Hi all,
Information on this topic seems a little scarce, so I thought I'd try
the list
Apart from the the coolness factor can anyone explain to me in what
situation one would use TDMoE rather than IAX for communication
betwwen 2 Asterisk servers?
__
I was dloading cvs over the top of a stable branch... (Matthew told me that
was a no-no...)
I've made progress but I'm still havng some troubles. I can get my devices to
register but when I try to make a call, I get silence and some SQL failures.
Below is console output from bootup to registration
Hey gang,
Getting ready to run some test bills for customers. Most SIP phones have
both an extension and a DID. If a person calls a DID asterisk redirects the
call to the right extension:
exten => 8005551212,1,Goto(companyA-internal,3022,1)
The problem is, that if someone calls 8005551212, the C
Folks,
This may be not
directly related asterisk, but hope some experts can help
here.
How would one start
offering a calling program based on 10-10 dial around basis. Are there companies
who could provide a 10-10 number just like a 800 DID.
What kind of
infrastructure needed for this
I'm having a similar problem. Do you have "operator=yes" in your
voicemail.conf under [general]?
http://bugs.digium.com/bug_view_page.php?bug_id=0003080
I think the expected behavior isn't what is programmed.
-Matthew
- Original Message -
From: "Thomas Niesel" <[EMAIL PROTECTED]>
To: "
Some info from the Cisco lddefault.txt file. I hope this helps you
understand the cfgfmt compiler.
# --
---
# This file "lddefault.txt" is provided as a convenience for upgrading the
# Cisco 7905G IP Phone with minimal effort
On 17/12/2004 22:21 Jon Bebeau said the following:
Let me jump in. Seems that the ChanSpy "patch" worked just fine in
pre-1.0.x. Provided MOH plus a bunch of there useful stuff. Now it
seems it's gone in 1.0.3 and scant little info on why or when (or if) it
will be back.
i've applied the Chan
> What is the best way to have a complete configuration for SIP/IAX friends,
Voicemail and Extesions??
> Should I use the new Real-time External config?
Yes. =)
You can find info on setting it up on the wiki.
-Matthew
___
Asterisk-Users mailing list
[
Hello all,
(Not sure if this is more appropriate for user or dev list)
Does asterisk have any sort of "standards based" api that can enable
an application to do call control on the switch ?
For example, if I am developing a call center application
using asterisk, I would like to be notified of inbo
Hello all,
(Not sure if this is more appropriate for user or dev list)
Does asterisk have any sort of "standards based" api that can enable
an application to do call control on the switch ?
For example, if I am developing a call center application
using asterisk, I would like to be notified of inbo
On Sunday 19 December 2004 01:41, Nabeel Jafferali wrote:
> Thanks for all the info so far!
>
> > Therefore a NAT device between two IAX systems has only a
> > single channel, on a well-known port number, to deal with,
> > and this is simple to do.
>
> So then how does IAX deal with the equivalent
On Sat, 18 Dec 2004, Doug Langley wrote:
> I see from reading the mailing list theres a way to set audio levels on
> the zap channels but I'm wondering if there's a way to set audio levels
> on either sip or iax channels. I'm using some BT-100's and people are
> saying the audio levels are a litt
I see from reading the mailing list theres a way to set audio levels on the
zap channels but I'm wondering if there's a way to set audio levels on
either sip or iax channels. I'm using some BT-100's and people are saying
the audio levels are a little low and I would like to bring them up a bit.
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone
I'd installed Asterisk, the app_radius.so and Freeradius. They all
startup ok, but what's next to configure Asterisk to use radius? I
looked at the WIKI already but am still confused. So what should I
set in the radius server and what do I need to set on the radius.conf,
extensions.conf, sip.con
WHy not just get the Polycomm IP 600 they are better phones and not
much more money and do not require a special cable.
Here is few dealers
http://www.apr-usa.com/store/store/comersus_viewItem.asp?idProduct=525265
http://www.tritechcoa.com/product/515647.html
http://www.c-source.com/tt
http://www.voip-info.org/wiki-RTP+Silence+Suppression
http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html
So I am confused. The first says that VAD is supported in RTP.
