Re: [asterisk-users] Best USB Handset and Softphone Combination

2007-10-23 Thread Omar A. Sabek 
Hello Steve,

I personally use an 'ultra-portable' headset from Logitech, I would
recommend this for any end-users that have laptops:

http://www.logitech.com/index.cfm/webcam_communications/internet_headsets_phones/devices/223&cl=us,en

There is also this model for desktops:

http://www.logitech.com/index.cfm/webcam_communications/internet_headsets_phones/devices/3622&cl=us,en

I've been using Counterpath Eyebeam with the USB headset for a year,
and I never had a problem. Eyebeam automatically switches it's profile
to use the USB mic and headset when it detects thats it's plugged in.

We haven't deployed Eyebeam to our clients because of the risk
involved with running a phone on an unmanaged desktop. Most of our
customers have little desktop management, if any. But I would
recommend either Eyebeam or Bria, they have a provisioning system for
easier management and the phones can be rebranded.

Good luck ;-)

Omar

On 10/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
> I have a client that want to try the softphone with USB handsets route
> to see if hardphones will even be needed.  I always push for hardphones
> (Polycom) so I am not sure about softphones or USB handsets.
>
> This is going to be for a 300+ seat call center onsite and many offsite,
> I plan on using OpenVPN for the offsite machines.
>
> Any advice on softphones, handsets, or practical experience with this
> sort of deployment?  It would be very nice if there was a central way of
> provisioning the phones.
>
> All machines are fairly new (newer than two years), they have very
> strict policies on downloads and streaming.
>
> Thanks in advance.
>
> Thanks,
> Steve
>
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Re: [asterisk-users] Snom 360 lights not working on subscription

2007-10-22 Thread Omar A. Sabek 
Hey Mike,

We started deploying exclusively Polycom and Linksys. The Polycom's
support presence, they call it 'Buddy List'. I am not sure about the
Linksys phones, I don't think they do although I did see support for
SLA (Shared Line Appearance).

Omar

On 10/23/07, Michael J. Liberatore <[EMAIL PROTECTED]> wrote:
> I also have problems with these phones.  I have deployed many of them
> and have had nothing but problems.  Omar, what phones did you switch to?
> I needed some of the features of the snom phones, like the multiple
> buttons with prescence lights.
>
> Mike
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Omar A.
> Sabek
> Sent: Monday, October 22, 2007 9:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Snom 360 lights not working on
> subscription
>
> I used to deploy these phones, it was these types of issues that forced
> me to drop it. It took way too long to troubleshoot the problems and
> there was a general lack of documentation. This was 2 years ago, things
> might have changed. If I remember correctly, it was this issue you are
> having that was the final straw.
>
> Good luck,
>
> Omar
>
> On 10/22/07, Carlos Maimone <[EMAIL PROTECTED]> wrote:
> > Dear friends,
> >
> > I am working around with a Snom 360 and Asterisk 1.4 + FreePBX
> >
> > In order to get subscriptions working and the Snom 360 lights turns
> > on, I have set everything just like all the pages in the net explain.
> >
> > So, I get subsciption working. I can list subscription on the asterisk
>
> > and if I use the SIP trace function built in at the SNOM nad see
> > NOTIFY messages and 200 OK responses. But I realized that content
> > length = 0 in all messsages and there isn't any XML content in those
> > Notify headers..
> >
> >
> > any idea of what's going on?
> >
> > IN SNOM 360 I am currently using firmware 6.5.12
> >
> > I am pretty sick dealing with this issue.
> >
> >
> > thanks and regards,
> >
> >
> > Charlie
> >
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Re: [asterisk-users] Snom 360 lights not working on subscription

2007-10-22 Thread Omar A. Sabek 
I used to deploy these phones, it was these types of issues that
forced me to drop it. It took way too long to troubleshoot the
problems and there was a general lack of documentation. This was 2
years ago, things might have changed. If I remember correctly, it was
this issue you are having that was the final straw.

Good luck,

Omar

On 10/22/07, Carlos Maimone <[EMAIL PROTECTED]> wrote:
> Dear friends,
>
> I am working around with a Snom 360 and Asterisk 1.4 + FreePBX
>
> In order to get subscriptions working and the Snom 360 lights turns
> on, I have set everything just like all the pages in the net explain.
>
> So, I get subsciption working. I can list subscription on the
> asterisk and if I use the SIP trace function built in at the SNOM nad
> see NOTIFY messages and 200 OK responses. But I realized that content
> length = 0 in all messsages and there isn't any XML content in those
> Notify headers..
>
>
> any idea of what's going on?
>
> IN SNOM 360 I am currently using firmware 6.5.12
>
> I am pretty sick dealing with this issue.
>
>
> thanks and regards,
>
>
> Charlie
>
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Re: [asterisk-users] Refrigerator Alarms

2007-10-17 Thread Omar A. Sabek 
Balu,

Do you want events passed to Asterisk from the refrigerator? Or does a
reminder type phone call need to be placed on an interval? Please be
more specific, since this sounds like a special purpose refrigerator,
does it have any way of passing events to an external device?

