[sip-comm] using Sip-Comm with Gizmo5.com / sipphone.com


#1

I want to switch from the gizmo sip client to somethine else and
Sip-Communicator looks the best to me (despite being pre-alpha).

Following Gizmo's setup instructions for VIOP phones, I set the proxy
host and registrar to proxy01.sipphone.com
I also set User ID and Auth ID to my SIP number, and Password to my
Gizmo password.

When I try to connect, I get a invalid password error.

Anyone have success with this service?

···

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#2

Hi Ron,

I have connected in the last days in a test using a DELL Mini-9 running
under Ubuntu 9.04 with proxy01.sipphone.com
It logged in and showed a green ON icon. Did not try a call.

I am having strange problems with WIN, so can not say anything about
WIN version.

Regards, Earl

Ron Wilson wrote:

···

I want to switch from the gizmo sip client to somethine else and
Sip-Communicator looks the best to me (despite being pre-alpha).

Following Gizmo's setup instructions for VIOP phones, I set the proxy
host and registrar to proxy01.sipphone.com
I also set User ID and Auth ID to my SIP number, and Password to my
Gizmo password.

When I try to connect, I get a invalid password error.

Anyone have success with this service?

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#3

Thanks. I'm running Mepis Linux 7 with Debian Lenny repositories.

I have attached a screen pic of my sip-comm settings.

FYI, I also tried my Gizmo account name, name@proxy01.sipphone.com and
name@gizmo5.com as user ID. Still did not work.

···

On Tue, Sep 1, 2009 at 3:48 AM, Earl<Large.Files@gmx.net> wrote:

I have connected in the last days in a test using a DELL Mini-9 running
under Ubuntu 9.04 with proxy01.sipphone.com
It logged in and showed a green ON icon. Did not try a call.


#4

Still having trouble.

I would have included the log file, but it is 24k. Is there some place
I can upload it to for people to look at?

I have attached a JPEG is the account configuration on the assumption
I am not putting the account information correctly.

I am having no trouble using the Gizmo client, but I would rather not
use that client.


#5

HI Im running windows and sip communicator. At ast i least i was. I entered my hotmail info but kept getting weird messages from unknown people.

I tried 2 sip: names i have but did workTThis is really difficult and theres no help on thi

···

Date: Tue, 1 Sep 2009 12:55:56 -0400
From: ronw.mrmx@gmail.com
To: users@sip-communicator.dev.java.net
Subject: Re: [sip-comm] Re: using Sip-Comm with Gizmo5.com / sipphone.com

On Tue, Sep 1, 2009 at 3:48 AM, Earl<Large.Files@gmx.net> wrote:
> I have connected in the last days in a test using a DELL Mini-9 running
> under Ubuntu 9.04 with proxy01.sipphone.com
> It logged in and showed a green ON icon. Did not try a call.

Thanks. I'm running Mepis Linux 7 with Debian Lenny repositories.

I have attached a screen pic of my sip-comm settings.

FYI, I also tried my Gizmo account name, name@proxy01.sipphone.com and
name@gizmo5.com as user ID. Still did not work.

_________________________________________________________________
Windows Live: Keep your friends up to date with what you do online.
http://windowslive.com/Campaign/SocialNetworking?ocid=PID23285::T:WLMTAGL:ON:WL:en-US:SI_SB_online:082009


#6

Hey Ron,

You may want to check the configuration instructions:

http://support.gizmoproject.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=409

It seems that the sipphone service expects you to use your phone number
(without attaching a domain name) as both a user id and an
authentication id.

Therefore, paste your 1747XXXXXXX number (only) into both the SIP ID and
authorization name fields.

You may also want to replace the gizmo.com registrar with
proxy01.sipphone.com. I don't think there's a registration service on
gizmo.com and I don't know whether the proxy would handle it gracefully.

The rest seems to be ok.

Hope this helps,
Emil

Ron Wilson wrote:

···

Still having trouble.

I would have included the log file, but it is 24k. Is there some place
I can upload it to for people to look at?

I have attached a JPEG is the account configuration on the assumption
I am not putting the account information correctly.

I am having no trouble using the Gizmo client, but I would rather not
use that client.

------------------------------------------------------------------------

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#7

I tried that. It does seem to connect sipphone, but when I make a
call, I ge no audio.

I have tested my audio with Audacity and know that it works. I made
sure to select the same ALSA device in sip-com that I selected in
Audacity.

Again, I would include the log file, but it is very large. Is there
somewhere I can post it so people can look at it?

···

On Wed, Sep 16, 2009 at 7:05 AM, Emil Ivov <emcho@sip-communicator.org> wrote:

It seems that the sipphone service expects you to use your phone number
(without attaching a domain name) as both a user id and an
authentication id.

Therefore, paste your 1747XXXXXXX number (only) into both the SIP ID and
authorization name fields.

You may also want to replace the gizmo.com registrar with
proxy01.sipphone.com. I don't think there's a registration service on
gizmo.com and I don't know whether the proxy would handle it gracefully.

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#8

If you don't have a website where to upload it, it's ok to send it here.

Oh, by the way, make sure you have alsa-oss installed.

Cheers,
Emil

Ron Wilson wrote:

···

I tried that. It does seem to connect sipphone, but when I make a
call, I ge no audio.

I have tested my audio with Audacity and know that it works. I made
sure to select the same ALSA device in sip-com that I selected in
Audacity.