Ok, I know that. The second is kinda ambiguous and seems to imply that * doesn’t
supp
Tracy R Reed wrote:
On Thu, Dec 16, 2004 at 01:11:34PM +0100, Roy Sigurd Karlsbakk spake thusly:
I think there is a bit more difference. The byte code of ulaw is a
monotonic function of the amplitude whereas in alaw the code is xor:ed
with a bit mask of 0x55.
Wow! Encryption!
Scary t
Nick Bachmann wrote:
Jim Gottlieb wrote:
I've been testing both T400P and TE405P boards and I'm running into
some kind of hard limit on the number of simultaneous calls. This is
on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1.
Everything is fine up to 190 channels, but the 191st call fail
On Thu, Dec 16, 2004 at 01:11:34PM +0100, Roy Sigurd Karlsbakk spake thusly:
> >I think there is a bit more difference. The byte code of ulaw is a
> >monotonic function of the amplitude whereas in alaw the code is xor:ed
> >with a bit mask of 0x55.
>
> Wow! Encryption!
Scary thing is, it would be
On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote:
> I have * running on Mandrake 10.1 and I to had similar problems in the
> begging but as soon as I had ztdummy configured correctly everything
> seemed to just fall into place and work with IAX and *, not that I have
> got a perfect dialpl
> Thanks. On a related note. The problem that I am troubleshooting has to do
> with an IAX connection to TELIAX. Outgoing calls are perfect. When I have
> incoming calls they are very crackly and break up. I have checked the
> jitter buffer and it is not overrunning so it doesn't appear to be
> >>Do you need any more facts?
> >
> > Sure, getting closer...
> >
> > Help us understand what "calls over sip" means. From what device to what
> > device when the call is bad (need to understand the path that you're
> > talking about including any transcoding going on (if any), what type
> > of
I have a channelized T1 with the first 12 channels set to FXS_GS. In my
extension.conf file, I have a variable in [globals] DIALOUT=ZAP/g1. The
problem is when I try to make an outbound call, the console tells me
that everything is busy, but is I change the variable to DAILOUT=ZAP/1,
I can dial o
> > Cisco phones use udp ports 16384-32776, while Xlite uses something like
> > udp ports 8000-8050, and Polycom phones use another range, etc. If you
> > worked for a large company that didn't have any sip phone standards and
> > you had to open everything that _could_ be used for rtp, then you re
On Sunday 19 December 2004 02:00, Keith O'Brien wrote:
> Since the incoming stream is using VAD, my assumption is that it is losing
> the timing during the pauses in the speech. Does anyone know of a way to
> just turn off VAD in *? This would have multiple benefits (if you have
> the bandwidt
Keith O'Brien wrote:
Thanks. On a related note. The problem that I am troubleshooting has to do
with an IAX connection to TELIAX. Outgoing calls are perfect. When I have
incoming calls they are very crackly and break up. I have checked the
jitter buffer and it is not overrunning so it doesn't
Fresh outta the oven, I have uploaded the latest CVS versions to the usual
ftp site. These packages have several updates including file permission
fixes so safe_asterisk will run properly, rpm upgrades should perserve the
current configuration and zaptel is now compiled against the smp kernel
(2.
Thanks. On a related note. The problem that I am troubleshooting has to do
with an IAX connection to TELIAX. Outgoing calls are perfect. When I have
incoming calls they are very crackly and break up. I have checked the
jitter buffer and it is not overrunning so it doesn't appear to be a jitte
Thanks for all the info so far!
> Therefore a NAT device between two IAX systems has only a
> single channel, on a well-known port number, to deal with,
> and this is simple to do.
So then how does IAX deal with the equivalent of SIP reinvites? Or are
all IAX calls' audio carried through the * se
> Try this:
>
> canreinvite=no
As I mentioned in my initial email, I tried that, and adding that line
eliminated 1 of 2 problems. The other problem, that of one-way audio
when a call is carried into the server from and IAX gateway provider to
that SIP client, will not go away.
