> Hi Dale,
>
> I know they are, unfortunately, following the "wide" SIP standard.
>
> Since media is relayed though sipX, Media Relay, when
> sipXBridge is used, I was wondering if it was possible to get
> an event when first media arrives? This in not relevant for a
> normal sipX deployment,
voip.ms
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Ken Fulmer
> Sent: Tuesday, July 20, 2010 9:43 AM
> To: sipx-users@list.sipfoundry.org
> Subject: [sipx-users] VoIP Providers
>
> We've been usin
> Ok. So. I'm trying to remotely re-provision the phone and
> downgrade the firmware. I figured i'd try TFTP first since I
> know that works in the local office. So I forwarded port 69
> on the local router to the internal IP of my sipX. Went into
> the network setup menu on the phone and
> On Fri, Jun 25, 2010 at 1:22 PM, JOLY, ROBERT (ROBERT)
> wrote:
> > The bug introduced by 18076 got undone in revision 18944 so
> presumably hunt group call pick up started working again
> after that revision...
>
> Robert,
>
> This checkin seems to exclusi
If call pickup to a hunt group ever worked then it would have been definitely
broken by the commit in revision 18076. The bug introduced by 18076 got undone
in revision 18944 so presumably hunt group call pick up started working again
after that revision... What rev. Are you using?
bob
> ---
> Tony and All - Thanks for the quick and thorough reply. I
> read the Remote User config carefully and realize it might
> work for us - I had some misunderstanding about the pinholes
> before. It seems a good in-box option and easy to use. I will
> try it out in NAT environ. At this point, I a
> We are trying out sipXecs for internal usage. An important
> factor for us is NAT traversal. We have some experience with
> Microsoft OCS and it uses ICE for NAT traversal. It seems a
> good way to handle this.
>
> I checked sipXecs docs and it is not very clear about how to
> configure STUN
> On Wed, Jun 16, 2010 at 11:07 AM, Martin Steinmann
> wrote:
> >>
> >>Well, you can clamp down the codec to a single supported
> codec such as
> >>G711 ( i.e. filter the request and response SDP to that single
> >>supported codec and never re-invite subsequently ).
> Asterisk has that
> >>op
> On Wed, Jun 16, 2010 at 7:46 AM, Martin Steinmann
> wrote:
> >>
> >> As you know we have a long
> >>> history of 'protocol purism', which generally has not
> served us well.
> >>
> >>
> >>This is really not the same thing. I am talking about basic errors
> >>with the implementation of the pr
not answering the question...
>
> On Fri, Jun 11, 2010 at 4:38 PM, JOLY, ROBERT (ROBERT)
> wrote:
> >
> >
> >> -Original Message-
> >> From: sipx-users-boun...@list.sipfoundry.org
> >> [mailto:sipx-users-boun...@list.sipfoundry.org] On
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Tony Graziano
> Sent: Friday, June 11, 2010 11:40 AM
> To: Paul Scheepens
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] IM group not d
> I am trying to work out what would be the best method to set
> up presence.
> I've learned quite a lot already and will try to make some
> wiki doc in the future, but..
>
> One of the things I played with was Instant Messaging for a
> User Group.
> I created a UserGroup "GroupChat", in Use
> Hi
>
> Click-to-call sends an initial INVITE with an SDP offer that
> only includes G.711u as audio codec. Once this call is
> answered, a REFER is then sent which effectively blind
> transfers the call to the remote party. During the transfer
> the two parties can negotiate new media howe
her to re-design it - would
be kinda hard to fix that one with the scheme in place. A re-design of
anything is usually not trivial work but I would not say it is massive either.
>
> For now I'll work with FQDNs to get sipX to do what I want it to do.
>
> On 06/10/2010 0
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Josh Patten
> Sent: Wednesday, June 09, 2010 4:51 PM
> To: sipx-users@list.sipfoundry.org
> Subject: [sipx-users] Use one IP address for multiple
> gate
s of the callee get used.
>
> I'll look into fixing that.
