If asterisk is going to be modified to support LiveVoip expectations,
then yet another Dial option would need to be implemented to
force ringback to occur as an audio stream for iax only. Guess
one could open a bug report for both LiveVoip and Asterisk, but
not likely to be addressed
Guys. I have a few IAX2 connectivity questions that maybe somebody can
clarify to me:
I have my * server and another one with a friend. We are both inside nat and
doing port forwarding:
* - nat - internet - nat - *
Now, what I dont understand is this, why FWD needs to be
Sorry everyone, I know this has been hashed over a bunch of times but I
can't find anything that pertains to specific cracking and popping on the
FXO modules of a TDM04. This happens on inbound or outbound calls. This is
the first install I have done with a TDM card for FXO modules so
Looks like its working fine now since you answered the call that
I placed to your fwd number. :)
From: Tim Pushor [EMAIL PROTECTED]
Subject: [Asterisk-Users] FWD IAX Problem
Date: Mon, 14 Mar 2005 13:58:28 -0700
To: Asterisk Users Mailing List - Non-Commercial
right now. After perusing the fwd forums I believe
this may actually be a fwd thing.
Thanks,
Tim
(Now I can receive calls, but not place them ;-)
Rich Adamson wrote:
Looks like its working fine now since you answered the call that
I placed to your fwd number
I'm guessing it's the sudafed that caused me to wildly try this, but I'm
glad I did, because though it creates a new concern, it solved my
problem. Just for kicks I tried setting the canreinvite parameter to no
for the broadvoice peer, and that fixed everything.
My server is on a live ip,
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based interface
works fine, in addition to dialing 8500,
which is mapped to:
exten = 8500,1,VoicemailMain
exten =
I have a quick question I hoping someone can help me with. I have
[EMAIL PROTECTED] running and working just fine. I've integrated it with
BroadVoice and so far I'm blown away by everything I can do.
I don't particularly like sitting my entire machine in the DMZ on my
network sitting
im my case im looking into 100 seats initially and going up to 1000 at
the end (over a 18 months period).
Looks like we will have to develop *a lot* if we want to use * for it.
Maybe a commercial solution will be better at this time.
On Cebit SGI announced a server solution based on
How exactly does Asterisk provide E911 service??
It doesn't do anything with 911. You tell * what to do when someone
dials 911 via your dialplan.
To avoid legal issues down the road, I'd suggest handling it via a
local pstn line (one way or another), and install a Red Phone with
a normal pstn
To avoid legal issues down the road, I'd suggest handling it via a
local pstn line (one way or another), and install a Red Phone with
a normal pstn line for emergency use. (The pstn line for the Red
Phone 'could' be used for incoming faxes as well, and when combined
with something like
Agreed 100%. Think about how one might config a spa3k to accomplish
everything noted, plus some. :)
Well, incoming call handling on SPA-3000 kind of sucks at the moment...
but I don't see how it could be configured to ring a bunch of phones
anyway. At best it can deliver the call to a
I've posted this question twice without a single reply. Does that mean no
one knows the answer, or no one cares to answer?
I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the
trunk fine and it registers and works fine. I'm able to make outgoing calls
from any
Is that with channels recording ? ;)
We are running 40-50 simultanious calls at the call center here, and
recording everycall in and out, with no problems
On a Pentium 3ghz with 1gig ram.
Can you share with us what type of system this is (or motherboard
model if not a commercial system)?
Well, incoming call handling on SPA-3000 kind of sucks at the
moment... but I don't see how it could be configured to ring a bunch
of phones anyway. At best it can deliver the call to a single
gateway/proxy, and even it really wants to answer the line first and
present a second dial
I've posted this question twice without a single reply. Does that mean no
one knows the answer, or no one cares to answer?
I've been having an issue with an IAX2 trunk setup in Asterisk. Setup the
trunk fine and it registers and works fine. I'm able to make outgoing calls
from any
Inline...