--
Nabeel Jaffera
- Original Message -
From: "Nabeel Jafferali" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, December 18, 2004 5:51 PM
Subject: [Asterisk-Users] One-way audio with SIP client only on certain
calls
Hello.
I have an * server set up on a public IP. I have SIP clients at three
Citat Keith O'Brien <[EMAIL PROTECTED]>:
> The URL you are looking for is:
>
> http://www.voip-info.org/wiki-Asterisk+timer
>
> Thanks. After reading through the notes I checked my server (Dell 1750)
> and
> noted that it uses a USB OHCI interface so the first option doesn't appear
> to be an o
Nabeel Jafferali wrote:
This is somewhat related to my other query on the list regarding NAT
traversal.
I have heard many times that IAX is "NAT-transperant". I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port
The URL you are looking for
is:
http://www.voip-info.org/wiki-Asterisk+timer
Thanks. After reading through the notes I checked my server
(Dell 1750) and noted that it uses a USB OHCI interface so the first option
doesn’t appear to be an option. Also it indicates that the s
Patrick Conroy wrote:
Can anyone out there confirm as in "Yes I am doing this right now"
that this can be done? I know that the stuff from 2.6 was backported
to 2.4.26 - and it works there (so says the wiki) but before I buy a
bunch of hardware (or don't buy hardware, depending on how you look
Hi Folks
Since updated to 1.0.1/2 I got a prob with the hotkey to
access voicemailmain.
According to the wiki
"0" jumps to extension "o"and"*" to "a"
"0" isn't working, I get vm-sorry followed by HangUp :(
"*" is working and I get access.
So I changed the dialplan to get my voicemail man
> As Per The WiKi:
>
> IAX sends both command information and voice data over the same
> connection. This allows it th transverse a NAT seamlessly.
>
> As for Double NAT, My setup is:
> Home PBX <[Wireless]> ISP WiFi NAT
> <[Ethernet]> Primary NAT <[Ethernet]> Work PBX
>
On Sunday 19 December 2004 00:26, Nabeel Jafferali wrote:
> I have heard many times that IAX is "NAT-transperant". I am unsure how
> it accomplishes this.
>
> I do know that SIP works like this: your SIP device send a request to
> the SIP server (usually on port 5060) with whatever command. The SI
> >> Can anyone out there confirm as in "Yes I am doing this right now"
> >> that this can be done? I know that the stuff from 2.6 was backported
> >> to 2.4.26 - and it works there (so says the wiki) but before I buy a
> >> bunch of hardware (or don't buy hardware, depending on how you look a
As Per The WiKi:
IAX sends both command information and voice data over the same connection.
This allows it th transverse a NAT seamlessly.
As for Double NAT, My setup is:
Home PBX <[Wireless]> ISP WiFi NAT <[Ethernet]> Primary
NAT <[Ethernet]> Work PBX
So, Yes it will wor
This is somewhat related to my other query on the list regarding NAT
traversal.
I have heard many times that IAX is "NAT-transperant". I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port 5060) with whatever co
Hi
I have cross posted this in serusers also...I was wondering if i wanted
to do call forwarding based on times of day, and possibly call blasting,
would it be better to do the routing logic in SER and then pass on the
rewritten number to asterisk to make the call, or would it better done
in aste
--> Sorry, Forgot to attach the files!! :)
I decided, since there was interest, to send them to the who list.
Here they are.
BTW: I am also sending an excerpt from my passwd and inittab files.
the passwd entry is so that i can ssh into [EMAIL PROTECTED] and get an
asterisk CLI.
This "Feature" is wh
I decided, since there was interest, to send them to the who list.
Here they are.
BTW: I am also sending an excerpt from my passwd and inittab files.
the passwd entry is so that i can ssh into [EMAIL PROTECTED] and get an
asterisk CLI.
This "Feature" is why I have "!" disabled.
