On Fri, 7 Oct 2011 14:58:01 -0400, Tony Graziano wrote:
> I simply don't think you understand the difference between an openfire sip
> implementation and ANY other sip implemntation.
I already said this is all surface to me right now. I just started looking into
trying to accomplish this a few da
t voice for sipx in a
> lan only setting. Good luck though.
>
>
> On Fri, Oct 7, 2011 at 2:15 PM, m...@grounded.net wrote:
>
>>> their sip implementation is not very good. you need ti be using
>> spark/openfire at every end to make that work. further, their sip plugin
> their sip implementation is not very good. you need ti be using
> spark/openfire at every end to make that work. further, their sip plugin
> has not been updated in years I think. good luck.
There are so many clients for openfire, you'd think there's got to be something
that would work.
Even i
though since the Registerar must be
> given a concept of unified XMPP and SIP presence so it could redirect to
> either a real SIP UA or to an XMPP user via mod_dingaling.
>
>
> On 10/07/2011 05:38 AM, m...@grounded.net wrote:
>>> You should look at sipdroid as well, not sure
client built in.
That could be a tie that makes it happen to off to search again.
> http://sipdroid.org/
>
> Kyle
>
> On Thu, Oct 6, 2011 at 1:09 PM, m...@grounded.net wrote:
>>> You need to state what you are doing, or trying to do in the terms of
>>> "from
>
> You need to state what you are doing, or trying to do in the terms of "from
> a softphone registered to sipx" or "from an xmpp client registered to
> openfire on sipx" when discussing this, because you appear to be jumping
> all over the place to people besides me.
It appears that way because th
> The android support is probably not quite there yet anyway.
It 'sort of' works with the latest sipdroid.
I can make a call from the tablet to a sipx connected x-lite.
I can make/answer the call on either sipdroid or x-lite
I can start sending video but it only sends the front facing video cam
> Install big blue button on a server, android supports flash so you
> *should* be able to do the video portion, create a gateway in sipx to
> your BBB server, and use bria or PSTN to dial into your BBB server for
> audio.
Hi Kyle,
I haven't tried bbb since when it was first announced here. I did
There's hope!
http://code.google.com/p/redfire/
On Thu, 6 Oct 2011 10:57:27 -0400, Michael Picher wrote:
> it's not bastardized in any way...
>
> i like xabber on my tablet / phone.
>
> On Thu, Oct 6, 2011 at 10:20 AM, m...@grounded.net
> wrote:
>> Is ope
> If you have sufficient bandwidth (wifi) and the softphone clients on both
> ends support it (and have the same codecs), it should work.
I just tested this and when you want to go into video mode while in IM, it just
automates a sipx call over x-lite/bria in this test case.
Since bria android do
on both
> ends support it (and have the same codecs), it should work.
>
> On Thu, Oct 6, 2011 at 11:15 AM, m...@grounded.net
> wrote:
>> On Thu, 6 Oct 2011 10:57:27 -0400, Michael Picher wrote:
>> it's not bastardized in any way...
>> i like xabber on my tablet /
o achieve.
Mike
> On Thu, Oct 6, 2011 at 10:20 AM, m...@grounded.net
> wrote:
>> Is openfire/jabber on sipx a bastardized version in some way or is it a
>> fairly standard, but integrated setup?
> I ask because it seems there are endless clients for pc/cell/tablets for
>
On Thu, 6 Oct 2011 09:57:32 -0400, Tony Graziano wrote:
> good luck. very few existing androids actually meet the requirements of
> adobe air to do what you are talking about now:
I was able to test an air/flex based method last night which did work but I
don't want to get into anything overly co
separate account for that user on openfire or is there just the
one sipx account which gives the user access to both?
On Thu, 6 Oct 2011 08:52:08 -0500, m...@grounded.net wrote:
> Looks like as of yesterday it might even be, adobe AIR does allow access to
> any camera.
> You need Androi
Looks like as of yesterday it might even be, adobe AIR does allow access to any
camera.
You need Android 2.3 and Adobe AIR 3.0 installed.
On Wed, 5 Oct 2011 11:59:01 -0500, m...@grounded.net wrote:
> I see that there is a bria client for the android but there is no mention
> of h.264 vi
On Wed, 5 Oct 2011 13:15:33 -0400, Tony Graziano wrote:
> they do? the real presence stuff looks interesting but its not a pure sip
> implementation(have to use their platform to support it), plus its not
> shipping yet. do they have another solution?