I am trying to get the drivers working with this device with 4 fxo
modules on it. I do a modprobe zaptel and no errors appear. But when I
do modprobe wctdm the following errors appear:
Notice: Configuration file is /etc/zaptel.conf
line 4: Cannot get number of tones chanel 1
I'm having with an echo or delay
I connect to the PSTN with a x100p and then connect a std. phone
to a FXS module on a TDM10B.
The std phone is only 2-wire so I know this is not helping.
(yes I have read the 2-wire 4-wire issue)
I have tried many echocancel values. The best thing to
I bought a couple Polycom Soundpoint 300's, and have them working nicely
with SIP... but I'd like to be able to do automatic config via FTP, but it
requires some XML config files. The docs discuss them in detail, but I
can't seem to d/l them from Polycom. [No, it doesn't appear to be on
Hello. I would like to know if somebody did a wireles voip with Asterisk PBX.
I think to deploy a wireless for about 500 potential customers, it's a 3 km
radius maximum coverage with houses without phone lines, I work for public
places telephony small enterprises ( a common bussines in
A lot of the BV config confusion is the result of users with registered
IP's vs nat'ed IPs. The patch _was_ only required for those that used
nat'ed systems (proven shortly after that patch was released, and backed
by those that wrote the patch).
So, for those that are still mucking around with
If you are outside the US, there isn't much you can do since the x100p
card was specifically designed to operate with US 600 ohm impedance
pstn lines.
If you have a x100p clone, it is likely the problem. Replace it with
something capable of matching the pstn impedance for whatever
I have tdm400p with 4 fxo modules on it. When I call into the asterisk
box from my mobile, I can see the asterisk console picks the call up
and routes it to my computer with x-lite. There was no sound coming
from either - just silence. I then decided to route it directly to
voice mail to
If you are outside the US, there isn't much you can do since the x100p
card was specifically designed to operate with US 600 ohm impedance
pstn lines.
If you have a x100p clone, it is likely the problem. Replace it with
something capable of matching the pstn impedance for
Has anyone used asterisk as a simple voip server? (I'm sure its been/ing
done).
If so... how did you provide 911 service? Did you setup different
contexts and put sip phones in those contexts per county?
I think that's what you'd have to do.
Be careful. 911 centers are not
I use Asterisk at home to filter the annoying people before they get a real
voice. So
basically if you don't know the extension of one of the occupants you have no
choice but leave a
message. Works well... perhaps I miss some improtant calls, but if you leave
no message it must
not be
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if they could produce one for
say $1000 retail?
Trouble is
Other than Broadvoice, are there any VoIP providers (Vonage, Packet8,
etc) that can be hooked into Asterisk directly? I read about a scheme
for Packet8 that involved routing it in through an analog connection
on a FXO port...I'd rather have something I can connect in directly.
Save
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if they could produce one for
say $1000 retail?
international dialing and the commercial plan is fixed price of $44.99
per month capped at 500 international minutes a month. Are you aware
if they have international rates based on usage?
MARK.
Rich Adamson wrote:
Other than Broadvoice, are there any VoIP providers (Vonage
Yesterday I was using one of the cheap Radio Shack phone polarity on various
phone outlets in my house and ended up plugging it into my IAXY. While the
regular phone jacks tested OK, the IAXY tested as being reverse polarity.
The tester was plugged directly into the IAXY so there is no chance
I've got a strange issue, that I haven't found addressed on the wiki.
My asterisk box is behind a firewall which routes udp/tcp requests on 5060
and
8000 to asterisk.
When I make a call from a Zap or SIP extension on the inside of the firewall
to any Zap or SIP extension on the inside
Excuse my ignorance here, but I am desperately trying to isolate the IRQ for
my TE110P card (shown below as t1xxp) Ive gone into my bios and disabled all
usb , parallel, serial and some other devices, those that I needed to keep,
I have moved off of IRQ 10 and onto IRQ 5, but everytime I boot
I'm running Asterisk 1.0.6 with zaptel 1.0.6 on Gentoo
Linux with a 2.6.11-gentoo-r2 SMP kernel (but no SMP
hardware) and mpg123 0.59s-r9.
When I leave a voicemail message via my X100P, the
message is way too quiet. I can barely hear it.
