should probably get a professional
recording.
On a related not there still is not a decent mechanism for recording IVR
If it would properly negotiate the codec, perhaps.
The problem you have is that your devices all around need to support g729.
What if the caller gets to voicemail and transfers out? I'd want to
understand how the negotiation process would work to make sure I don't get a
stranded call.
I'd pose
Hi Scott,
I've raised http://track.sipfoundry.org/browse/XX-8231 Internal Error on
OK/Apply of User Group - Unified Messaging screen
Thanks for letting us know.
-Paul
paulmoss...@avaya.com
From: sipx-users-boun...@list.sipfoundry.org
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Abdul Mayat
Sent: Tuesday, April 20, 2010 6:12 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] MWI to an external system - is this possible?
A script would definitely be helpful. Thanks!
So for now, how do sipx users record the IVR from TTS? I'd hate to put a mic
in front of my PC lol
On Apr 20, 2010, at 10:00 AM, Josh Patten wrote:
Depends on how much money you want to spend and how often the prompts
will change. If the
On 4/20/10 9:59 AM, Tony Graziano wrote:
Use a windows pc with IE?
When you choose your sbc route the page should refresh with a template
list.
it does
That's when your system (tapestry) is throwing up.
not till I select VOIP.MS.
I select ANYTHING ELSE, it works fine.
I use windows XP
Yes, opening a JIRA is needed here.
Also, you might test to see if you get prompted to restart services even
though no changes are made to the gateway. If so, then add that to the JIRA.
On Tue, Apr 20, 2010 at 10:06 AM, Michael Scheidell scheid...@secnap.netwrote:
On 4/20/10 9:59 AM, Tony
Looks like Mybuddy is still being auto populated via the user groups code.
Poor guy, I never even got to know him.
attachment: mybuddy.JPG___
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On 4/20/10 9:59 AM, Tony Graziano wrote:
Use a windows pc with IE?
JIRA created:
http://track.sipfoundry.org/browse/XX-8232
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
*| *SECNAP Network Security Corporation
* Certified SNORT Integrator
* 2008-9 Hot Company Award Winner,
You see that because you probably upgraded from 4.1.x to 4.2 or just never
removed him from your buddy list.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
I need to provide toll free calls from (84)Vietnam, (65)Singapore and
(866)Taiwan. I am looking for any recommendations on ITSPs in those
countries. Hopefully they are compatible with Sipx. Anyone have
experience with this scenario?
___
sipx-users
On Tue, Apr 20, 2010 at 9:23 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
You see that because you probably upgraded from 4.1.x to 4.2 or just never
removed him from your buddy list.
Nope, was a clean 4.0.4 -- 4.2.0
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rather than doing this over a PRI (ISDN)
My topology is...
The sipx server has wan ip address. It is behind transparent
firewall. Can you point me appropriate configuration sample?
Thanks in advance
___
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List Archive:
It does a passthrough. If you have a media gateway, it will pass it through
to the end device.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Picher, Michael
Sent: Tuesday, April 20, 2010 4:22 AM
To: Winson
Have you tried it with a soft client to make sure it's not your phones?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Tuesday, April 20, 2010 6:34 AM
To: sipx-users@list.sipfoundry.org
Subject: Re:
There are currently people using static registrations on sipXecs. I have
one customer with three static registration on three gateways. Works
flawlessly.
However, I have not updated them to 4.2 yet. They are on 4.0.4.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
On Tue, 2010-04-20 at 09:42 -0400, Michael Scheidell wrote:
On 4/20/10 9:37 AM, Tony Graziano wrote:
Wrong. You want to send to them on port 5060, you want them to send calls to
you on 5080.
When you say inbound and outbound it makes me think you think the two
paths are related.
On Tue, 2010-04-20 at 11:57 -0400, Roman Gelfand wrote:
My topology is...
The sipx server has wan ip address. It is behind transparent
firewall. Can you point me appropriate configuration sample?
Consider the following excellent advice when requesting help...
I used a polycom v 3.13c and cisco 7960 .
On 4/20/10 12:11 PM, Todd Hodgen wrote:
--
Michael Scheidell, CTO
Phone: 561-999-5000, x 1259
*| *SECNAP Network Security Corporation
* Certified SNORT Integrator
* 2008-9 Hot Company Award Winner, World Executive Alliance
* Five-Star
On 4/20/10 12:00 PM, Scott Lawrence wrote:
If they send a call to you on port 5060, that call will 'work', but
transfers of that call will not. If they send it to you on port 5080,
the call will work and will be transferable. You cannot distinguish
these except by looking at the network
On Tue, 20 Apr 2010 09:00:34 -0400, Matt White
mwh...@thesummit-grp.com
wrote:
We are testing 79xx series with P0S3-08-11-00, the only issue we are
seeing right now is that the MWI never turns off with 4.2
I'm having the opposite issue, I cannot get the MWI to turn on now..