Again, I would include the log file, but it is very large. Is there
somewhere I can post it so people can look at it?

On Wed, Sep 16, 2009 at 7:05 AM, Emil Ivov <emcho@sip-communicator.org> wrote:

It seems that the sipphone service expects you to use your phone number
(without attaching a domain name) as both a user id and an
authentication id.

Therefore, paste your 1747XXXXXXX number (only) into both the SIP ID and
authorization name fields.

You may also want to replace the gizmo.com registrar with
proxy01.sipphone.com. I don't think there's a registration service on
gizmo.com and I don't know whether the proxy would handle it gracefully.

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#9

Each time I tried to make a connection to my ekiga.net account with
sip-communicator I get a message telling me the connection was
impossible.

After looking in the issue tracker I found an issue which looks like
mine (#435) and which is marked solved. Unfortunately it seems not
solved in the last version of sip-communicator I use
(alpha3-nightly-build-2024).

This is bad since it renders the tool completely unusable ?

Any ideas how to fix it or what I did wrong ?

I tried the same version under Ubuntu 9.04 and MacOS X (10.5.8), with
the same bad results. I am behind a NAT (like almost all french people
using ADSL or cable broadband access).

···

--
Bruno Beaufils

Mobilisons-nous
Pour un service PUBLIC d'enseignement supérieur et de recherche
http://agp.univ-lille1.fr


#10

Bruno,

Bruno Beaufils wrote:

Each time I tried to make a connection to my ekiga.net account with
sip-communicator I get a message telling me the connection was
impossible.

Yes, for some reason ekiga.net seems to be ignoring our requests. What's
even stranger is that requests from another client do get responses.

After looking in the issue tracker I found an issue which looks like
mine (#435) and which is marked solved. Unfortunately it seems not
solved in the last version of sip-communicator I use
(alpha3-nightly-build-2024).

I believe this one is different. #435 is about ekiga.net refusing to
accept registrations from clients behind NATs that do not use STUN ...
for signalling. At one point they removed the constraint which was when
we closed the issue. However, the constraint seems to be back there
since I am getting the same 606 responses as before. I've therefore
reopened and reclosed the issue with the appropriate message.

https://sip-communicator.dev.java.net/issues/show_bug.cgi?id=435

This is bad since it renders the tool completely unusable ?

Well not really, unless ekiga.net is the only service that you are able
to use. There are a bunch of other providers around that work perfectly
well. You can try iptel.org, ippi.fr, sipphone.com and hundreds of
others. freephonie.net for French subscribers of the free.fr ITSP is
also working.

Hope this helps,
Emil

···

Any ideas how to fix it or what I did wrong ?

I tried the same version under Ubuntu 9.04 and MacOS X (10.5.8), with
the same bad results. I am behind a NAT (like almost all french people
using ADSL or cable broadband access).

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#11

I do have alsa-oss installed. And I have verified that /dev/dsp does
refer to my USB headset. (/dev/dsp1 refers to he internal audio. If
sip-com were using that, I should still here the sipphone.com greeting
message)

I have attached the log file.

sip-com.log (392 KB)

···

On Wed, Sep 16, 2009 at 7:39 AM, Emil Ivov <emcho@sip-communicator.org> wrote:

If you don't have a website where to upload it, it's ok to send it here.

Oh, by the way, make sure you have alsa-oss installed.

Cheers,
Emil

Ron Wilson wrote:

I tried that. It does seem to connect sipphone, but when I make a
call, I ge no audio.

I have tested my audio with Audacity and know that it works.


#12

Ron,

I'll try to have a look at this in the following days.

However until then, I just noticed that the only successful echo test
that I am able to make with sipphone.com was when calling

sip:17474743246@proxy01.sipphone.com.

Anything else (including the same number but with dashes) was resulting
in silence. I haven't checked the differences in the media flows in the
different cases yet so I don't have an explanation but i saw you were
calling 1-747-474-5000 .. so maybe you'd want to try with the above uri.

Cheers,
Emil

Ron Wilson wrote:

···

I do have alsa-oss installed. And I have verified that /dev/dsp does
refer to my USB headset. (/dev/dsp1 refers to he internal audio. If
sip-com were using that, I should still here the sipphone.com greeting
message)

I have attached the log file.

On Wed, Sep 16, 2009 at 7:39 AM, Emil Ivov <emcho@sip-communicator.org> wrote:

If you don't have a website where to upload it, it's ok to send it here.

Oh, by the way, make sure you have alsa-oss installed.

Cheers,
Emil

Ron Wilson wrote:

I tried that. It does seem to connect sipphone, but when I make a
call, I ge no audio.

I have tested my audio with Audacity and know that it works.

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--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#13

Thaanks.

I tried that. At one time, I had tried just 17474743246, but never
thought to try the full uri.

It worked, but. strangely, the audio output was the internal audio,
even though the audio input was the USB headset. As I previously said,
my current setup has the USB audio configured for /dev/dsp and the
internal audio on /dev/dsp1 - which I just again verified with
Audacity.

I have yet to try a "real" sip call. No time for now.

···

On Thu, Sep 17, 2009 at 5:17 AM, Emil Ivov <emcho@sip-communicator.org> wrote:

I'll try to have a look at this in the following days.

However until then, I just noticed that the only successful echo test
that I am able to make with sipphone.com was when calling

sip:17474743246@proxy01.sipphone.com.

Anything else (including the same number but with dashes) was resulting
in silence. I haven't checked the differences in the media flows in the

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