From what I've been coming across, for what i
> dont confuse bria android and bria desktop.
> bria mobile does not have h.264. period.
That's what I've come up with also but figured it was worth asking as
sometimes, there are things which aren't very well documented.
Now, when you say don't confuse the two, you mean that even if I found this
om: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
> Sent: Wednesday, October 05, 2011 10:07 AM
> To: m...@grounded.net; Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Does Bria on cell/
I see that there is a bria client for the android but there is no mention of
h.264 video or any other mention.
Is there a way for android bria clients to send their video to other sipx users
connected also via bria?
I need a solution which allows cell/tablet users to be able to send their video
> The only problem in the open source version of HylaFAX is building that
> damn T38 modem. I still haven't figured that one out and just don't have
> the time. Building open source HylaFAX itself isn't that hard.
We have asterisk/hylafax/avantfax set up to allow both incoming and outgoing
faxi
I still don't know where I can set the relay however so sipx doesn't try to
send directly.
We tested this morning and vm's are coming in as they should be.
The SMTP server is the same which is being used with the 4.2.0 installs so
should not be any issue with that.
We'll keep an eye on it and te
Yup, sure is.
On Fri, 30 Sep 2011 00:42:44 +0300, George Niculae wrote:
> On Fri, Sep 30, 2011 at 12:39 AM, m...@grounded.net
> wrote:
>> We are receiving VM notifications but not VM attachments on a 4.4.0
>> system.
>> I am trying to find the SMTP settings to that I
Some routers are more trouble than others when dealing with remotes. I found
that netgear routers were one of the worse.
Port forwarding doesn't need to be on but you need to make sure the router
doesn't have ALG enabled.
What I do now for remotes is to install my own Linksys RT31P2-NA in front
We are receiving VM notifications but not VM attachments on a 4.4.0 system.
I am trying to find the SMTP settings to that I can change them to use a relay
rather than sending direct.
The system is using the EmailFormats.properties placed in the
/etc/sipxpbx/sipxivr directory, perhaps it has som
> Receive only Mike.
> Look at t38faxvoip.com for a comm port product that you may be able to use.
> I have a customer using it with 3cX.
Oh darn, misunderstood then. Thought Steve was saying this is here or coming?
The avantfax server does the trick it's just that not having to maintain
separate
Say What???
Even while looking for fax related news, I sure as heck never found that
information? That would solve a world of hurt for me. I have to maintain a
separate asterisk/hylafax/avantfax setup so that users can send/receive faxes.
We have to manually create and delete accounts as needed,
> trunk). You can also take an ATA device and register it to sipx and send
> out via a t.38 trunk connected through sipx. I have done this myself using
> a patton fxs gateway.
Just to add to this, I have several people who have their fax systems connected
to a simple LinkSys ATA and those work p
This SO reminds me of when I was trying to make the argument of sipx needing to
do in/out faxing in order to really be a unified communications solution.
On Thu, 29 Sep 2011 14:46:29 -0400, Tony Graziano wrote:
> Yes, except I have to run outlook to make that happen. I'd prefer it
> to occur at a
On Thu, 22 Sep 2011 17:27:28 -0400, Tony Graziano wrote:
> not yet, see http://track.sipfoundry.org/browse/XX-9776
I have to believe that many would like that choice. In our case, everyone using
this feature wants it in pdf format.
Thanks for the heads up.
Any option to receive PDF faxes instead of TIFF?
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On Fri, 16 Sep 2011 09:54:23 -0700, Todd Hodgen wrote:
> Mike, we seem to be running in circles in this discussion that doesn't end.
I don't believe so. As I said early on, I'm going with the phones simply
registering to our main site instead of putting a server at the remote.
This thread is sti
> in your case you need a tunnel type ans ipsec take 2 minutes to setup with
> pfsense at both sides.
It would be pretty cool if pfsense ran on the Linksys RT31P2-NA routers. I've
got about a hundred of them.
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Someone mentioned something along the lines of the remote office not even
needing a sipx box.
The office is running polycom 550 phones but I've never done a remote polycom
setup.
My limited knowledge of this would be that I should take the phones, update
them to a certain level, bring them back
fsense at both sides.