Omar A. Sabek

On 10/17/07, Balu Raman <[EMAIL PROTECTED]> wrote:
> Hi,
> I want asterisk to call a person on the phone for monitoring the
> refrigerator storing vaccines.
> I am clueless where to look. Can someone clue me in ?
> Thanks,
> balu raman
>
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Re: [asterisk-users] DID to hunt group?

2007-10-16 Thread Omar A. Sabek 
Hey Rich,

This functionality is included in freepbx. It's called ringall-prim
and can be found as an option for ringing phones in a Ring Group. You
may want to it out for a solution to your issue or install it to
manage your system ;-)

Omar

On 10/16/07, Rich <[EMAIL PROTECTED]> wrote:
> Asterisk 1.4.2
>
> I have spent much of today trying to make a DID (from SIP GW)
> ring to 4 extensions in a hunt (roll-over) group.
> Results from searching docs and forums seem to indicate it is doable
> and so trivial no one includes an actual example.
>
> I can make all 4 exts ring at once with the like of
>
> exten => _1655,1,Ringing()
> exten => _1655,2,Dial(SIP/1655&SIP/1656&SIP/1657&SIP/1658)
> exten => _1655,3,Hangup()
> exten => _1655,103,Congestion()
>
> but I want to ring the 1st ext, if it is busy then ring the 2nd, etc...
>
> I am trying to emulate a 10-line 20 phone key system.
> The end user does not like all the phones ringing at once.
>
> I have tried lots of combinations in extensions.conf
> including the following that I would think should work...
> the 2nd line never rings, just dead air, or I hear a click and dead air...
> where am I going wrong
>
> !! This did not work, not sure why
> exten => _1655,1,Ringing()
> exten => _1655,2,Dial(SIP/1655)
> exten => _1655,103,Dial(SIP/1656)
> exten => _1655,104,Hangup()
> exten => _1655,204,Dial(SIP/1657)
> exten => _1655,205,Hangup()
> exten => _1655,305,Dial(SIP/1658)
> exten => _1655,306,Hangup()
> exten => _1655,406,Congestion()
>
>
> !! this didn't work , I hear no ringing, 2nd call gave some wierd error about 
> codec and slin
>
> exten => _1655,1,Answer()
> exten => _1655,2,Dial(SIP/1655,30)
> exten => _1655,3,Hangup()
> exten => _1655,103,Dial(SIP/1656,20)
> exten => _1655,104,Hangup()
> exten => _1655,204,Dial(SIP/1657,10)
> exten => _1655,205,Hangup()
> exten => _1655,305,Dial(SIP/1658)
> exten => _1655,306,Hangup()
> exten => _1655,406,Congestion()
>
>
> Any clues will be greatly appreciated!
> Thanks,
> Rich
>
>
>
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Re: [Asterisk-Users] Cisco 7970 SIP - few questions

2006-04-26 Thread Omar A. Sabek 
Restarting the 7970 is like unlocking it twice, *-*-# to unlock, *-*-#
to reboot. I don't believe hint functionality works on the SIP
firmware for the 7970.

Omar A. Sabek

On 4/18/06, Tomislav Parčina <[EMAIL PROTECTED]> wrote:
> - How to restart the phone? (On 7960 it is *+6+Settings)
> - How to setup working dtmf?
> - How to setup hinting?
> For line is
> 
> 9
> ...
>
> For speeddial is
> 
> 2
> 341
> 341
> 
>
> How to define hinting?
>
> - How to login true ssh? I have setup username and password, and when I try 
> to log in it sends me challenge!?!
>
> login as: root
> [EMAIL PROTECTED]'s password:
> login: root
>
> challenge: YDXWGXMTpassword:
>
> Invalid Username/Password Entry.
> login:
>
> That is all, for now :))
>
>
> --
> Tomislav Parcina
> tparcina#lama.hr
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Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?

2006-03-18 Thread Omar A. Sabek 
It seems the proxy address is added to all incoming calls to the Cisco phone.

On 3/16/06, Tim Connolly <[EMAIL PROTECTED]> wrote:
> I'm not sure this is the issue. Every call seem to get the proxy
> address added whether it's the main proxy or the backup. What has to
> match to make the phone NOT append the proxy address?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tim
> Connolly
> Sent: Wednesday, March 15, 2006 1:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
>
> That's probably what is happening on my end. Any suggestions on how to
> fix this?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Aaron
> Daniel
> Sent: Tuesday, March 14, 2006 7:22 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
>
> We only had the problem when the call was redirected from one server to
> another.  So if a phone was called from another phone on the server, the
> called worked perfectly, but if it was redirected from another server,
> we got the proxy added to the end.  Doesn't help when you're trying to
> make the existence of multiple servers transparent.
>
> Aaron
>
> Chris Stenton wrote:
> > Maybe I have something strange in my dial plan but I have no problem
> > just hitting dial from missed calls under 8.2.
> >
> > Chris
> >
> > - Original Message - From: "Aaron Daniel" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Monday, March 13, 2006 8:44 PM
> > Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
> >
> >
> >> We rolled back to 7.4 cause of that too.  7.5 has a strange bug where
>
> >> if the server loses connection, the phone's just don't try
> >> re-registering.
> >>
> >> Aaron
> >>
> >> Tim Connolly wrote:
> >>> Just curious, why not 7.5 ? -Original Message-
> >>> From: [EMAIL PROTECTED]
> >>> [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
>
> >>> Jafferali
> >>> Sent: Monday, March 13, 2006 2:28 PM
> >>> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> >>> Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
> >>>
>  I'm using P0S3-08-2-00.. I noticed the callerID started showing
> >>> up
>  with the number, then @... So the callerID on the phone
>
>  looks like: [EMAIL PROTECTED] which of course is logged in the
>
>  missed calls exactly like that, and completely foobars the dialing
>  string if you try to dial a missed call by simply hitting the dial
>  button. Can anyone else verify this problem?
> >>>
> >>> Yeah, that bothered me so I rolled back to SIP 7.4.
> >>>
> >>> Nabeel
> >>>
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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

