The company I work for has gotten the go ahead to start dipping its foot
into the shallow end of the asterisk pool. The client we will be setting
up for currently has an NEC PBX of some sort with 8 analogue lines in.
They use lines 1-4 as indial on a rotary group, lines 5-6 as indial for
two 1800-
Hi,
I have the following requirements I'd like to implement with asterisk:
1. Asterisk notifies interested PC's on the network that there's an
incoming call so that the telemarketing app can bring up the customer
automagically
2. If a telemarketer makes a call and the customer isn't there and
the two together
Comments?
Thanks
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Harper
Sent: Monday, 10 January 2005 14:43
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] telemarketing application
Hi
I think no one should abuse Asterisk and make it into a telemarketer
tool. In fact, it is designed to supposedly drive telemarketers away!
There's telemarketing and then there's telemarketing. Everyone's opinion
is different but I think the type you are referring to are probably the
ones that
If I were going to bet a couple bucks on this, I'd suggest the spa3000
will
disappear alltogether, and a replacement in the form of a linksys box
with
a faster processor is not far behind.
One of the features I'd like to see in such a box is TDMoE support (eg
it would be a 'mini' channel
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Darrell Long
Sent: Saturday, 28 January 2006 05:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Nagios and Asterisk
Is anyone using Asterisk
My home asterisk system has failed the wife test (still too much echo
with the current hardware... my voice seems okay but when she talks she
complains of echo), but I'd still like to be able to use it to send and
receive fixed line sms messages to and from my mobile.
My x100p card has 2 ports on
Juan Carlos Castro y Castro wrote:
How many TDM2400P cards can I safelly install in one PC? I'm loking
for
answers from whoever has a working scenario with * and a number of
cards
higher than one.
Depends on the specs of the server. For example, a quad Xeon will be
able to service
I am trying to set up a linux based faxing solution for a client, and
have found that the modem they have (ancient dataplex external unit)
just isn't up to the job. It talks to some remote fax machines but not
others.
A new external modem ranges from AUD$75 to AUD$400, which got me
thinking of
virtually all software echo cancelers cannot get double echo removed
completly. It can get the first one but not the second one. There
are
instances where you get a 2nd echo, so ... Asterisk is no exception
from this afaik nothing software only based is.
If you really want good echo
Has anyone every attempted to set up a mini PC to achieve much the same
functionality as the fonebridge box?
The sort of thing I'm imagining is a micro itx board case in a
completely solid state configuration (flash disk, maybe a psu fan but
only if really required), with a TE210P (or equiv)
I am running 2 TDM400's in a single machine without a drama. It is a
scenario that will work, but you have to be careful with your PCI
slots
and IRQ assignments. I basically disabled everything on board that was
not needed (USB, floppy, IDE2 etc) and had to play around with which
slot the
3 TDM cards here, I had artifacts if any of the cards were sharing
interrupts, the trick was to add the cards 1 at the time to get them
each on their own irq. The system isn't in production yet, so I don't
know how well it'll hold up under load, so far so good in testing
though.
9xFXO
RAD appear to have bucketloads of products which bridge between various
interfaces (E1, BRI, POTS) and their own TDMoIP protocol. The attractive
thing about them for me is their availability in Australia.
The voip wiki says not much about it
(http://www.voip-info.org/wiki/view/TDMoIP), and
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDMoIP and Asterisk
James Harper a écrit :
Does anyone know anything more about using Asterisk and TDMoIP together?
Well, these boxes have an E1 interface. So you should be able to use a
Digium
James Harper a écrit :
I want to do it the other way around.
Asterisk---TDMoIPRADE1Telco
You'll need 2 RAD boxes, i.e.
Asterisk - RAD - TDMoIP - RAD - E1 - Telco
I think you missed the point of my question, which was to know if there was any
attempt to make
To my knowledge, it hasn't been done.
Thanks :)
The fonebridge would do what I want, but I can't get it in Australia,
and
if I imported one, I wouldn't legally be able to attach it to the
phone
system.
No, the phone bridge does TDMoE, not TDMoIP.