I googled this a bit, and I saw similar complaints
Can anyone tell me what the normal number of
interrupts per second is for an X100P card?
I've used FreeBSD 5.3 and a linux 2.6.11 kernel
on the exact same hardware (only the disk changed)
and `systat -vmstat 1` on FreeBSD and
`procinfo -dS -n1` under Linux. For both, I'm
seeing roughly
Attached to the bottom of this e-mail is an edited version of an e-mail I
originally wrote to Digium tech support regarding Ouch and Power alarm
errors I have been receiving on my TDM400. It contains a great deal of
detail regarding my setup. In the end, I have found that one of the 5
I've improved the stability of my card by adding a capacitor on the
reset line. Hasn't taken a hit in over two weeks.
Is this the E/F or revised H card? Where and what cap did you install?
My card reports as E/F; only have one, so not sure what the differences
are between the various
I'm a little confused on whether the GR303 support in * will accept
calls from a Siemens central office that has GR303.
Anyone know for sure?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I have a x100p card and it doesn't detect a hangup from the calling
party when going in voicemail(). My PSTN provider is sending open loop
disconnect (voltage decrease for a given moment of time). Actually
Progress Detection is HIGHLY EXPERIMENTAL so it should not be required
to fix
I am using TDM FXO (4) with one of my server , in middle east and there
internet not so good, every time its has some packet loss happend. but speed
is good. quite enough for 4 port with ILBC. my problem is i setup the same
thing with same config in several country like singapore, bangladesh
1) When an incoming call to my DID number is initiated, a prompt is played so
that the caller
can enter an extension number or
zero for the operator. However, at least 30%-50% of the time the digits that
are entered from
the touch tone phone is slightly
different from what is received by
Then try the following in zapata.conf:
echotraining=800
echocancel=yes
echocancelwhenbridged=yes
as a starting point for each fxo channel.
Does echotraining *improve* echo cancellation at all? All I've ever found it
to do is help the canceller converge faster. i.e. if the echo
1) When an incoming call to my DID number is initiated, a prompt is
played so that the caller
can enter an extension number or
zero for the operator. However, at least 30%-50% of the time the
digits that are entered from
the touch tone phone is slightly
different from what is
This is a known issue with livevoip.com service. It's my opinion this
is really a design issue within asterisk, but Mark disagrees.
Your are correct - I do not agree with Mark but, he has never replied to
any emails about this.
The problem is * must answer the incoming iax call from
I tried to found documentation about openloop disconnect on
Asterisk/Zaptel. And up to now, I didn't find anything. Is openloop
disconnect supported by zaptel/wcfxo drivers?
Yes, it works for me and have verified by watching a voltmeter placed
across the pstn line and noting a
I just installed a new asterisk box with a wctdm with 4 FXO modules. The
lines
in the office have terrible static (using standard analog phones) and this
static can obviously be heard through the asterisk box on the sipura sip
phones
we installed. This by itself would not be a problem as
I can't believe that the 7960 doesn't have a hot keypad. That has to be
one of the more annoying things I've heard.
Can you point me to a good dialplan.xml example that I can use on my phones?
Nope, you have to create your own based on what numbering scheme
you require, and what you've set
card. how would I go about getting the cards on
different interupts if they are on the same one?
Tom
Quoting Rich Adamson [EMAIL PROTECTED]:
I just installed a new asterisk box with a wctdm with 4 FXO modules. The
lines
in the office have terrible static (using standard analog
Thanks for your help Rich,
I think it was a combination of poor line quality and shared IRQs and a couple
setup mistakes (oops), we set busydetect=no in zapata.conf (it wasn't there
before), and that seemed to clear up the 1 ring problem, then we got the fxo
card on its own IRQ, and that
Might try modprobe zaptel then modprobe wcfxo (or wctdm). The order
makes a difference and I don't remember exactly which one comes first.
Thanks!