2006-03-14 Thread Omar A. Sabek 
Chris, you may have a different and simpler setup. Internal calls work
fine here, since the proxy server on the CallerID is the same proxy
server used for all internal users. I was referring to calls that
originate outside of the enterprise. I should have been more clear.

Omar



On 3/14/06, Chris Stenton <[EMAIL PROTECTED]> wrote:
> Yes it does display caller id as @ but
> that does not interfere with me hitting dial from missed calls. Seems the
> Cisco phone sends the  sip INVITE as  @
> rather than @ but asterisk
> ignores the info after the @?
>
> Chris
>
> - Original Message -
> From: "Omar A. Sabek" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Monday, March 13, 2006 11:45 PM
> Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
>
>
> > The 8.2 firmware displays Caller ID as @...
> > this becomes problematic for users that want to dial from their
> > 'Missed Calls' log.
> >
> > Omar
> >
> > On 3/13/06, Nathan Bowyer <[EMAIL PROTECTED]> wrote:
> >> On 3/13/06, Chris Stenton <[EMAIL PROTECTED]> wrote:
> >> > I have had no issues with 8.2 so far!
> >> >
> >> > Chris
> >> >
> >>
> >> Except the Caller ID issue reported in another thread?
> >>
> >> > >>
> >> > >> This issue has been fixed in SIP firmware 7.5
> >> > >>
> >> > >> Omar A. Sabek
> >> > >
> >> > > Yes, and I read that SIP 7.5 firmware have some other issues. They
> >> > > recommend using 7.4 firmware. I'm not sure how good in new 8.2
> >> > > firmware.
> >> > >
> >> > >
> >> > > Tomislav
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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

2006-03-13 Thread Omar A. Sabek 
To be totally honest, I have 7.5 running on many phones and I have yet
to receive a report on a firmware related issue.

Omar

On 3/13/06, Tomislav Parcina <[EMAIL PROTECTED]> wrote:
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Omar A. Sabek
> > Sent: 9. ozujak 2006 18:12
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time
> >
> > This issue has been fixed in SIP firmware 7.5
> >
> > Omar A. Sabek
>
> Yes, and I read that SIP 7.5 firmware have some other issues. They recommend 
> using 7.4 firmware. I'm not sure how good in new 8.2 firmware.
>
>
> Tomislav
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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

2006-03-13 Thread Omar A. Sabek 
The 8.2 firmware displays Caller ID as @...
this becomes problematic for users that want to dial from their
'Missed Calls' log.

Omar

On 3/13/06, Nathan Bowyer <[EMAIL PROTECTED]> wrote:
> On 3/13/06, Chris Stenton <[EMAIL PROTECTED]> wrote:
> > I have had no issues with 8.2 so far!
> >
> > Chris
> >
>
> Except the Caller ID issue reported in another thread?
>
> > >>
> > >> This issue has been fixed in SIP firmware 7.5
> > >>
> > >> Omar A. Sabek
> > >
> > > Yes, and I read that SIP 7.5 firmware have some other issues. They
> > > recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware.
> > >
> > >
> > > Tomislav
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Re: [Asterisk-Users] 7940/60 SIP 8.2

2006-03-09 Thread Omar A. Sabek 
I just started looking at the differences between 8.2 and 7.5.

In addition to the new "Security Configuration", there is also
compatibility added for provisioning the phone to CCM. The firmware
appears to be working fine.

Also, the copywright date range has been changed to 2000-2006.