>
> bob
>
> > -Original Message-
> > From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> > Sent: Monday, May 31, 2010 11:39 AM
> > To: JOLY, ROBERT (ROBERT); jpat...@co.brazos
Message-
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: Monday, May 31, 2010 11:39 AM
> To: JOLY, ROBERT (ROBERT); jpat...@co.brazos.tx.us;
> sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] tel: forwarding in IM status
> message not working
>
> thanks,
> I'll try it out an let you know.
>
> It matters if you have (theoretical speaking) the remote user
> trying to call an outside number, then the media goes like this :
>
> remote user internet - sipx -- internet --
> voip provider
>
> and i want the media to go dire
>
> OK I have found this only works on internal calls meaning:
>
> set forward to outside number (eg 5551212) for IM user (lets
> say 6001) call from internal extension to 6001 - the call is
> forwarded as it should be call from outside line to 6001 -
> the call is not forwarded
>
> Could thi
> Ok, let me see if I can get more granular with my
> explanation. I was in a hurry before and didn't give the
> detail I should have. I apologize.
Not a problem. Most times, it's hard to converge on the right solution without
the full background.
>
> I am running sipXecs 4.2 with Polycom 6
>
> On Sun, May 16, 2010 at 1:19 PM, Gabe Casey
> wrote:
> > I am seeing a strange issue periodically in 4.2 where a call is
> > getting stuck in the system and recalling the outside
> number every 30
> > min. From the Logs each call has the same Call-ID as the
> original. Anyone else seen t
> When I upgraded to 4.2, I did a quick test of the IM system.
> I sent a message from my phone to my office manager's phone.
> The problem, we can't get rid of that message. I've tried
> rebooting the phone and we've cleared all the messages on the phone.
>
>
>
> Suggestions?
Yes, pleas
>
> I am using Polycom 331 and Polycom 550 at a remote site. When
> I call an extension, it rings for half a ring, then stops. I
> have captured Wireshark, and I think one problem might be
> that the INVITE message is being fragmented because they it
> is 1674 bytes in size, and are marked wit
> We are transcoding 711 to 729 for calls to the PSTN. Music on
> Hold sounds choppy to the remote side.
>
>
>
> Is there a way to tweak the system so it will accept wav
> files recorded in a more 729 friendly format?
G.729 exploits attributes of the human speech to do compression while
m
> On Wed, 12 May 2010, Scott Richesson wrote:
>
> > Yup... I had that problem intermittently. Downgrading to
> SIP 3.2.2 fixed the problem.
> >
> > I believe this is the problem:
> > http://track.sipfoundry.org/browse/XTRN-1046
> >
> > My phone experiencing the problem was not a remote worker,
tely generating a bad ACK. That bug effectively makes the ACK impossible
to route back to the called party.
>
> sdm
>
> > -Original Message-
> > From: JOLY, ROBERT (ROBERT) [mailto:rj...@avaya.com]
> > Sent: Wednesday, May 12, 2010 9:08 AM
> > To:
as well.
>
> sdm
>
>
> On May 11, 2010, at 4:21 PM, "JOLY, ROBERT \(ROBERT\)"
> wrote:
>
> >>
> >> Posting failedcase.pcap for review.
> >>
> >> Here's a brief on the players in the trace:
> >>
e put in the
Record-Route as opposed to the other end's contact. Even if you fix #1, #2 is
going to be a killer.
Let me know,
bob
>
> sdm
>
> > -Original Message-
> > From: JOLY, ROBERT (ROBERT) [mailto:rj...@avaya.com]
> > Sent: Tuesday, May 11, 2010 10
>
> I have a newly installed sipxecs 4.2 system (replaced a
> previously installed v3.10 system). System is behind a NAT
> firewall. All works as expected (internal extension to
> extension calls, outbound calls, inbound calls, etc). I have
> also followed the v4.2 remote worker config guide
> Enabling IM at the group level finally allowed my IM
> clients (Spark, Miranda, Adium) to start working.
> However, I have a problem. My users were assigned to various
> groups: administrators, sales, support, etc.
> Members of the support group only see other members of the
> same group as
> On 5/6/10 12:37 PM, Jermaine Pinder wrote:
>
>
> I have a sipx server setup on an ethernet 3MB line form
> an ISP that also provides
> ADSL to multiple locations for my remote phones.