But we have
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Abdul Mayat
Sent: Tuesday, April 20, 2010 11:57 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] MWI to an external system - is this
It appears that configuration files were not upgraded. Is there a way
to reset all configuration and start from clean slate?
Thanks in advance
___
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List Archive:
Can you back up a step and clarify how this works?
Is the idea to set up a dedicated IMAP server for voicemail that we
have our users configure their email clients to access in addition to
any servers/accounts they already have for email?
Or are we having sipx authenticate with whatever
I find that faxing works fine on my sipx system, using Linksys
SPA-2102 boxes as client devices. My ITSP connection is through
SIPXBridge, using G711U.
Jeff
On Apr 20, 2010, at 12:00 PM, Todd Hodgen wrote:
It does a passthrough. If you have a media gateway, it will pass it
through
to
Do you mean to 4.2?
It might help to understand the specific version of 4.1.7. I am unclear if
4.1.7 1as upgradeable to 4.2 as the schema might have changed.
I would export and import. I would delete the superadmin from the
exported file before importing. The import to a fresh 4.2 install.
I've been making some simple AA prompts by going to Cepstrals demo page and
typing in the annotated phoneme stuff (starting with just english then making
it fancier to tweak it to sound right) and when it sounds right I just 'save
as' the file as a .wav file locally and then upload it to the
On Tue, 2010-04-20 at 13:19 -0400, Roman Gelfand wrote:
It appears that configuration files were not upgraded. Is there a way
to reset all configuration and start from clean slate?
To be clear ... upgrading into or out of development versions ( X.Y.Z
where Y is odd ) is Unsupported.
Yes,
I forgot about the conference bridge.
Like I said, I sure would like to use G.729 for our mobile/remote
phones. 8k on their DSL/home broadband is better than taking up 64K.
(At least in theory).
I would also like to find a sip client for my windows mobile (or wait
till I get my android based
Well, you could set up a separate IMAP server dedicated only to storing
voicemail. I was going to do this when I found out Groupwise has a very
watered down IMAP implementation that didn't support the necessary IMAP
functions (we have since decided to ditch Novell altogether and run
Exchange
I may be misunderstanding your configuration, but it seems like if you don't
care about AA/Voicemail/Conferencing then the freeswitch license isn't going to
help (or hurt) you at all.
If your remote callers would be better off with a different codec you could try
to limit the codecs their
gmail does not work since they don't support header searching. Yes, this is
rather ironic for a Search company.
Peter
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Tuesday, April 20,
I really ONLY care about AA/Voicemail/Conferencing (since everything
else seems to work).
I am assuming the only time I will need it is in freeswitch, caller
using G.729, hits AA/vmail or conference. I get 4 licenses, I am
assuming 5th concurrent caller (using G.729) will probably not hear
Yes, this is development version. I guess I will install the 4.2 version.
On Tue, Apr 20, 2010 at 1:46 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Do you mean to 4.2?
It might help to understand the specific version of 4.1.7. I am unclear if
4.1.7 1as upgradeable to 4.2 as the
Gmail does not work.
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:
I would 'hope' that freeswitch simply stops advertising g729 and the call will
negotiate at a crappier codec (assume the remote guy allows g722 as a lower
priority codec) if you're out of licenses, but I don't know that for sure. If
you only allow g729 on the remote end then, sure, the call
Does your ITSP support speex or iLBC? If so, consider the following:
in /etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml:
param name=codec-prefs value=G722,p...@20i,p...@20i,speex,L16/
speex is supported by FreeSWITCH and it's a very robust low bandwidth
codec albeit at some CPU
On 4/20/10 12:38 PM, Geoff Brozny wrote:
On Tue, 20 Apr 2010 09:00:34 -0400, Matt White
mwh...@thesummit-grp.com
wrote:
We are testing 79xx series with P0S3-08-11-00, the only issue we are
seeing right now is that the MWI never turns off with 4.2
I'm having the opposite issue, I
Wow... new to me!
He wants to use another IP system though...