Omar A. Sabek

On 3/9/06, C F <[EMAIL PROTECTED]> wrote:
> Does that mean that since CCM supports SIP, Cisco will just make sure
> that their SIP images work with CCM?
>
> On 3/9/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
> > The image is located in the non-sip section, go figure.  They're harping
> > that this is for their new sip ccm...
> >
> > Aaron
> >
> > C F wrote:
> > > Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5.
> > > Are you guys talking about SIP?
> > >
> > > On 3/9/06, Mailing List <[EMAIL PROTECTED]> wrote:
> > >> - Original Message -
> > >> From: "Nabeel Jafferali" <[EMAIL PROTECTED]>
> > >> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> > >> 
> > >> Sent: Thursday, March 09, 2006 10:42 AM
> > >> Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2
> > >>
> > >>
> > >>>>> So has anybody tried installing the new SIP version?
> > >>>>> It seems nobody has had luck with the 7970 and it's new SIP image
> > >>>>> and the description for the 7940/60 specifically says "for CCM
> > >>>>> v5.0".
> > >>>> Just downloaded it after your email and got it working on the first
> > >>>> try. Give me a few minutes to write up the procedure.
> > >>> OK, I had a 7960 running SIP 7.4 and managed to upgrade to SIP 8.2. I
> > >>> haven't tried any of the new features, but can make and receive calls 
> > >>> fine.
> > >>>
> > >> Sweet, guess I'll give it a go.
> > >>
> > >> _
> > >> Mobilcom
> > >> http://www.mobilcom.net
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Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time

2006-03-09 Thread Omar A. Sabek 
This issue has been fixed in SIP firmware 7.5

Omar A. Sabek

On 3/9/06, Nathan Bowyer <[EMAIL PROTECTED]> wrote:
>
> On 3/9/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> >
> On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote:
> > > Is there a way to display the time of the 7960 running firmware 7.4? Im
> > > unable to find any information.
> >
> > Add the following to SIPDefault.cnf or SIP.cnf:
> >
> > sntp_server: "time.nrc.ca"
> > sntp_mode: unicast
> > time_zone: EST
> >
> On my 7960 with 7.4 firmware, the time automagically disappears for some
> unknown reason.   The phone still functions, but the time goes away
> until I reboot it.  Not a big deal to me, so I have not investigated it
> further.
>
> -Greg
>
>
> I use anycast.  Seems like I read something about directbroadcast not
> working in recent SIP versions.
>
>
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Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones

2006-03-04 Thread Omar A. Sabek 
The only two programmable buttons are the 'Messages' and 'Services'
and 'Directory buttons'. They are all configured in sip_default:

messages_uri: "number to dial"
services_url: "xml file to load"
directory_url: "xml file to load"

Cheers,

Omar

On 3/4/06, Kevin Steil <[EMAIL PROTECTED]> wrote:
>
>
>
> Does anyone have a good resource to learn how to program the soft and hard
> buttons on a Cisco 7940 or 7960 phone?  Using SIP Firmware…thanks.
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Re: [Asterisk-Users] Linksys WIP300 WiFi Phone

2006-02-27 Thread Omar A. Sabek 
I purchased my WIP300 from voipsupply.com, as far as I know they are
the only ones carrying it.

Cheers,

Omar

On 2/27/06, Me <[EMAIL PROTECTED]> wrote:
> Where are you folks getting the best deal on this phone right now?
>
> After my experience with the last Wifi phone, I am a little gunshy at
> the moment. I am not sure if I should wait to see how this product
> plays out or not.
>
>
>
> On 2/27/06, Philip Edelbrock <[EMAIL PROTECTED]> wrote:
> >
> > Omar A. Sabek wrote:
> > > Like BJ, I'm sorry you had bad luck Phil. I have been playing with
> > > this phone all weekend, and I have had minor problems. The voice
> > > quality is as good as my cisco and polycom sip phones. I asked a
> > > friend to guess what kind of phone I was talking on and he said it
> > > sounded like a regular home or office phone. I have been very happy
> > > with the voice quality.
> >
> > My first day was a huge disappointment.  Three crashes, calls wouldn't
> > work over my work's wifi (eventhough it registered ok), short battery
> > time, lost settings after a crash, etc.
> >
> > However, after I went in and cleared my settings back to default, the
> > troubles went away!  I'm been using it for over three days without a glitch.
> >
> > So, I would recommend to anybody else who is getting one of these
> > phones, to immediately set all settings back to 'default' (under the
> > Tools menu) before spending too much time configuring it.
> >
> > > I reported on the voip-info page dismal talk times but it must have
> > > been an anomoly. Today I spoke for over an hour on the phone and still
> > > had plenty of juice left.
> >
> > My battery life seems to have improved as well.  I don't know if that's
> > was a glitch fixed by setting things back to the defaults, or if cycling
> > the battery is helping.  I also have less of a tendency to play with the
> > menus, and the backlight could be a power drainer (it is quite bright).
> >
> > >
> > > All-in-all this phone is a winner. It works with Asterisk flawlessly.
> >
> > As long as my troubles don't come back, I would agree.  I think my phone
> > was shipped to me in a funny state causing it not to work right. It's a
> > winner now.
> >
> > There are some little things I would wish for, but I'm quite happy with it.
> >
> >
> > Phil
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Re: [Asterisk-Users] Linksys WIP300 WiFi Phone

2006-02-26 Thread Omar A. Sabek 
Like BJ, I'm sorry you had bad luck Phil. I have been playing with
this phone all weekend, and I have had minor problems. The voice
quality is as good as my cisco and polycom sip phones. I asked a
friend to guess what kind of phone I was talking on and he said it
sounded like a regular home or office phone. I have been very happy
with the voice quality.