>
> The problem I'm having is all the phones registers as
> No nat (@x.x.x.x
>
> I believe there is a cisco issue. We had another customer
> report this issue with a set of cisco handsets yesterday. I
> have not heard back from our techs what they found yet, but
> the issue reported to us was not via sipxbridge but for a
> remote worker calling to an internal extensi
Josh,
I do not believe that there is a way to disable this through config files but
I'll let sipXconfig people comment.
In the meantime, if you absolutely *must* get rid of default gravatars, there
may be some other alternatives:
1- Actually make use the gravatar integration and a create free
> http://wiki.sipfoundry.org/display/xecsuserV4r2/Upgrade+or+Ins
> tall+Planning+for+4.2+and+XMPP
Very nice, Tony - thank you. And bonus points for the humor...
>
>
> On Thu, Apr 29, 2010 at 4:20 PM, Scott Lawrence
> wrote:
>
>
> On Thu, 2010-04-29 at 14:33 -0400, Tony Graziano wrot
>
> I think I'll avoid doing that and let the phones reregister
> over time.
> Shouldn't be more than about 10 or so minutes for them to
> reregister I would think.
For remote users, worst case is 5 minutes. For non-remote users, it's 60
minutes.
> On 4/30/2010 8:33 AM, Scott Lawrence wrot
> Sometime i see that one of our phones cannot refresh its
> registration and its registration expires.
What do you mean by 'cannot'? Does it send a registration buit fails the
challenge or does send no REGISTER at all before the expiry?
> After some seconds
> (maybe around 5 seconds ) phone
> and registration refresh time
>
> On Fri, 2010-04-30 at 14:33 +0300, an...@iguanait.com wrote:
> >
> >
> > Do you know why sipxecs has chosen 300 seconds as minimum
> by default?
>
> Registration can create a lot of load on the system - if you
> had a few hundred phones re-registering every
openfire-3.6.4-3.noarch.rpm
>
> -Original Message-
> From: JOLY, ROBERT (ROBERT) [mailto:rj...@avaya.com]
> Sent: Thursday, April 29, 2010 9:52 PM
> To: Wen Jun; sipx-users@list.sipfoundry.org
> Cc: 'jun,wen'
> Subject: RE: [sipx-users] sipXopenfire service
> I already know one of these answers, but they seem related
> and I wanted to make sure my logic was correct. I don;t
> normally cross post, but wanted a broader audience if I can
> get it because it will come up be be a question for a lot of
> people upgrading to 4.2, and helps with proper pl
>
> I am seeing what I think is some odd behavior with Openfire,
> but then it may just be me.
>
> I have installed a sipxecs 4.2 test system on Centos 5.4
> x86_64 using the YUM repos. sipxecs-setup has been run to do
> the base configuration.
> DNS Advisor reports no issues and the only Con
>
> From: jun,wen [mailto:jun@msn.com]
> Sent: Wednesday, April 28, 2010 1:15 PM
> To: 'sipx-users@list.sipfoundry.org'
> Subject: sipXopenfire service failed
>
>
> Hi, I am running sipX 4.2.0, build 18724. The sipXopenfire
> cannot be started which gives me the following standard error -
m/upgrade-to-bria-3.0.html>
>
> Then you can upgrade with a discount.
> As we bought most of our licenses after December 1st, 2009 we
> can actually have a free upgrade :o)
>
> Best regards / Mit freundlichen Grüßen / Sincères salutations
>
> Paul Scheepens
>
> "JOLY,
>
> > >
> > > 5. Which value is good to be set as Refresh time on the phone? I
> > > know that default hardcoded minimum time is 300.
> > > Does it make sence to put a lower value or a high value?
> >
> > A hardcoded minimum of 300 is not enough. Phone is expected to be
> > able to go lower.
> anything new for sipx 4.2?
The fundamental procedure has not changed but the new wiki goes more in depth
than the one it replaces.
>
>
> On 2/26/10 9:07 AM, Robert Joly wrote:
>
> I have created a new and improved remote user
> information page in the
> new wiki. This new one
om Bria to Bria 3.
>At this time pricing has not been set for this upgrade.