I hope that this h
What about the limit of 200 Zap channels ? The server doesnt want to create
the channel 201...
- Original Message -
From: "Nick Bachmann" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Saturday, December 18, 2004 6:23 PM
Subje
Hello.
I have an * server set up on a public IP. I have SIP clients at three
different locations, all behind NATs. I have all the SIP users set up
this way:
[user1]
type=friend
username=user1
secret=password1
callerid="User 1"<101>
host=dynamic
qualify=yes
context=outgoing
All three SIP clients
On Sun, 2004-12-19 at 12:25 +1300, Matt Riddell wrote:
>o ztdummy (in the standard zaptel distribution on the
> Asterisk CVS) on 2.4 Linux kernel uses USB-UHCI timers in USB drivers on
> platforms with UHCI USB support...
Uses USB on kernel 2.4, but not on 2.6. On the latter it's su
On Sat, 2004-12-18 at 14:55 -0600, Steven Critchfield wrote:
> I highly suggest you work on getting either the RTC or USB driver loaded
> to provide timing if you don't already have a PSTN card for that job.
OK, this is all softphones and one AVM passive BRI card here, so no
digium hardware. And
David Ishmael wrote:
Is it possible to send the incoming PSTN caller ID to a Grandstream Budge
Tone-100 SIP phone? I've configured the extensions.conf file and the log is
definitely showing the incoming caller ID information, but its not being
sent to the SIP phone (assuming that's possible). I'v
Keith O'Brien wrote:
I highly suggest you work on getting either the RTC or USB driver loaded
to provide timing if you don't already have a PSTN card for that job.
I am having a similar problem. Can someone point me to the procedure to
install these virtual drivers for timing. I searched the wi
Is it possible to send the incoming PSTN caller ID to a
Grandstream Budge Tone-100 SIP phone? I’ve configured the
extensions.conf file and the log is definitely showing the incoming caller ID
information, but its not being sent to the SIP phone (assuming that’s possible).
I’ve searched th
Jim Gottlieb wrote:
I've been testing both T400P and TE405P boards and I'm running into
some kind of hard limit on the number of simultaneous calls. This is
on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1.
Everything is fine up to 190 channels, but the 191st call fails every
time with err
I highly suggest you work on
getting either the RTC or USB driver loaded
to
provide timing if you don't already have a PSTN card for that job.
I am having a similar problem. Can someone point me to the
procedure to install these virtual drivers for timing. I searched the wiki
bu
RaÃl GÃmez Cabrera wrote:
Hi everyone,
We are using the IAXy boxes and Asterisk over the internet and I was
wondering if Asterisk can do a codec translation during a call in order
to lower the bandwidth that the comunications consumes?
I mean, the IAXy boxes only support the ADPCM and uLAW codecs,
<>
Sure. Here's a how-to. If you'd like a working example, let me know.
Open a TCP connection to * either directly of via, say the sampleproxy.pl
example (http://www.popvox.com/simpleproxy.pl).
Login using the manager API by sending the text string on the connection:
"Action: login\n\n"
Aft
I have asterisk auto-magicly start from inittab.
I have yest to try something that can do anything to the server (other
than stop asterisk) from the * CLI.
If anyone would like a copy of the scripts i use, contact me off-list.
BTW: I had already removed the ! command from * before using it this w
On lör, dec 18, 2004 at 05:31:50 +0100, Anders F Eriksson wrote:
> I've never tried softphones on Linux, but my guess is that since you run
> kphone and asterisk on the same server you get a port conflict. If the
> client uses port 5060 (default sip port) it would defenitely have problem
> connecti
Hello list,
I am looking for some ActiveX or - better - Java client that I can embed
in a web page and connect to Asterisk. I would like to implement a "click
here to talk to a live operator" button to be put on a web page. Is there
something already available, both in the commercial and free
Grady:
This might help:
http://www.jimradford.com/asterisk/sjphone/
Regards,
Jim
On Sat, 18 Dec 2004, Grady Trew, Jr. wrote:
> Greetings...