The only problem I had was my phone would not connect to my wireless
network this morning. I was a bit puzzled because it worked last night
and all my settings were correct. In a move of frustration I deleted
all the 'preset' profiles that came with the phone and recreated the
current wi-fi profile. It picked up my network and connected to * no
problem.

The GUI is much like a cell phone's. It has a center joystick that
takes a bit to get used to handling, it's pretty sensitive. One great
thing about the configuration of the phone is the seperation between
'network profile' and 'sip account'. In other words, after configuring
the network settings, you select the SIP account you want to use when
on that network. This is especially useful for users that have unique
wi-fi settings per location but a common SIP proxy like *.

I reported on the voip-info page dismal talk times but it must have
been an anomoly. Today I spoke for over an hour on the phone and still
had plenty of juice left.

All-in-all this phone is a winner. It works with Asterisk flawlessly.
Don't look for much support from Linksys though as I called them when
I received the phone and they told me they dont carry any item with
that model number. That should be a temporary thing though, I have a
feeling these wi-fi linksys phones are going to be pretty popular.

Thanks,

Omar A. Sabek

On 2/26/06, Me <[EMAIL PROTECTED]> wrote:
> How is the voice quality?
>
> >  I've just plugged mine back into the charger after having used it
> > nearly all day. I didn't have any of the problems you've described.
> > Sorry you're having such bad luck with it. I'm not certain what the
> > phones are rated to do, but I probably got better than 3 hours talk
> > time on it today which is definitely the best I've gotten with any of
> > the WiFi phones up to this point.
> >
> >  BJ
> >
> > --
> > Bird's The Word Technologies, Inc.
> > http://www.btwtech.com/
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Re: [Asterisk-Users] simulating a few thousand SIP clients?

2006-01-29 Thread Omar A. Sabek 
Hello Roy,

Have you heard of Sipp? http://sipp.sourceforge.net/.  I am pretty
sure it can do what you desire.

Also a commercial tool from Empirix, Hammer NXT.
(http://www.empirix.com/default.asp?action=article&ID=64)

Cheers,

Omar



On 1/29/06, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote:
> sure, but I need to simulate the SIP REGISTER and OPTION traffic sent
> by ATAs as well.
>
>
> Best regards
>
> Roy Sigurd Karlsbakk
> [EMAIL PROTECTED]
> ---
> In space, loud sounds, like explosions, are even louder because there
> is no air to get in the way.
>
>
> On Jan 29, 2006, at 7:26 PM, Wai Wu wrote:
>
> > Set up another * and use the manager api to make lots of calls to
> > the other one. You can even make hundresd calls at a time.
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf Of Roy
> > Sigurd
> > Karlsbakk
> > Sent: Sunday, January 29, 2006 1:19 PM
> > To: Asterisk Non-Commercial Discu
> > Subject: [Asterisk-Users] simulating a few thousand SIP clients?
> >
> >
> > hi
> >
> > i'm setting up a rig to handle quite a few SIP clients, so i need a
> > way to simulate, say, 20k SIP ATAs. Does anyone know how? This should
> > of course be as close as possible to 'reality', meaning n% calls per
> > client and the usual REGISTER/OPTION traffic.
> >
> > thanks
> >
> > Best regards
> >
> > Roy Sigurd Karlsbakk
> > [EMAIL PROTECTED]
> > ---
> > In space, loud sounds, like explosions, are even louder because there
> > is no air to get in the way.
> >
> >
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Re: [Asterisk-Users] Odd Polycom Dialing Problem

2006-01-17 Thread Omar A. Sabek 
Modify or delete the digitmap. The digitmap allows the phone to
intelligently detect numbers dialed (and presumably save the user from
hitting send after a specified number of seconds). The digitmap is
available on the web interface under SIP and Local Settings. The
default is:

[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT

We pass all digits to Asterisk, which handles it intelligently.

Omar A. Sabek

On 1/18/06, calvis <[EMAIL PROTECTED]> wrote:
>
>
> I am starting to play around with my first Polycom phone.  I have it
> registered to my asterisk box, but the problem is dialing.  When I try to
> dial with the handset I get my provider telling me that they can not
> complete my call afterwards Allison pops in and tells me all circuits are
> busy and can not complete my call.
>
> However, if I punch in the numbers and then hit the dial button the call
> completes with no problem.
>
> Does anyone have a clue what is going on?  This is a SoundPoint IP 601 phone
> and I am using Asterisk at Home 2.2.
>
>
> Charles Alvis
> Internet Technology Group, Inc.
> Redmond, WA
>
>
>
>
>
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Re: [Asterisk-Users] Voipsupply - my experience

2005-12-15 Thread Omar A. Sabek 
I also have nothing but wonderful things to say about voipsupply.com,
they are first on my list for equipment needs. They have great methods
of communication and they free up my time to better serve my clients.