I actually want TDMoE, in order to make
I have finally stumbled across a PCI Multiport BRI adapter available in
Australia, but I have a sinking feeling that while it _may_ support
CAPI2.0, the drivers are probably closed source.
Can anyone please confirm or deny that the HST chipset has open sourced
drivers and works with Asterisk
What about the fonebridge (http://www.red-fone.com/fonebridge.html)?
It uses POE, so you could hack something together to supply 48V @ 15W if
you don't have access to a power point, and it appears to have a 2 port
switch built in so you could still chain to a VoIP phone (or more if you
daisy
I may have found a source of an A-Ticked HFC 4BRI PCI adapter in
Australia, and will be testing one next week if all goes well. I don't
want to post the details of the reseller online unless invited to do so,
so if nobody replies and says they are interested then I won't :)
I'll follow up once
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Harper
Sent: Saturday, 4 March 2006 12:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] MultiBRI in Australia - found one - maybe
I may have found a source of an A-Ticked HFC 4BRI
Best of luck :-D
I would be interested in your progress on this.
I am having very little problem in convincing ppl to upgrade their
multiple
BRI cricuits for a single pri. The cost difference between a te110
(or a
Sangoma A101) MORE than covers the difference from the customer stand
[EMAIL PROTECTED] wrote:
Noted - I may need to grab one for an install coming up.
Just a heads-up: If you plan to do any faxing with the board, don't
get
the V-4BRI which is voice-only. Get the bigger, badder 4BRI, which
supports voice, fax and data.
Does this allow the faxing to be
I have been involved with a BRI install using 3 x Draytek minivigor
128
BRI
adapters and chan_mISDN. The draytek units use the HFCS-USB chipset,
are
USB and take power from the USB interface. Each adapter will support
PTP,
PTMP, TE and I think NT mode with a maximum of 8 adapters (16
Note that
it is only the currently available minivigors that have the HFCS-USB
chipset, older ones on the secondhand market and eBay most likely use
a
Winbond chipset.
Is there any chance that they would sell me an old one? Do I need to ask
specifically that they supply the HFC one?
Thanks
Hi!
As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use iLBC to connect to my *. My gateway provider only
allows ulaw. Hence, I'd like to use the follwing setup:
SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway
I get the following error:
Unable to
Just to follow up on my previous comment,
/usr/share/doc/asterisk/copyright contains the following on my Debian
system:
* The iLBC codec library code has been removed from the Debian asterisk
package as it does not conform with the DFSG.
James
___
I did it back in the xen 2.x days with a BRI adapter (Traverse NetJet).
It worked fine for the testing I was doing.
I'm not sure of the status or performance of the PCI mapping through to
DomU these days, but that should be the only extra step required.
James
-Original Message-
From:
On 6/12/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
I don't know of any other GE phones.
However...
Why in the world would you ever need GigE sip phones?
I think the advantage would be in the
Hi all,
How come I occasionally get messages with the subject NOT containing
[asterisk-users] ?
I'd be interested in an answer to that one too...
It stops my filters working!
Try filtering on some of the other headers, eg:
X-BeenThere: asterisk-users@lists.digium.com
List-Id:
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Asterisk is latest Debian 'etch' packaged 1.2.13. sip.conf looks like:
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Instead of SIP debug, try capturing the traffic with tcpdump etc. on
I've bought a Sipura SPA 3000, and succesfully connected it to my Mac,
where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well
configured).
However, living in Brazil, I'd like to know if there are optimal
settings
to my PSTN that I should enter into the config of the device. I
A dialplan of '(S0:s)' will get your phone to jump straight into
the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
Just remember that having Asterisk supply the dialtone does add (a
slight) additional load,
I tried one of these and pretty much got it working under visdn. If you
do decide to try one, make sure you get the HFC version. Earlier ones
used another chipset and definitely weren't supported using open sourced
drivers.
Please post back if you do get one and get it going though.
Thanks
I'm getting messages like 'WARNING[10263]: chan_sip.c:2552 sip_write:
Asked to transmit frame type 8, while native formats is 1024 (read/write
= 1024/1024)', where 8 = alaw and 1024 = ilbc.
