And a Linux version for Spark SIP Phone plugin will be developed ?
Thanks.
Barata7,
Iâve already had my accounts created on my SIP server. What Iâve tried to say was that SIP Phone plugin doesnât start. So the dialpad doesnât appear on Spark.
Eu tenho as contas jå criadas no Servidor Asterisk e tb jå fiz o mapeamento dessas contas no Servidor Openfire. Só que quando eu inicio o cliente spark, o Plugin SIP não inicializa e consequentemente o discador não aparece no cliente Spark. Eu uso o cliente Spark 2.5.8 em estaçÔes Windows.
Regards,
Eduardo
When you setup your sip phone mappings in openfire, did you click the test
button?
Did it pass.
If it failed, then your openfire server can
not see the asterisk server.
On Mon, December 3, 2007 11:38 am,
When I click the test button I recieve an error (Timeout), but I can make calls anyway (with some problem yet).
Trapper, your tip solve part of my problem.
Thanks.
Where are you making calls from. I thought the spark dialpad does not show
up?
In Spark, it looks like it doesnât work for me also. Until I call myself. The test in Openfire fails, and the dialpad in Spark doesnât ever start so this is really unusable, but I thought it was interesting that I had Spark running and Openfire and my SIP server (Zultys MX250) configured.
I was seeing the same trouble as most of you report- i.e. the test in Openfire admin failing and the dialpad in Spark was not appearing, I was about to give up, but then my phone rang and Spark notified me of the call. This is when I realized it was partially working⊠This has years of development work left (in my opinion) to be of any use to me. Zultys already has an IM client/SIP phone called MXIE that has been around for 5 years and works perfectly. It can integrate seamlessly with openfire for external IM and the SIP phone stuff just works.
I can see and use the Spark dialpad. Callings are completed, but in one machine we can hear nothing. I think it is a Windows problem.
But the connections, registrations, etc are all fine. The plugin is working in both Openfire and Spark.
Great, almost there
I had the same problem.
I resolved my issue by adding the
externip=mypublicip and the nat=yes in the sip.conf file.
I also had
to add nat=yes to my sif extensions conf file int he sip.d
directory.
On Mon, December 3, 2007 11:56 am, bjsvec
Iâll take a look in MXIE.
But the Spark plugin problem is because I was trying to use it in Linux, and It runs just in Windows and Mac.
It is a NAT issue.
VoIP connection are made on port 5060 but he audio
is a random high port.
Computers on the same LAN and Subnet will
work, but as soon connecting or passing though any other computers or
devices outside the same lan will fail, unless you tell asterisk what its
local nets are, and what its exterial ip is.
This way asterisk can
route packets effiently.
this is done i the sip.conf file on the
asterisk server.
Please add the localnet, nat and externip to your
sip.conf.
look at my example further up on this forum.
Travis
I use Twinkle in linux, and it works like a champ.
Travis
On
The computers are in the same network.
My problem is about codecs. If we call from Spark to a Polycom VoIP Phone, there is no sound in one direction. We have a similar problem in a client and we correct that reencoding the packets.
Weâll do this and report later.
I use Twinkle on linux. It works like a champ.
On Mon, December
I need a XMPP and SIP client, so Twinkle is not an option.
Check Zultys MXIE.
http://zultys.com/index.jsp?tab=productdetail&type=options&detail=summary-mxie&p roduct=mxie&site=US
It does XMPP, SIP and much more. It does require Zultys SIP server though, so you wonât use it with Asterisk.
The whole reason I was trying out Spark/Openfire/Asterisk was to see if there was any competition. I sell Zultys systems and people often say they are doing all these things for free with Asterisk, etc⊠Zultys is not free, but all this stuff just works and is very easy to configure and admin.
I need to use Asterisk and Openfire, so Iâll continue to try Spark and SIP plugin.
Thanks
Gizmo is supposed to do that.
And it now allows you to add your own
sip server.
But I have not got the jabber part if to work properly
yet.
This ttimeout issue ight be a firewall blocking access to your SIP Server, or multiple Network IPs in your Openfire Server.
Can you tell me if it matches any of this causes?
Best Regards,
Thiago
Firewall is not the problem. Maybe the question with IPs, cause we have 3 ethernet cards, but I configure asterisk with the outside IP.