>
> I'm really having a problem getting a SIP client setup on my end. I keep
> getting the "Proxy Authentication Required" popup dialog from SJPhone whe
On 2004-12-17 at 12:13, Vitaly Nikolaev ([EMAIL PROTECTED]) wrote:
> Have you analized quality of the calls ? what was quality of 190 call ? :)
Quality was perfect and a load average of only about 2.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
htt
John Lange wrote:
I have some Cisco 7905 phones with the SIP load 1.02.00(040406A).
When the phone is off-hook but no call has been placed, or when the Do
Not Disturb is activated, the phone returns a 302 "Moved Temporarily"
message back to asterisk as follows:
---
-- Executing Dial("SIP/50
I have some Cisco 7905 phones with the SIP load 1.02.00(040406A).
When the phone is off-hook but no call has been placed, or when the Do
Not Disturb is activated, the phone returns a 302 "Moved Temporarily"
message back to asterisk as follows:
---
-- Executing Dial("SIP/5060-0811bb00", "
On Sat, 2004-12-18 at 18:34 +0100, Bruno Hertz wrote:
> On Fri, 2004-12-17 at 12:01 -0700, Nihal wrote:
> > I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM.
> > 15KRPM Drive.
> > Using the default configs and added one Soft Sip phone.
> >
> > While listening to the demo
Greetings...
I'm really having a problem getting a SIP client setup on my end. I keep
getting the "Proxy Authentication Required" popup dialog from SJPhone when
the number finishes dialing out. It really doesn't matter what the number
is or if it goes out IAX, PSTN, Local Extension, etc. The CL
On Sat, 2004-12-18 at 20:31 +, Antony Stone wrote:
> On Saturday 18 December 2004 20:27, Rodolfo Grave wrote:
>
> > Hi and thanks once more.
> >
> > I moved the card around, and it kept the same IRQ. Then I went into
> > setup and changed it. This is the output of lspci -v now:
> >
> > 01:04.0
On Saturday 18 December 2004 20:27, Rodolfo Grave wrote:
> Hi and thanks once more.
>
> I moved the card around, and it kept the same IRQ. Then I went into
> setup and changed it. This is the output of lspci -v now:
>
> 01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537
>
It's also very encouraging for someone to accidently find some bug they
dont really care about, post information about it and get -2 for not
following up in X hours.
Zoa
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/
Hi and thanks once more.
I moved the card around, and it kept the same IRQ. Then I went into
setup and changed it. This is the output of lspci -v now:
01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537
Subsystem: Unknown device 8085:0003
Flags: bus master, medium d
On Saturday 18 December 2004 20:19, Bill Seddon wrote:
> Detecting the ringing state of a specific device from, say, a desktop
> running Windows or Linux AGI is trivial.
Care to share a trivial example with us?
Sounds like a useful link for several applications...
Antony.
--
Software developm
On Fri, Dec 17, 2004 at 07:54:19AM +0100, Wilson Pickett wrote:
> > I am searching for a new PBX for the company. My choice is Astrisk. My >
> > Boss wants background music via all the telephones. This is done in a
> > conventional PBX that he wants, but I can use the Asterisk PBX if it can
How
Hi Grady, Thanks for the info.
Actually, I had already found this site. Much of what I'm looking for IS
there, but the City / State data is not "real" in all cases; it's more where the
switch is located. Often this is a real City, but not always.
Thanks for the info.
Jon
- Orig
North American Numbering Plan Administration
(NANPA)
http://www.nanpa.com/
Reports -> Central Office Code
Assignment Records
http://www.nanpa.com/reports/reports_cocodes_assign.html
This may be just what you are looking for.
Grady
-Original Message-
From:
[EMAI
Rodolfo Grave wrote:
The X100P card and the SCSI storage controller both have IRQ 15. Is this
what you thought about? What can I do to solve it?
For most systems the onlything you can do is move the cards around.
Motherboards generally assign specific IRQs to specific slots. So if
you move the
Hi again and thanks.