> what you do with it is totally up to you.
I'll set up a little test and see how things go some time soon. Sounds like a
solution for other things that come up now and then too.
Thanks very much.
> On Sep 14, 2011 4:29 PM, "m...@grounded.net" wrote:
I don't recall seeing IPSEC mentioned but it's an idea.
> vpn and host use the other end. we do that all the time. it doesnt
> matter if you are trying to use the gateway, or sipx. it's the same
> "network" as private, but routed.
I love it. I'll have to set up a test and give it a try.
> with a
On Wed, 14 Sep 2011 11:16:44 -0700, Todd Hodgen wrote:
> Setup an IPSEC tunnel between the two locations - you should have all of the
> sipXecs features. Except 911 calls dispatching to the right location. Put
> in a local FXO gateway for that.
Cool, local VPN after all! Yet another possible so
On Wed, 14 Sep 2011 11:40:38 +0200, pscheep...@epo.org wrote:
> My view on the world: Simplicity for an
> extra 4 extensions (as in "4 trunks") in a remote location can
> be achieved without an extra SipX.
So you're saying that for only 4 simultaneous calls, to considering simply
using remote SIP
I am very sure this thread will be helpful to others at some point. There are a
variety of ways to do this.
Since it's only 4 trunks, it's not worth the complexity of buying all sorts of
hardware.
The simplest approach seems to be 4 SIP trunks and not much else.
On the other hand, since we al
Hijack the thread, it's all useful information anyhow. However, practically
everything mentioned is overkill :).
It's just a remote 4 trunk sipx setup which will use G.729 so there won't be
much bandwidth involved.
Using SIP trunks is the simplest install but I'd love to test a vpn back to the
, and what numbers
> to point to which sipx server for DID.
>
> On Sun, Sep 11, 2011 at 5:28 PM, m...@grounded.net
> wrote:
>> On Sun, 11 Sep 2011 13:54:11 -0400, Tony Graziano wrote:
>> either make the remote branch remote users or connect the two sites via
>> vpn
&g
dial plan to handle the other sipdomain equally, and what numbers
> to point to which sipx server for DID.
>
> On Sun, Sep 11, 2011 at 5:28 PM, m...@grounded.net
> wrote:
>> On Sun, 11 Sep 2011 13:54:11 -0400, Tony Graziano wrote:
>> either make the remote branch remote users o
after now.
> On Sep 11, 2011 1:34 PM, "m...@grounded.net" wrote:
>>> without explaining what you have now in a way other than "a really good
>>> LD
>>> deal".
>>>
>> LD meaning that with our PRI services, we get a good long dist
w the remote and local servers to communicate via
firewall rule.
Mike
> On Sat, Sep 10, 2011 at 1:56 PM, m...@grounded.net
> wrote:
>
>> Sorry if I've posted this multiple times, had some mailer problems and
>> don't see it in the list so am repostin
Sorry if I've posted this multiple times, had some mailer problems and don't
see it in the list so am reposting.
---
I have to install a small system at a remote location. The system will only
have about 4 trucks so makes no sense to order lines/gateway etc.
What I was wondering about is either
n the way to go off hook an
> answer every call and incur a toll charge on every call.
> On Sep 2, 2011 6:40 PM, "m...@grounded.net" wrote:
>>> No, and thus noo.
>>>
>> Dang and dang dang but thanks! Saved me many hours of continued looking.
>>
>>
> No, and thus noo.
Dang and dang dang but thanks! Saved me many hours of continued looking.
> On Sep 2, 2011 4:28 PM, "m...@grounded.net" wrote:
>> Mediant 2000, running 5.6
>> Sipxecs 4.4
>>
>> I cannot find this anywhere, probably because not many are
Mediant 2000, running 5.6
Sipxecs 4.4
I cannot find this anywhere, probably because not many are doing it or can't be
done so thought I would ask here since there are lots of Mediant users here.
I think it is a two part question.
One, can one sipx DID handle both voice and fax receipt? It does
> We are jotting down some notes for gconf changes on RHEL6/Fedora 15,
> will that help? The Telify website has good info on how to get it
> working on Windows.
I see it now. Thanks very much for pointing this to everyone.
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si
> the TLD. However, it does take some Desktop-fu to get working.