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Tuesday, April 20, 2010 9:19 AM
To: sipx-users@list.sipfoundry.org
Essentially sipX is sending the NOTIFY(?) to the mediant gateway based
on the dial plan and technically, other than the configuration aspect,
the Mediant is the same thing as an unmananged SIP gateway in all but
name, so assuming the other IP PBX is properly routed to the sipX
installation and
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Picher, Michael
Sent: Tuesday, April 20, 2010 3:30 PM
To: Josh Patten; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] MWI to an external system
Hi all,
Sipxecs seems to perform a little weird on transferring calls from the IVR when
they're coming from an ITSP trunk. Incoming calls work fine, and are (by DID)
answered by the auto attendent/IVR. When the user selects a menu option, the AA
says 'Please hold while I transfer your call'
On Tue, 2010-04-20 at 22:18 +0200, r.vanv...@raffel.nl wrote:
Hi all,
Sipxecs seems to perform a little weird on transferring calls from the
IVR when they're coming from an ITSP trunk. Incoming calls work fine,
and are (by DID) answered by the auto attendent/IVR. When the user
selects a
aside from 'traces, always do traces' (full traces)
some minor things to look at, consider: poraone is doing nat? why not
just set sipx up to do it for you.
codex? AA and IVR use G.711. make sure all parties involved arn't
rejecting G.711 (including any issues with bandwidth)
other than
Josh,
Pandion and Spark are XMPP-only.
What exactly does Pidgin not do that would keep it from working with
SipX? I use Pidgin exclusively (since the Gaim days) and have never had
issues with it.
-- Robert
On 4/19/2010 9:32 AM, Josh Patten wrote:
Use Pandion or Spark for your IM client.
I never could get user searches to work on Pidgin
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/20/2010 3:29 PM, Robert B wrote:
Josh,
Pandion and Spark are XMPP-only.
What exactly does Pidgin not do that would keep it from working with
SipX? I use
Thanks for the replies! In reply to the considerations:
- Portaone isn't doing NAT, as the sipxecs box is running without any nat
(public routable IP). However, portaone does nat detection and will proxy media
when UA is natted. But this is not the case when the UA isn't behind NAT.
- G.711
Josh,
Pandion and Spark are XMPP-only.
What exactly does Pidgin not do that would keep it from
working with SipX? I use Pidgin exclusively (since the Gaim
days) and have never had issues with it.
99% of the whole M testing and soak we have done leading up to 4.2 has been
done using
On 4/20/10 4:31 PM, r.vanv...@raffel.nl wrote:
G.711 isn't a problem. Using a phone fixed on G711 no problems
encoutered. Moreover, prompts and DTMF are correctly transmitted.
are you saying that when you fix the phone at G.711 it works FINE? than
you have a codex problem. AA and vmail use
On Tue, Apr 20, 2010 at 4:31 PM, r.vanv...@raffel.nl wrote:
Thanks for the replies! In reply to the considerations:
- Portaone isn't doing NAT, as the sipxecs box is running without any nat
(public routable IP). However, portaone does nat detection and will proxy
media when UA is natted. But
FYI I just confirmed using the X-Lite Client that FreeSWITCH supports
both the narrowband and wideband speex codecs. From what I understand
it's lower bandwidth than G.729:
http://lists.xiph.org/pipermail/speex-dev/2006-May/004453.html
I couldn't get iLBC to work with FreeSWITCH/Polycom/xLite
We are using Polycom phones and I don't think they support speex or ILBC (at
least not all the models).
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Tuesday, April 20, 2010 3:57 PM
To:
I believe the newer models support iLBC, but I couldn't get it working.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/20/2010 4:08 PM, Ken Fulmer wrote:
We are using Polycom phones and I don't think they support speex or
ILBC (at least not all the
Yeah, the codec appeared in the web browser menu, but I couldn't get it to
work either.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Tuesday, April 20, 2010 4:09 PM
Cc: sipx-users@list.sipfoundry.org
Subject: Re:
See what happens when you speak your mind.
How does one get pandion to use your connection to sipx? How do you create
the resource required for that within the client?
On Mon, Apr 19, 2010 at 10:32 AM, Josh Patten jpat...@co.brazos.tx.uswrote:
Use Pandion or Spark for your IM client. Pidgin
I don't follow you. Are we talking about the same Pandion?
http://pandion.im/
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/20/2010 4:15 PM, Tony Graziano wrote:
See what happens when you speak your mind.