I've found that I have a shared IRQ...
This is the output of lspci -v (only the relevant part):
00:06.0 SCSI storage controller: Adaptec AHA-2940U/UW / AHA-39xx /
AIC-7895 (rev 04)
Subsystem: Adaptec AHA-2940U/2940UW Dual AHA-394xAU/AUW/AUWD
AIC-7895B
Flags: b
Hi Tom, thanks for the heads up, it looks interesting but I'm not sure
whether it would be easier for someone to start from scratch rather than
modify reflector.
I have found these 2 links
http://www.geektimes.com/michael/CU-SeeMe/faqs/reflectors.html
https://sourceforge.net/projects/cuseeme/
C
These are my interrupts... I dont know enough to say if there is
something wrong there.
ghostserver:~ # cat /proc/interrupts
CPU0
0:1377563 XT-PIC timer
1: 13 XT-PIC keyboard
2: 0 XT-PIC cascade
8: 2 XT-PIC r
[EMAIL PROTECTED] wrote:
> Jim Van Meggelen schrieb:
>> [EMAIL PROTECTED] wrote:
>>
>>> - ADSL 1000kBit/s Downstream, 128kBit/s Upstream
>>
>> That upstream bandwitch will need to be managed carefully.
>
> I know.
>
>> If you're using G.711, one channel would be using roughly 80kbit of
>> your upst
Norman Zhang wrote:
Hi,
I compiled * and chan_alsa.so is loaded. But I can't hear any busy
signal messages when calls cannot connect. Do I need to record my own
message, or does * use some default ones? May I ask where can I find them?
Are you using "r" option on the Dial line? That option overr
Hi,
I compiled * and chan_alsa.so is loaded. But I can't hear any busy
signal messages when calls cannot connect. Do I need to record my own
message, or does * use some default ones? May I ask where can I find them?
Regards,
Norman Zhang
___
Asterisk-U
On Saturday 18 December 2004 18:07, Dorn Hetzel wrote:
> I wouldn't say I hate SIP, it sucks less than H.323 and
> so on by a large margin. But, having said that, if you
> can use IAX, it sucks even much than SIP does :)
Um, are you saying IAX is good, or that it is not good? I'm not sure I
u
I realize this may be a dumb question, but is there a standard address
name for the compiler? After looking in the file I noticed that it
mentions a suggestion for it, but on my asterisk machine that suggestion
would not work.
Dan Adams
On Fri, 17 Dec 2004, brian wrote:
atus: RO
X-UIDL: B006671
Rodolfo Grave schrieb:
The thing is that when I run "modprobe zaptel" everything seems to be ok
(I've left the PC running after it long time and nothing happens). Then,
after I execute "modprobe wcfxo" (it gives no messages or warnings) the
PC reboots.
What about the used interrupts? Maybe you'v
On Fri, Dec 17, 2004 at 07:54:19AM +0100, Wilson Pickett wrote:
> > I am searching for a new PBX for the company. My choice is Astrisk. My Boss
> > wants background music via all the telephones. This is done in a
> > conventional PBX that he wants, but I can use the Asterisk PBX if it can do
>
> W
On Fri, Dec 17, 2004 at 07:51:00AM +0100, Wilson Pickett wrote:
> > I'm looking to change from a standard telephone line to a VoIP phone line at
> > home. I'm looking for recommendations for VoIP providers that I can use
> > with
> > Asterisk.
>
> Don't forget about emergency services (lack of)
Jim Van Meggelen schrieb:
[EMAIL PROTECTED] wrote:
- ADSL 1000kBit/s Downstream, 128kBit/s Upstream
That upstream bandwitch will need to be managed carefully.
I know.
If you're using G.711, one channel would be using roughly 80kbit of
your upstream. Who has the most quality complaints: you, or the
Hello.
I have installed asterisk in a IBM NetFinity (single Pentium-II, SCSI
controller, SuSE9.0, one X100P card).