Any chance you've come across anything that gives some example/s on how to use
it with sipx or that desktop-fu :)
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> I made an interface for the XML browsers on Polycom and Snom phones,
> that goes kinda in the same direction and a mobile browser page would
> probably use a similar principle. I just added pages to SipXconfig that
> display properly on those phones, no SOAP or REST necessary that way. To
> me th
2011 at 5:21 PM, Tony Graziano
> wrote:
>
>> I have a mobile developer looking at it, but there are some missing
>> elements in soap from a user interface that makes it incomplete at this
>> time.
>>
>>
>> On Wed, Aug 10, 2011 at 5:16 PM, m...@grounded.net
ted with customizing the interface for standard browsers but
not for smaller devices such as mobile phones and other devices.
> Also, did you post that from an iPhone? Autocorrect me if I'm wrong :-P
From a pc.
>
> On Wed, Aug 10, 2011 at 2:47 PM, m...@grounded.net
> wrote:
> I think he means a mobile browser version of the admin panel so users can
> modify their settings and listen to voicemail. Something for iPhone or
> android.
Correct. I was pretty sure that was clear :).
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Has anyone ever built a wep or mobile interface for sipx or is there one
available somewhere and I've not seen it?
That is something which sipx could very much use so that remote users could get
in and properly display to manage.
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an coming this week.
Thanks.
>
> Mike
> On Tue, Aug 9, 2011 at 11:35 PM, m...@grounded.net
> wrote:
>
>> Old thread from last year - any way to do this in 4.4?
>>
> Customer wants to call forward line to cell phone but see their DID when
> calls come in from that
Old thread from last year - any way to do this in 4.4?
Customer wants to call forward line to cell phone but see their DID when calls
come in from that number.
Here's the message link - not sure what protocol is when replying to a year old
message, or do you start a new thread?
change caller i
hing I'd ask in return is that I get a documented procedure of
> what is done on the your side to configure SIPx. Thanks.
>
>
> On 8/5/2011 12:53 PM, m...@grounded.net wrote:
>> We did receive more information and the person is aware that there are
>> tariffs, laws to be
I'd recommend
> using something like G.729, since bandwidth can get pretty expensive
> over in Africa.
>
> The only thing I'd ask in return is that I get a documented procedure of
> what is done on the your side to configure SIPx. Thanks.
>
>
> On 8/5/2011 12:53 PM, m...
.
On Fri, 5 Aug 2011 08:26:45 +0100, shouldbe q931 wrote:
> On Thu, Aug 4, 2011 at 12:57 AM, m...@grounded.net
> wrote:
>> Thought I sent this privately. As always doing too many things at once.
>>
>> We don't know much else at this time, only that someone wants to kn
using the state
> sponsored telco and side stepping toll fees... "crossing an operator"
> might simply mean having a telco tattle on you because you sidestepped
> their revenue stream also...
>
> On Wed, Aug 3, 2011 at 5:17 PM, m...@grounded.net wrote:
>> Hi,
>&g
Hi,
I thought I would contact you direct on this at this point because it's kind of
getting away from sipx.
> The implementation shouldn't be difficult. You have to really be sure
> what you are getting into on this, if the customer is short on detail on
> use I'd be wary, you probably don't w
> You really shouldn't need multiple servers, your channel density is
> quite low (4-8 channels?), and would only really add as much stress to
> your system as an equivalent number of desk set calls. Will the gateways
> be residing in one country or multiple countries?
The calls would all originat
I'll pass this along to Kelly who will give it a shot and I'll update this
thread to help anyone else needing this.
Wish there was a way to get rid of a typo "Please enter mailbox number please'.
Or maybe I should have added to it,..
'Hello? Please enter a number. Any number would be good. Jus
> to be trying to run traffic over 3G or WiFi from the mobile, GSM has a
> huge footprint worldwide, it's not going to be matched by any other
> coverage anytime soon.
>
> On 8/3/2011 12:17 AM, m...@grounded.net wrote:
>> I am still waiting for more information. The way
Searching has not lead to finding any information on doing the following. Is
the following possible.
We need to have a main number that people call and after a greeting, are given
the option to key in a persons extension.
Thanks very much.
Mike
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I am still waiting for more information. The way it would work is that calls
would originate from one city and be routed to several other cities where all
of the remote users would be on cell phones.
Thanks.