I've edited /etc/zaptel.conf to be
fxoks=4
and then ran
[EMAIL PROTECTED]:~# more /proc/interrupts
I have been able to setup three different providers successfully, but only
one at a time. I would like to have all active in a fail over configuration
so that one failing would not be noticed by the users. I know it's probably
easy to configure but I have not been able to find out how. Can
I'm VERY new in using VoIP. I'm looking for any tip or trick to connect a
physically PABX behind an Asterisk-System(or similar) via an SIP to Analog-
or ISDN-Converter. The point is, I _need_ to deliver calls to extensions in
the connected PABX directly (in ISDN-speech DDI (DirectDialIn))
So, can I take it that most admins are using one provider or doing the
switch over manually when there is a problem? I have been testing voipjet
and it has good quality, how has the reliability been?
There really isn't a reliable way to accomidate all potential failures
via automated means.
Does someone have a working config file they could send me?
In /etc/zaptel.conf put something like this:
defaultzone=us
fxsks=1
loadzone=us
where =1 is the fxo module for the pstn line. (I don't recall for
sure, but if the fxo module is in module position #4, then I think
you'll need
If several phones register to the same sip.conf section what will happen
with a Dial SIP/shared in asterisk?
All phones ringing and the first one to answer gets the call?
Undefined behavior?
I believe the last one to register will be handed calls destined to
that extension.
If you want
It doesn't arrive. It's all done instantly via email.
There's a whole package apparently (hence the £150 postage I was quoted,
although I suspect they just weren't interested in selling).
Even the entry on voip-info.org says it takes two weeks... Once you buy
it the request goes to
Ok, I just got my 8 channel setup to dial out and back in but here is the new
issue. It sill
dials in fine with all the channels, but dialing out from
inside the asterisk system only works on the 1st channel of my 1st TDM400P
card.
Now I dont have all 8 PSTN lines going into my
When someone calls into our * system over a PTSN line, we answer with
a recorded prompt. (Thank you for calling, etc..)
The first second of this prompt ALWAYS skips. After that, everything
sounds great and works perfectly. There is nothing wrong with the
prompt.
Yeah, there's
No, that's a service, or at least I think it is, the sales garbage obscures
what it really is so who knows.
What I need is a little box that diverts calls if the PBX goes down.
FYI, the topic has been discussed previously on the list, and the
problem that you're trying to address is far
Has anyone come up with a way to get power to a TDM400P card installed
in a Dell PowerEdge 1750?
The TDM card only needs the external power connector if fxs modules
are installed. The fxo modules don't use it that power.
If fxs modules are present, only the 12 volt lead is used. Therefore
I've spent many hours to make my 2 Fritz PCI v2 work with Asterisk :-)
I was not able to make them work with the fcpci drivers (even with
custom driver modifications).
The solution was to use mISDN (with chan_capi) instead of fcpci.
You have a guideline at
Somewhere in the Wiki I read that the best way to adjust the rxgain and
txgain is to dial a type 102 milliwatt test line.
This line is usually found in xxx-958- or xxx-959- ranges.
I'm in area code 323 in Los Angeles.
Does anybody know the test number here??
The number assigned
Is it possible to have 2 (working) iax2 phones behind port restriced nat?
Interesting you ask, since I just had an incident concerning this. I
have an IAXy and got an IAX hardphone which I tested at home behind
the same NAT. Using IAX soft clients before in this situation, they
would
Cross posted on purpose
FYI, just upgraded from cvs-head from March 23 to this morning (March 31).
All compiles and installs completed normal.
Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the
standard oche... message. Piped the output to a text file and it
appears the
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium it'll be their own inability to
engineer reliable hardware.
I appreciate what Digium has done for Asterisk, but reliability expectations
for phone
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium it'll be their own inability to
engineer reliable hardware.
I appreciate what Digium has done for Asterisk, but reliability expectations
for phone
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium it'll be their own inability to
engineer reliable hardware.
I appreciate what Digium has done for Asterisk, but reliability
This subject has come up about every two months for the past year
or more, and the exact same answers still apply. If you want a forum,
go set it up; it ain't going to happen at digium.
Others have already set up forums; go find them and use those.
I completely agree.
I do not claim/pretend to speak for everybody on this list, but I *do*
think that others that promote web forums should not do so either...