The thing is that when I run "modprobe zaptel" everything seems to be ok
(I've left the PC running after it long time and nothing happens). Then,
after I execute "modprobe wcfxo" (
I wouldn't say I hate SIP, it sucks less than H.323 and
so on by a large margin. But, having said that, if you
can use IAX, it sucks even much than SIP does :)
On Thu, Dec 16, 2004 at 08:53:01PM -0500, Andrew Kohlsmith wrote:
> On December 16, 2004 08:47 pm, Gary Carr wrote:
> > Why is IAX termi
On Thu, Dec 16, 2004 at 05:58:08PM -0500, Gary Carr wrote:
> So they offer termination via SIP for $0.013/minute?
>
Most of the good deals I have found are for IAX termination,
but maybe the same deal are available for SIP.
-Dorn
___
Asterisk-Users
Norman Zhang <[EMAIL PROTECTED]> writes:
> Does performance suffers from this?
There shouldn't be any difference.
> Do I need canreinvite=yes?
The question is, rather, "will reinvites work?". As long as each
local SIP phone is able to initiate UDP communication with an outside
partner, and the
I'm running asterisk 1.0.3 and seeing occasional messages on the
asterisk console that say:
Dec 18 09:09:55 NOTICE[26411]: chan_zap.c:7381 pri_dchannel: PRI got
event: HDLC Bad FCS (8) on Primary D-channel of span 1
or
Dec 18 09:25:46 NOTICE[26411]: chan_zap.c:7381 pri_dchannel: PRI got
event:
-Original Message-
Message: 1
> HI all - I know, slightly off list, but.. I'm looking for a NPA NXX
database with City and State.
The North American Numbering Plan Admistrator has some info at
http://nanpa.com/nas/public/assigned_code_query_step1.do?method=resetCodeQue
ryModel
for th
David Ishmael wrote:
I’m not sure if this is possible, but I was hoping to find an address
book that runs on Windows XP that will allow me to select a phone number
and send that to my Asterisk. The Asterisk system would make the call
and connect the call to a SIP phone (Grandstream Budge Tone-1
Tom Ivar Helbekkmo wrote:
I guess the first few packets from them to you might get dropped
because they don't match an "established" outbound connection, but
as soon as you start sending packets to them, your firewall will
allow two-way flow...
That's the trick, yes. It works because RTP streams l
On Fri, 2004-12-17 at 12:01 -0700, Nihal wrote:
> I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM.
> 15KRPM Drive.
> Using the default configs and added one Soft Sip phone.
>
> While listening to the demo the quality isnt very good. It's kind of crackly
> and skips a b
Rich Adamson schrieb:
Do you need any more facts?
Sure, getting closer...
Help us understand what "calls over sip" means. From what device to what
device when the call is bad (need to understand the path that you're
talking about including any transcoding going on (if any), what type
of sip phone,
Hello, and I'm glad to be a member of this list. Perhaps an enlighted
person can help me.
An entry in my extensions.conf has:
[macro-stdexten]
exten => s,1,Dial(${ARG1},20)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-ANSWER,1,Wait(1)
exten => s-ANSWER,2,Monitor(wav,record)
exten => s-NOANSW
Cisco phones use udp ports 16384-32776, while Xlite uses something like
udp ports 8000-8050, and Polycom phones use another range, etc. If you
worked for a large company that didn't have any sip phone standards and
you had to open everything that _could_ be used for rtp, then you really
would be op
[EMAIL PROTECTED] wrote:
> Hi Noah, I have been contacted by 2 people but nothing so
> far. If you want to add $500 please email me your details and
> I'll add it to the wiki to co-ordinate this.
>
> I agree I'm really surprised why no one has shown more of an
> interest in video calls on asterisk
[EMAIL PROTECTED] wrote:
> Rich Adamson schrieb:
>>> In the past I had problems with the audio over sip. Then I tried the
>>> "-p" Option and increased the memory. Now it is better but not
>>> perfect.
>>>
>>> Are there any more possibilities to increase it more? By now I'm
>>> using a P-II/333.
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