Omar A. Sabek

On 12/15/05, Dennis Gilmore <[EMAIL PROTECTED]> wrote:
> On Thursday 15 December 2005 11:55, Terry H. Gilsenan wrote:
> > Hi,
> >
> > I would just like to let everyone know about the support I have rec'd
> > from voipsupply.
> >
> > I travelled from .au to the US to do some urgent server upgrades for
> > $EMPLOYER. During this trip I has impressed upon me (with absolutely
> > zero notice) the requirement for a IP Phone link to the .au office.
> >
> > With only a couple of day left in the US I was stumped.
> >
> > I called Voipsupply and left a message (it was 9:30PM CST - Houston) at
> > 10:30PM Mark emailed me and sent me his Cell phone number.
> >
> > The upshot is:
> >
> > I placed an order for some Digium kit at Midnight, on the 14th and I
> > have just rec'd the goods in Houston now.
> >
> > Mark: Fantastic service! I appreciate the help.
> >
> > Regards,
> Having moved from AU  to US last year I know how important a good reliable and
> cheap link back home is.   for this i use digium equipment and asterisk  with
> an AU iax provider.  i have a local number in Brisbane.  I have also
> purchased some VoIP equipment from Voipsupply for my employer and have
> received fantastic service and support from them for that also.
>
> Dennis
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Re: [Asterisk-Users] Slightly OT - Anyone know of an external ringer compatible with Cisco phones

2005-11-22 Thread Omar A. Sabek 
Hey Cory,

I havent come across a voip ring amplifier or visual indicator. Here
are some amplifiers and visual inidicators for an environment using
ATAs:

http://www.soundbytes.com/Merchant2/merchant.mvc?Screen=CTGY&Store_Code=SB&Category_Code=PhoneRingAmplifier


Omar A. SabekOn 11/22/05, Cory Andrews <[EMAIL PROTECTED]> wrote:
Have an application where Cisco phones are being used in a noisyenvironmentlooking for some type of external ringer or amplifier sousers can hear the phones ringing over the background noise.  Anyonefamiliar with such a device?
Thanks,--Cory J AndrewsPartner / Purchasing+++VOIPSupply.com - Everything you need for VOIP454 Sonwil DriveBuffalo, NY 14225+++tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22f - 716.630.1548e - [EMAIL PROTECTED]AIM - b2Cory___--Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance

2005-11-16 Thread Omar A. Sabek 
If I understand this thread correctly, they are discussing monitoring
the status of other agents. This cannot be done with the SIP firmware
on the CP-79x0 phones. 

We are currently working on testing the SCCP firmware with the
revisions made to the chan-sccp module, but I am stuck on getting the
phone to accept the CTLSEP.tlv file. Does anyone have a
working version of the xml contents of this file?

Thanks,

Omar A. SabekOn 11/16/05, Chris Stenton <[EMAIL PROTECTED]> wrote:
- Original Message -From: "Sergio Chersovani" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"<
asterisk-users@lists.digium.com>Sent: Tuesday, November 15, 2005 2:25 PMSubject: Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance> Matt Hoskins ha scritto:>>> I'm currently using asterisk 
1.0.7 with cisco 7960 SIP 7.5 phones.  I'd>> like to use this phone for a receptionist so that she can take calls for>> 4 other people.  Is this possible?>> The SIP firmware does not support it.
> You have to use SCCP to do that>>> Is there any way to do this with SIP and the 7960?  I've seen the 7914>> but then I'd have to use SCCP and I'm not sure if it is stable enough for
>> production use.>> Well give it a chance :-)>> http://chan-sccp.berlios.de>I've got all six  SIP lines registered as extention 201 and I've had all six
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Re: [Asterisk-Users] Snom 360 Opinions

2005-11-12 Thread Omar A. Sabek 
Curren,

Can you tell us a little more about the environment you are deploying these phones in, how many phones and what kind of setup?

Omar A. SabekOn 11/12/05, Curren C. Calhoun <[EMAIL PROTECTED]> wrote:
Thanks that's the type of info I'm looking for...I've heard some early grumblings but wanted to see if anything else has comeup...> From: Remco Barende <
[EMAIL PROTECTED]>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion> > Date: Sat, 12 Nov 2005 09:50:01 +0100 (CET)
> To: Asterisk Users Mailing List - Non-Commercial Discussion> > Subject: Re: [Asterisk-Users] Snom 360 Opinions
>> I'm not too pleased with the phones, I have about 40 of them, some of the> displays tend to die and the dial pad feels to 'mushy' IMHO, just like the> keys on a good old ZX80 computer>
> Also I'm having some issues with sound quality on some phones, but I still> need to switch some phones to see if that is really an issue of the phone.>> Also if you want to use * call files, with the 360 you will run into a big
> where the call is being redialled as if it failed while in fact the call> is ongoing. Annoying and haven't found out if that is an * bug or Snom> bug. The Snom 190's do not have this problem.>
> Just my $0.02 (which is really not a lot these days!) :)>> On Sat, 12 Nov 2005, Curren C. Calhoun wrote:>>> I¹m looking to add in some Snom 360 phones, could anyone give thoughts or
>> opinions about the speakerphone, general quality... Also the phone would>> need to be powered over Ethernet... I like some of the listed features and the expandability of the phone but am
>> open to any other suggestions as well... Thanks>> Curren>>___--Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] user name