Hear hear!!
Let's let it die, folks; there are more pressing issues to deal with.
It's true that as long as the Digiumites hang out here, it's
I read in the archives a number of discussions about livevoip, DID,
and DTMF not working.
However, no resolutions.
I just setup a livevoip DID and indeed the DTMF does not work.
The same asterisk context works via broadvoice and via
direct dialing in to the asterisk server via SIP.
I am starting to use livevoip but when I configure they way they suggest, I
see errors.
[livevoip]
exten =_51NXXNXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
snip
Heres the error message:
-- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-6, 1000|15) in new stack
Mar
Is there a way to specify the codec in the dial plan for an outbound
call using IAX?
Sure, just use something like this in iax.conf:
[diamondcard]
type=peer ; outgoing calls only
host=1.2.3.4
username=myuserid
secret=mypassword
disallow=all
allow=gsm
Then in your Dial statement, simply
I read in the archives a number of discussions about livevoip, DID,
and DTMF not working.
However, no resolutions.
I just setup a livevoip DID and indeed the DTMF does not work.
I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
Thanks to all who responded.
DTMF still doesn't
iax.conf:
[general]
bandwidth=high
allow=all
jitterbuffer=no
tos=low
register = 1234567:[EMAIL PROTECTED]
[livevoip]
type=friend
secret=1234567890
deny=0.0.0.0/0.0.0.0
permit=217.160.244.186/255.255.255.0
context=from-livevoip
sip.conf:
I have dtmfmode=inband for both
I'm trying to get firewalling working but I am clueless as to which ports
I need to open, I keep opening more ports and it's not working :(
Basically I want SIP and IAX2 to work. IAX2 works fine, but SIP is giving
me a headache. It seems that the stateless firewall is not able to handle
The apparent packet loss you are seeing may be just fine tuning
of the routers in question.
This is the conclusion I came to as well; however, with the way
PingPlotter works the router is not sending ICMP unreachables but rather
ICMP TTL expired responses. In any case, the routers in
I've got a problem with my incoming calls (SIP). First I tried to route
different providers to different extensions in which ._ matched the call
and called the internal phones and so on.
Then I got this Nikotel Account. I managed to get it working. Small hint
for the people trying Nikotel
No, I'm not ignorant of how this works. You'll notice I put it
appears bad when I posted my results. Yes, it's not a perfect way to
show problems -- but taken with a grain of salt it's not half bad.
Especially when sampled over a longer period of time, and if the
original poster can correlate
If I were to buy 20 did's how do I know within asterisk which number was
dialed? (like say I want a few of the did's to ring specific extensions
if they are dialed and others to go through the menu)
Is there any ${var} that has the number dialed in on? (that would be
optimum).
It varies
Argh. I can't figure out what I'm doing wrong. I can dial with my SIP
phones just fine, but I want to set up an analog phone plugged into my FXS
port... and, while it gets dialtone, no matter what digit I press, I get
stuff like:
VERBOSE[21963]: -- Starting simple switch on 'Zap/1-1'
While on my network I can register ok with xlite but outside my firewall my
Xlite says that
regestraion has failed but I am still able to make calls
through it. I have opened ports: 5060 udp/tcp and 1-2 udp/tcp is
there another port
Xlite needs for proper regestration? Is is this
I am using CVS latest
Is it correct there is no jitter buffer for SIP (RTP)
Are there any plans for this?
prob a stupid question:
Is it required / do the endpoints handle this - if the
src and destination are both SIP and there is no
transcoding but asterisk is still in the media
The story so far:
Some of us fail to get DTMF via livevoip IAX. Others get
a little, others get a lot.
here is a 'iax2 debug' call with version CVS-v1-0-04/04/05-11:22:55
Still no recognition of DTMF by asterisk (at least the IVR doesn't
respond). If you search for DTMF below
BV allows unlimited incoming, and up to 3 outgoing. My understanding
is that they intend to charge for more 3 outgoing, but have not done
so at this time.
This is good to hear--do you have anything from BV that documents this?