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I have an application where I need to set up a local sipx server and a remote
for intercommunication's.
However, on the remote side, there are no phone lines available, only cell
phones so need something between sipx and gsm that would take incoming calls
and forward them over their local cell
his level of
> integration it was quite effective. I think I had a few post about this
> back then.
>
> Cost if remember was under 2k.
>
> -M
>
>>>> "m...@grounded.net" 06/24/11 10:32 AM >>>
> I doubt I can get into building RPMs but I'll res
s Hubler wrote:
> On Thu, Jun 23, 2011 at 10:29 AM, m...@grounded.net
> wrote:
>> Tony, you're heavy into development. Have you heard of any RTMP (like
>> red5) apps that would work with sipx in terms of being able to have a
>> browser based phone? I know of a few apps
Hi Tony,
Guessing you would know who maintains the list. My SMTP server is reporting
that the list server host drive is full so is rejecting emails. Not sure who to
contact about it that would get immediate attention but figured you would.
Mike
On Wed, 25 May 2011 12:14:40 -0400, Tony Graz
Testing to see if I can reach the list.
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wrote:
> I use om independently a test just do a sipx conference call for audio.
> On Jun 21, 2011 6:47 PM, "m...@grounded.net" wrote:
>> I'm on the OM list and when I asked if OM works with sipx, I was given
>> only two servers that it runs on.
>> A while back
I'm on the OM list and when I asked if OM works with sipx, I was given only two
servers that it runs on.
A while back, there was talk about integrating bbb, then om, but neither seems
to have made it or I've missed it in the features list.
Anyways, wondering if anyone knows if OM can indeed wor
ed the server yet so maybe there's something else waiting but I
kinda doubt it.
I'll run an update again soon since I've seen fixed being posted which I assume
are still making it to 4.2.1.
Mike
On Sat, 28 May 2011 21:15:39 -0400, Gerald Drouillard wrote:
> On 5/25/2011 11:
Hi Users,
Does anyone know if it is possible to make the default/Home "tab" on
the users portal be the "My Information Tab" instead of the voice-mail
tab without rebuilding sipxconfig?
Thanks!
Kyle
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Wasn't sure how else to put this.
I have a requirement for a system that could handle 2,000 or so calls per day.
However, depending on weather, the system might have to scale up to ~20,000 per
hour.
I was thinking perhaps calculating the total amount of traffic for bandwidth
needed then maybe u
> I never recommend restoring and older version to a newer one. others may
> disagree.
Ya, I've seen your posts about this. Just kinda stuck and looking for the best
option at this point. If a backup/restore works, that would be my best
solution. I can just see where things will head if I start
Anyone try it? Is it possible?
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On Wed, 25 May 2011 14:46:39 -0400, Tony Graziano wrote:
> i think you might do well to test it against your trunks before deploying
> it.
I'll go that route if I have no other options I guess.
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L
Based on the ones you've done, do you think it would be safe to upgrade the
4.2.1 to 4.4.0.
On Wed, 25 May 2011 14:23:58 -0400, Tony Graziano wrote:
> you would never undo the last update.
> On May 25, 2011 1:18 PM, "m...@grounded.net" wrote:
>> Wait now, are you
Wait now, are you guys talking about doing an upgrade from 4.2.1 to 4.4.0?
I just wanted to update a 4.2.1 system but I don't have a problem upgrading if
everyone feels it's safe to do so.
My only next concern is that I've already run the update earlier, still not
sure if everything will be ok
directly between the
two. Just wanted to get the latest updates for 4.2.1 since I had not updated in
a while.
> On Wed, May 25, 2011 at 12:34 PM, m...@grounded.net
> wrote:
>
>>> ive always upgraded by removing the packages and updating to 4.4.0x.
>>>
>>> Do you m
> ive always upgraded by removing the packages and updating to 4.4.0x.
>>Do you mean a full re-install?
> no, just the complaining freeswitch package and dependencies (which are all
> codecs). they reinstall when you update.
Well, interesting then because my system didn't get updated to 4.4.0 w
On Wed, 25 May 2011 12:10:46 -0400, Tony Graziano wrote:
> ive always upgraded by removing the packages and updating to 4.4.0x.
Do you mean a full re-install?
Would I simply yum remove sipx?
How would I make sure that users/settings/vm's were not lost?