>
>
>Regards,
>
>Customer Support
>CounterPath Corporation
>
> I am still waiting for "some upgrade path" other then just
> buying Bria 3.
>
> Best regards
Very good analysis of the problem. You have hit a limitation of Counterpath's
implementation of the XMPP protocol. We have made Counterpath aware of the
limitation a while back and they are working on it.
Basically, if your XMPP domain is not the same as your FQDN, Bria will fail to
connect t
> Hi,
>
> We use sipxecs 4.0.4 and it is not behind NAT. Nat Traversal
> feature for remote workers is enabled.
>
> I have some questions related with outbound proxy and
> registration refresh time (registration expiration time).
>
> Questions related to outbound proxy.
>
> 1. How do i test t
> Am I correct in assuming that multiple users are a remote
> site are hair-pinned at the sipXecs server? If not, great.
> If so, is there a feature request we should be considering to
> keep that media path local?
Navigate to System->Servers->[cliack on server]->Media Relay->Media Relay
Tem
> When I call from user1 to user2 and they both have IM
> accounts logged in and enabled, I can see user1 shows on the
> phone to user two. I do not see any changes for user2.
>
> Is IM presence tied to phone events or simply the presence
> server in sipx?
The IM presence ties into the RLS ser
> OK, I'm trying to set up Pidgin.
>
> I have a "sandbox" DMZ for testing. On this DMZ are the
> sipXecs server (4.2), A Windows PC (for admin and to run
> Pidgin), and a phone (Polycom IP550).
>
> I did a fresh install of 4.2, told sipXecs to be the DNS and
> DHCP server, and imported the us
> I am curious as to whether the following "should" work in 4.2
> under the following:
>
> sipxecs
> server|Internet|<---firewall---> branch, users: thing1 and thing2>
This is definitely a scenario that is working. Support for this kind of
deployment was one of the major just
> yes and yes. Pidgin works fine, so does spark.
Just tried Pandion against my 2 sipXecs 4.2 boxes.
I can register fine against the sipXecs that does not use DNS SRV but I
*CANNOT* register it against the sipXecs that uses DNS SRVs.
My investigation shows that DNS SRV records are set up properl
> 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 us
> they are aware of it and working on a fix.
>
> As far as we know right now, this will not be a problem
> except with phones that are on networks that significantly
> delay responses from the phone (we found it with one of our
> remote users who's got a slow DSL link). The symptom you'll
> s
> Our issue with calls suddenly disconnecting after 18 secs has
> been fixed.
>
> However, it appears another problem that I initially thought
> was linked to this is not. We cannot seem to call another
> extension on a different location. Nor can we transfer calls
> between phones on differ
> Hi, All,
>
> I am running 4.0.4-018580 in my lab and want to test
> Aggressive mode in Media Relay. My sipx stays in 192.168.2.4
> and two of my test end points reside respectively in
> 192.168.2.31 and 10.33.150.150.
>
> When I turn on the Aggressive mode in Medial Relay and then
> make a
>
> Also I don't see any 64 bit RPM's. perhaps someone got trigger happy?
What you see is part of the preparation work required to create the 4.2 release
but we do not have our golden load yet. We are all pretty excited about
releasing 4.2 because it is a big milestone and trust me that it wil
> +1
Adding one user to the list would not significantly improve things. I would
say more like '+30' :)
>
> On 4/14/2010 11:11 AM, m...@grounded.net wrote:
> > Is there some way of making the lists display more items in
> the Users listings for example?
> > Only 20 at a time show up, I would
> Catchy subject eh :).
>
> Anyhow, in the Internet Calling section, then then Intranet
> Subnets section, in my case, I have 192.168.0.0/16.
> Is there any chance that sipx might be unhappy with this mask
> and might prefer a 24, even keeping SIP devices in the same
> segment rather than a 16?
ould share with us your
Cisco router model then I could add it to the (currently empty) router list
with ALGs we maintain in our Wiki.
> On Thu, Mar 25, 2010 at 8:43 AM, JOLY, ROBERT (ROBERT)
> wrote:
> >>
> >> To me that looks like whatever router you are using in
>
> The remote can call out. We tested by making a call to a cell phone.