2005-10-20 Thread Omar A. Sabek 
I think you might be referring to the asterisk-users listserv. Please visit the following page and obtain a password reminder:

http://lists.digium.com/mailman/listinfo/asterisk-users

Omar A. SabekOn 10/21/05, Jerry Richmond <[EMAIL PROTECTED]> wrote:
I am not spaning I just want to stop all this e-mail with out giving up my corprate e-mail address. HelpAndres <
[EMAIL PROTECTED]> wrote:
I don't get it either. He is either clueless or is someone spamming this list with nonsense we don't need to know about.
Jonathan k. Creasy wrote:> I don't get it.>> >> *From:* 
[EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED]
] *On Behalf Of *Jerry > Richmond> *Sent:* Thursday, October 20, 2005 9:46 AM> *To:* 
asterisk-users@lists.digium.com> *Subject:* [Asterisk-Users] user name>> I am geting e-mail but asterisk doesn't know my user name or password. > My user name has always Been Jerry Richmond, my e-mail address 
> [EMAIL PROTECTED] I > need a password of some kind.
>>
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Re: [Asterisk-Users] Problem with Cisco phone

2005-10-20 Thread Omar A. Sabek 
Unfortunately, I think you're right. It's a good thing you only have to reboot the phones for config and firmware changes.

Omar A. SabekOn 10/20/05, Tom Tune <[EMAIL PROTECTED]> wrote:
It happens on everything I have tried. I do not have any vlans or qos
configured. Even on a generic 5 port with only the phone and asterisk
server connected.
I am beginning to think that it is a Microsoft style mutual upgrade scenario.. can't use the phones without the switch.On 10/20/05, 
Omar A. Sabek <
[EMAIL PROTECTED]> wrote:
Tom,

Do you have VLANs or QoS configured on your switch? I had the same
problem with my CP-7960 connected to a Catalyst 3524XL switch. I solved
it by tweaking my QoS settings on the switch, there was an option to
enable dot1p priority tagging on the voice VLAN. I am not sure if the
Dell switch you are using has a similar feature, if it does check there.

Also, how have you troubleshot this problem so far, does it happen on
every switch you've tried? Is it happening on every CP-7960G you've
tried? You're right about it being a simple problem, that aux switch
port has always worked fine IMLE.

Cheers,

Omar A. SabekOn 10/20/05, Tom Tune <

[EMAIL PROTECTED]> wrote:
My Cisco 7960g SIP phones share an annoying "feature": Anything I plug
into their 2nd Ethernet or PC port loses connection every thirty
seconds or so. I did a ping -t and can see regular drops. I do not have
access to Cisco's tech support archives. I am pretty sure that this is
a simple configuration problem. Can somebody point me in the right
direction? Thanks in advance.

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Re: [Asterisk-Users] Cisco 79XX and VLANS

2005-10-20 Thread Omar A. Sabek 
Hey Matthew,

Simply turn on dot1p priority tagging on the switchports your phones
are connected to. This won't interefere with the auxiliary phone
switchport and it will enhance (to a certain degree) your voice QoS. My
CP-7960s boot up in less than 5 seconds flat.

Omar A. SabekOn 8/11/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
Hey gang,  We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We arealso using all Cisco Switches and Routers. Everything works great exceptthat when you reboot a phone it takes like 3-5 minutes for it to come up.
  The phones spend tons of time 'Configuring VLAN..' We don't run anyVLANs. Is there some way to skip this?  In the 'Network Settings' I have both 'Operational VLAN Id' and 'AdminVLAN Id' set to blank values.
  Any ideas?Thanks,Matthew___Asterisk-Users mailing listAsterisk-Users@lists.digium.com
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Re: [Asterisk-Users] merchant account

2005-10-20 Thread Omar A. Sabek 
Hey Bret,

Most companies will get around this quandry by assigning their
customers account numbers. This will then be x-referenced to a DB that
contains that information (Name, Billing Addy, maybe last 4 SS). If
this is not an option, there is always the "Please say and spell your
first and last name after the tone." But, not too clean from an
automation point of view. 

Would switching to a CC processor that doesn't require a name be an
option? I remember making international calls years ago by entering
only my CC # and expiration. I am not sure if this is an option any
longer.