On the other hand, have you tried a backup/
So the real question is, since sipx-freeswitch was 32bit anyhow, it didn't get
updated but is that a problem? Will it still work with the rest of the system
which did get updated?
If so, then it's a non issue, I simply reboot, all the new things are put into
use but I don't get the 64 latest sip
rpm -ql sipx-freeswitch gives me;
/etc/ld.so.conf.d/freeswitch.ld.so.conf
/etc/monit.d
/etc/monit.d/freeswitch.monitrc
/etc/sysconfig/freeswitch
/usr/local/freeswitch/*
> then do
> rpm -e --justdb sipx-freeswitch-1.0.5-17188.16739.2.i386
Isn't there a problem with rpm or yum not always knowing w
> unfortunately sipxecs 4.2.1 yum repo mixes 32 and 64 bit archs and
> although the OS is supposed to do the right thing, often folks get
> into situations were system has 32 and 64 bit rpms installed. i'm not
> sure how it gets into this situation.
I didn't come across any problems when building
> Why would the package contain a 64 bit
> file when a 32 bit file is installed?
> Is this a 32 or 64 bit machine?
2.6.18-194.17.1.el5 #1 SMP Wed Sep 29 12:50:31 EDT 2010 x86_64 x86_64 x86_64
GNU/Linux
64bit hardware. It was built using the repo, not from ISO.
_
order to update properly i think...
>
> On Wed, May 25, 2011 at 11:11 AM, m...@grounded.net
> wrote:
>> I haven't updated my 4.2.1 in a while so when doing it I get the
>> following error.
> Thought I would check with the list before rebooting, just in case.
>
>
&
I haven't updated my 4.2.1 in a while so when doing it I get the following
error.
Thought I would check with the list before rebooting, just in case.
Transaction Check Error:
file /usr/local/freeswitch/bin/fsxs from install of
sipx-freeswitch-1.0.5-17188.16739.2.i386 conflicts with file from
Last I heard, quite some time ago, the talk was that both pdf and tiff would be
supported for fax.
Is this still the case and if it is, how does one change the output to pdf?
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Is EmailFormats.properties still needed in the sipxivr for custom output?
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> There are a lot of other details in their network that you havent
> described as well type of switch, what do they do with the PRI on the
> other end, etc. But, like the rest, its not important to what you are
> needing to deal with.
Sorry, maybe the replies are out of sync. I recently sent
As additional information, I am able to receive faxes into my asterisk based
fax server and sipx now.
I added a T.38 Relay profile option for DID's being used for inbound faxing to
sipx.
So the question remains, even though it's working, the setup I've explained
from my provider doesn't sound r
On Sun, 24 Apr 2011 19:08:29 -0700, Todd Hodgen wrote:
> I think Mike is misunderstanding and mixing up protocols here.
I've always admitted I just get by on this stuff, I'm no voice/telco engineer,
it's just one of the many things I need to have a basic understanding of.
Indeed, I sometimes don
> I don't understand. If it is t.38 end-to-end, it would work by routing thh
> specific did's to the new system, it will work. If the provided is not
> truly supporting t.38 all bets are off.
That's the thing, they aren't providing SIP services to the end user, they are
only using it between thei
> The provider is not offering true t.38. If thy send it via g711 to the t1
> and convert it locally it will be very unreliable unless it originates and
> gets sent all the way thru as t.38.
I want to make sure I'm being clear because I'll have to find a way around this
problem.
So, even if I tol
27;t have T.38 on the fax server and needed to make sure faxes were
not affected.
It's kind of a weird setup since it's sort of using SIP trunks yet it's not to
the customer, it's only to their own gear. What a mess.
> On Apr 24, 2011 7:44 PM, "m...@grounded.ne
tandard PRI trunk to the mediant so faxes and
calls are coming in over G.711.
Can the mediant work any magic that sipx would accept?
> On Sun, Apr 24, 2011 at 6:35 PM, m...@grounded.net
> wrote:
>> The wiki seems to touch on lots of troubleshooting but I can't seem to
>> f
The wiki seems to touch on lots of troubleshooting but I can't seem to find an
answer on the following.
The new 4.4.0 system is connected to what sounds to me like a weird hybrid
setup.
The provider has a T1 line between their network and the system site.
At the site, they are using a router wh
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