Also, being relatively new to *, I don't know if there is
The story so far:
Some of us fail to get DTMF via livevoip IAX. Others get
a little, others get a lot.
I get similar behavior with the [demo]. Works via broadvoice,
myphonecompany or direct SIP dialin. No response to DTMF when
called via IAX Livevoip.
(though the
The story so far:
Some of us fail to get DTMF via livevoip IAX. Others get
a little, others get a lot.
here is a 'iax2 debug' call with version CVS-v1-0-04/04/05-11:22:55
On Mon, Apr 04, 2005 at 01:49:52PM -0600, Rich Adamson wrote:
As you noted, the above
Is there any way I can send callerId information to livevoip? I have
added the following to my extensions.conf, but when I place calls
through livevoip, no callerId information is sent to the called party.
SWC_CALLERID=14031234567
SWC_CALLERNAME=foo
exten =
I don't understand you're confidentiality arguement. If asterisk is
switching the call, it /can/ save a copy of the transmission.
Of course, we know that. But the perception is that the fax machine is
private, so that's what the clients want.
None the less, you should be able to switch a
Cisco TAC service told me that they will not support RFC 2848/3265 for
the 7960 phones
So no busy status line notification with subscribe/notify system. This
is really a bad news for me.
So they are not planning to backport sip firmware new features to the
old phones.
Since the 7960
I'll be damned... I changed my format to match yours, and both
the SetCIDNum and SetCIDName work just fine. I could never get
the name to work properly prior to your post. Thanks!
I am able to set name and number with Livevoip. Make sure your
variables are actually being set.
exten =
I have also seen this problem on two different asterisk servers using
TDM400p cards.
I have not been able to resolve it. If you do an lspci you can see that
the system can see the devices but the zaptel drivers don't see them.
I have other systems that work fine and so this has to
I'm just curious if someone had/has a problem with livevoip. When I try
to make an outgoing call, I receive:
-- Called username:secret@217.160.244.186/x037378896
Apr 2 16:47:21 WARNING[10153]: chan_iax2.c:5546 socket_read: Call
rejected by 217.160.244.186: No authority found
The
Inline...
I keep hearing DTMF type beeps when on phone calls, I know this is some sort
of trait of VOIP but it's driving me nuts..
Not really.
I noticed that it happens MUCH more when I am on the phone with one
particular person.
We are using SPA-2000's from Sipura on both ends.
I'm
I wish to configure my Sipura with static IP. I have
set the static
IP, but there is registration failure on doing so. Could you please
tell me how do I go about configuring my Sipura for static IP and register it
successfully
with the Asterisk server.
A few of
Yes.
http://www.voip-info.org/wiki-Channels+and+Groups
A channel that belongs to a pickupgroup, can pickup
all incoming
calls on the same callgroup by hitting *8
Thanks answering me, that works with the *8 (and *02
th e pattern in my company works too) but there is a
problem : how
On the iax2 show registry I only see an entry for my SixTel account,
no livevoip.
This is all I received from them on my account activation:
Example for your dial plan:
exten =
_1NXXNXX,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,2,Hangup
Does not say
What you are asking for (in US terms) is directed call pickup.
Asterisk does not have a directed call pickup implemented
within it. Not sure how one would try to implement that, but
a guess would be that it would require an external script
or app of some sort.
Actually I've just
I use the West Coast server. It is located in San Jose.
IP Address: 217.160.244.186
As to the replies, I usually get good replies by sending my questions to
[EMAIL PROTECTED]
Have also had good responses from [EMAIL PROTECTED]
BTW - When I signed up I got an email that had all of my
Ok... I've done a bit of emperical testing but don't really know what the
results mean. I'm
starting to think I need an oscilloscope to measure this properly. All I have
is a DMM, I'm
measuring on both the AC and DC scales...
AC MeasurementDC Measurement
my setup consists of an asterisk server with a TDM400P and a couple of
softphones (SJphones) ... everything works well, but the sound coming
from the analog line is really reaaly quiet, even though everything
local (echotest, voicemail, sip to sip) works fine. In fact, I have to
dial up
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