Omar A. SabekOn 10/20/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote:
On Thu, 2005-10-20 at 22:29 -0400, Omar A. Sabek wrote:> The CC merchant machines I've encountered require entry of the account> number, exp date, total charge, etc. before dialing and transmitting> the data. Even though we are able to pass DTMF successfully through
> the gateway, we still make the recommendation that any application> that requires a negotiation phase (ie, fax machines, CC merchant> machines, dial-up modem) remain on a traditional POTS line. And just
> like you mention, alternative methods are available including web> access.>> Hopefully, the technological disconnect between voip and dial-up data> transmission will come to pass sooner than later.
I should have added one more requirement.  Internet authentication likeso many do.  Although all that I have worked with require the name to beentered as text, something that isnt trivial to do (and asking people to
tap out their name via dtmf would be too bothersome).I would like a merchant solution that would work 100% off someonecalling into the system and entering their data, authentication willthen occur via the internet.
--Trixter http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605   Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378
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Re: [Asterisk-Users] Problem with Cisco phone

2005-10-20 Thread Omar A. Sabek 
Tom,

Do you have VLANs or QoS configured on your switch? I had the same
problem with my CP-7960 connected to a Catalyst 3524XL switch. I solved
it by tweaking my QoS settings on the switch, there was an option to
enable dot1p priority tagging on the voice VLAN. I am not sure if the
Dell switch you are using has a similar feature, if it does check there.

Also, how have you troubleshot this problem so far, does it happen on
every switch you've tried? Is it happening on every CP-7960G you've
tried? You're right about it being a simple problem, that aux switch
port has always worked fine IMLE.

Cheers,

Omar A. SabekOn 10/20/05, Tom Tune <[EMAIL PROTECTED]> wrote:
My Cisco 7960g SIP phones share an annoying "feature": Anything I plug
into their 2nd Ethernet or PC port loses connection every thirty
seconds or so. I did a ping -t and can see regular drops. I do not have
access to Cisco's tech support archives. I am pretty sure that this is
a simple configuration problem. Can somebody point me in the right
direction? Thanks in advance.

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Re: [Asterisk-Users] merchant account

2005-10-20 Thread Omar A. Sabek 
The CC merchant machines I've encountered require entry of the account
number, exp date, total charge, etc. before dialing and transmitting
the data. Even though we are able to pass DTMF successfully through the
gateway, we still make the recommendation that any application that
requires a negotiation phase (ie, fax machines, CC merchant machines,
dial-up modem) remain on a traditional POTS line. And just like you
mention, alternative methods are available including web access.

Hopefully, the technological disconnect between voip and dial-up data transmission will come to pass sooner than later.

Omar A. SabekOn 10/20/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote:
I am interested in hearing some user experiences of anyone using amerchant account.  The constraints are that everything entered must beDTMF-able.  Card number, CCV, exp, numeric portion of the streetaddress, zipcode are all easy.   name however is not so easy.
How have others solved this problem?  Or have they only set up systemswhere web access is required?--Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDWE5s+1olxlzQw5cRAuJvAJ43RS5gsbSD5YYyzVdij/HlbpSp+wCdHRIK
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Re: [Asterisk-Users] Please recommend a phone

2005-10-20 Thread Omar A. Sabek 
The Cisco CP-7940/60 flashes it's MWI during incoming calls.

If you are using an ATA, there are several devices that can display
flashing/blinking lights during incoming calls by simply putting it
between the ATA and phone.

OmarOn 10/19/05, Christian Stredicke <[EMAIL PROTECTED]> wrote:
Take a look at snom.com...CS> -Original Message-> From: [EMAIL PROTECTED]> [mailto:
[EMAIL PROTECTED]] On Behalf Of> Jesse Keating> Sent: Wednesday, October 19, 2005 5:31 PM> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Please recommend a phone>> On Wed, 2005-10-19 at 16:39 -0400, Jesus Mogollon wrote:> >> >I'm in need of a phone that would blink a led to let the callee
> > know that there is an incoming call. The GXP-2000 does this> but I want> > an alternative to Grandstream. Any help is appreciated.>> Polycom IP301s and 501s have a red LED that blinks when calls
> are coming in.>> --> Jesse Keating> GameHouse -- Systems Engineer>> ___> --Bandwidth and Colocation sponsored by 
Easynews.com -->> Asterisk-Users mailing list> Asterisk-Users@lists.digium.com> 
http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] SIP Dialler

2005-10-06 Thread Omar A. Sabek 
Hello Benni,

Check out gnudialer (www.gnudialer.org), I was looking into this for a
client but had to abandon the project. I remember reading that the
dialer allows recording but I am not sure what the method is. It may be
local, check it out.

Omar SabekOn 10/6/05, Benni A. Aswin <[EMAIL PROTECTED]> wrote:
Hi,Any of you have any experience with SIP softphone dialler that capableof local recording? (recording to files in harddrive)So far I only know eyeBeam and Express talk. eyebeam fine but thereare known error with recording. Express talk recording looks ok, but
sometime it doesn't have incoming voice with *.CheersBenni-___--Bandwidth and Colocation sponsored by Easynews.com --
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[Asterisk-Users] Snom 360 Phones - Administrator/User Feedback

2005-10-06 Thread Omar A. Sabek 
I'm looking for some feedback on the Snom 360 phone. After deploying
six of them to an office, I'm not as enthusiastic about them as I was
when I was testing one before the deployment. The firmware seems to be
consistently buggy, some of its problems are intermittent which makes
it frustrating to troubleshoot and the support from Snom is lackluster
to say the least. I find myself favoring Cisco and Polycom phones, not
only from a user POV but also the automation of deployment... Does
anyone share similar sentiments?

Omar Sabek
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