> Using an internal to call the remote leads to ringing,
> nothing else, not even vm.
> Using an external phone, cell phone, calling the remote,
> leads to a fast busy.
I think the next logical step is gathering a snapshot fo
> >?This is a 'normal error'. ?The sipXproxy will periodically
> (every 20
> >secs.)
>
> That's what we gathered but wanted to confirm, thanks.
>
> >?Stay away from Internet calling if you can avoid it. ?Please visit
>
> Ok, will do that.
> ?
> >?Check out "Problem #2" in link above and check
> When I take a phone and plug it into a wireless remote
> network connection, the following keeps showing up in sipXproxy.log.
>
> "2010-03-25T19:48:50.052314Z":3099:INCOMING:INFO:uc.domain.com
> :SipClientUdp-8:B78FCB90:SipXProxy:"Read SIP
> message:\nRemote Host:xxx.xxx.xxx.215 Port:
>
> To me that looks like whatever router you are using in front
> of the phones is doing NAT compensation via a SIP ALG.
> Sometimes it's difficult to disable the ALG, depending on the router.
I concur with you. Either that or the phones have STUN enabled.
Does x-lite work properly when you
> >?You mean terminating a user agent's Registration ahead of its
> >scheduled ?expiry? ?If so, the only non-hacker way to
> achieve this is
> >for that user ?agent to terminate its own registration by
> sending a REGISTER with expiry=0.
>
> Yes, that's what I mean. So I guess it can be done,
> Is there any way at all of quickly expiring a user rather
> than having to wait?
You mean terminating a user agent's Registration ahead of its scheduled expiry?
If so, the only non-hacker way to achieve this is for that user agent to
terminate its own registration by sending a REGISTER with
> Thanks, more info / update:
>
>
> * Configuraiton is Blank
> * DHCP Information is being handed out correctly
> * Default Gateway is being set to the firewall and going
> across the VPN
> * Internet Subnet is set to 10.50.11.0/25
> * Enable NAT Traversal is 'unchecked' (whi
>
> rjoly
> I've sent you email. Even twice :)
I've seen it and I think we came to the same conclusion. Read on...
>
> Tony Graziano
>
> Quote:
> > What does the registration look like for the remote user in sipxecs?
>
> Account tab
> username: xxx
> password: xxx
> domain: yukon.cv.ua
> Reg
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> In-Reply-To:
> <6369cb70bfd88942b9705ac1e639a33821f6fbb...@dc-us1mbex4.global
> .avaya.com>
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <43569>
> Message-ID:
>
>
>
> Quote:
> >
>
> Content-Type: text/plain;
> charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <43551>
> Message-ID:
>
>
>
> Hi there,
>
> I have searched for solution of my problem but unfortunately
> I didn't find it yet.
>
>
> Hi,
>
> I think I followed all the 7 steps mentioned in
> http://wiki.sipfoundry.org/display/xecsuserV4r0/Remote+User+NA
> T+Traversal.
>
> I met the problem #3 - Remote Worker (gxp2010) registers and
> can make calls but media is blocked in both directions.
>
> But the remote worker (x-
all on hold,
keep it like that for about 15 seconds and then retrieve it from hold. Do you
have two-way speech path after you retrieve the held call?
>
> -Original Message-
> From: JOLY, ROBERT (ROBERT) [mailto:rj...@avaya.com]
> Sent: Tuesday, March 16, 2010 4:47 PM
> To:
ailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> Nathaniel Watkins
> Sent: Tuesday, March 16, 2010 4:48 PM
> To: JOLY, ROBERT (ROBERT); sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] issues from remote caller vpn into
> office - using sip trunks for oubound dialing
> I'm using polycom soundpoint ip 450's and most of my calling
> features are working properly.
>
> I have a setup with audiocodes analog lines inbound and
> broadvox outbound.
>
> If I try to conference via the polycom phone I can call the
> first leg and call the second leg, but when I press
>
> I have a remote user that uses a VPN connection to connect to
> the courthouse. They are using x-lite to make calls. I
> recently changed their dial-plan to route external calls via
> an ITSP. The call connects correctly, but after 30 seconds,
> the call disconnects.
>
>
>
> As a te
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