How does one get pandion to use your
On Tue, 2010-04-20 at 22:18 +0200, r.vanv...@raffel.nl wrote:
1.) SipXbridge gets the REFER from sipxecs' proxy, and correctly
translates it to re-INVITE for the ITSP.
2.) The ITSP replies with 100, trying, followed by 200 OK.
3.) SipXbridge replies with BYE, Reason: Protocol error 200 OK
Yes. I cranked it up and it evidently found my email account and proceeded
to let me login. I signed out but am unable to get it to let me into my 4.20
acount. I don;t see how I can create a resource (/Home) in order to tell
it to use the domain. I get login failures.
On Tue, Apr 20, 2010 at 5:17
login format:
username: use...@sipx.domain.name
password: IMPassword
Works for me.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/20/2010 4:20 PM, Tony Graziano wrote:
Yes. I cranked it up and it evidently found my email account and
proceeded to let me
2.6.90 stable.
If you're using SRV have you set up the proper XMPP SRV records? If not,
try pointing it to the XMPP host instead of the domain name.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/20/2010 4:38 PM, Tony Graziano wrote:
Doesnt for me,
Doesnt for me, What version pandion?
On Tue, Apr 20, 2010 at 5:23 PM, Josh Patten jpat...@co.brazos.tx.uswrote:
login format:
username: use...@sipx.domain.name
password: IMPassword
Works for me.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On
yes and yes. Pidgin works fine, so does spark.
On Tue, Apr 20, 2010 at 5:41 PM, Josh Patten jpat...@co.brazos.tx.uswrote:
2.6.90 stable.
If you're using SRV have you set up the proper XMPP SRV records? If not,
try pointing it to the XMPP host instead of the domain name.
Josh Patten
Is there a way to configure SipX to try SIP over UDP BEFORE attempting SIP
over TCP? It seems at the moment, that my install of sipx tries SIP over
TCP first for all unmanaged gateways and only if that fails will it switch
to SIP over UDP.
Thanks!
Orrin
What kind of gateway are you using? The norm is to use udp. If the message
is too large its preferable to switch to tcp.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems
P0S3-08-11-00 is for 7940 and 7960 models, I guess I should have been
more detailed.. We are using 7941, 7961 and 7970's with Polycom 3xx
series.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matt White
Sent:
I have an SBC which allows both SIP over UDP and over TCP (for a specific
application). Preferably though, I'd like all SIP Signaling over UDP to
have as much consistency as possible. The SIP message can't be more than
400 bytes or so. This wouldn't qualify as too big for Sipx, would it?
Orrin
If you add G729 to the codec list for FreeSWITCH (assuming you care
about G729 support for bridged calls and you do not want to hear
silence for MOH and assuming that G729 support implies FreeSWITCH will
stream MOH using G729), you may want to make sure it is also added to
the allowable codec set
If you are really serious about using it you should invest in a device for
trunking and remote users that performs transcoding.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control
Hi All,
I had a working Sipx installation (3.10.2-013143 2008-07-23T18:09:14
ecs-centos5) with self certification.
I obtained a domain wildcard certificate from Go Daddy, and tried
installing it as per http://sipxecs.sipfoundry.org/doc/INSTALL.ssl.html
however have run into trouble.
When I
Hi All,
Some further information:
I got SipX to start, by changing /etc/init.d/sipxpbx
## check certificate
/usr/bin/ssl-cert/check-cert.sh \
--name ${SIPXCHANGE_DOMAIN_NAME} --name ${MY_FULL_HOSTNAME} \
--fail 5 /etc/sipxpbx/ssl/ssl.crt
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Orrin Doyle
[or...@yeagerworks.com]
I have an SBC which allows both SIP over UDP and over TCP (for a specific
application). Preferably though, I'd like
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Orrin Doyle
[or...@yeagerworks.com]
Is there a way to configure SipX to try SIP over UDP BEFORE attempting SIP over
TCP? It seems at the moment, that my
Wildcard certificates are known to work in sipX 4.2. Won't work on any
version prior to 4.2
Graeme Allen wrote:
Hi All,
Some further information:
I got SipX to start, by changing /etc/init.d/sipxpbx
## check certificate
/usr/bin/ssl-cert/check-cert.sh \
Hi Scott,
I turned the logging level to debug. Here's the result of siptrace.
Calling the cisco 7970g from polycom will redirect to IVR with an error The
owner of extension 112 is not available.
Thanks in advance.
Rhon
On Tue, Apr 20, 2010 at 9:45 PM, Scott Lawrence xmlsc...@gmail.com wrote:
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