If I do show translation I get this:
*CLI show translation
Translation times between formats (in
I've configured our PBX so that when a user dials 80 on the PBX
extension, it goes out an ISDN TE interface on the PBX and into an NT
interface on my asterisk machine, where it jumps into the 's' extension.
Asterisk then does a DISA(no-password|sip_provider_out) which allows the
call to go out
Is there any way I can change the dialtone that DISA produces within
extensions.conf?
I have a PAP2 configured in 'batphone' mode, so as soon as you pick it
up it jumps into an 's' extension. Asterisk then does a DISA() to accept
the number (I like keeping my dialplan centralised).
I have
James Harper schrieb:
I tried one of these and pretty much got it working under visdn. If
you
do decide to try one, make sure you get the HFC version. Earlier
ones
used another chipset and definitely weren't supported using open
sourced
drivers.
Please post back if you do get one
: Re: [Asterisk-Users] MultiBRI in Australia - found one -
maybe
We didn't ask specifically for new ones. I believe the old ones went
out
of
stock a long time ago. We ordered four at once and they all came with
the
HFC chipset.
Craig
- Original Message -
From: James Harper
I have successfully placed a call into and out of the card from Asterisk
using vISDN. The current vISDN snapshot now contains the PCI id's for
the card so no patching should be required.
Most of the initial testing failure was due to misconfiguration of vISDN
by me and a bad entry in my sip.conf
Got my 2 dreytek adapters today...
Dropped them on to my test system. After wadding thru my Memory of
how to
setup mISDN, I had it up and running within about 2 hours.
You might be receiving an email from me shortly then if I get stuck. If
it wasn't for these annoying public holidays
please post the brand/model of the card and
supplier/cost etc.
Cheers,
Mark.
On 3/13/06, James Harper [EMAIL PROTECTED] wrote:
I have successfully placed a call into and out of the card from
Asterisk
using vISDN. The current vISDN snapshot now contains the PCI id's
for
the card so
Definitely one line per FXO port, but the wording of the original poster
was two numbers, not two lines, and while it may not be universally
true, distinctive ring should allow two (or more) phone numbers to be
present on an FXO port, and asterisk should be able to tell which one is
calling.
If
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of David Phelan
Sent: Tuesday, 14 March 2006 13:28
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one -
maybe
This is more an isdn question than an asterisk specific one, but is
there any end to end signalling channel available during call setup? Eg
if AParty dials BParty, can any information be conveyed (in both
directions preferably, and in addition to CLI[PR]) before the call is
answered?
The only way
One more thing. Cisco 7905 phone that is working is 74-3092-04 Rev.F0.
Cisco 7905 phone that is not working is 74-3092-08 Rev.A0.
Anybody know about any hardware issue with this revisions?
Nothing for sure, and you may already know this, but some early Cisco
phones only knew how to speak
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t)
exten = _3XX,3,Hangup
Why do you Answer before you Dial here? I had a problem where calls were
misbehaving and someone asked me that same question. Without really
understanding why I removed the Answer and it then just worked.
I
If you feel like updating the wiki, It definitely works for me in
Australia.
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tim Robinson
Sent: Saturday, 18 March 2006 02:56
To: Asterisk Users Mailing List - Non-Commercial
Is anyone in the world making gsm 'picocells' which could be connected
to an Asterisk server and allow gsm mobiles to roam to them (and
therefore become just another extension) when in the office?
Obviously lots of things to consider (it's a licensed band) which I
think was the big holdup last
Hi James, how would you feel about writing a quick howto and extension
configuration for SMS in Australia. There is very little information
on
Google or voip-info as to how this could be done. I have tried myself
however I keep getting the message from Telstra as opposed to the
actual
data.
Siemens makes them, as do a few others. Googling should provide you
with
the manufacturers, and ebay has some used equipment for sale.
Care to give me any more clues? Google only wants to tell me about
articles about the use of picocells in aircraft and how much better the
world will be when
On 3/17/06, James Harper [EMAIL PROTECTED] wrote:
Care to give me any more clues? Google only wants to tell me about
articles about the use of picocells in aircraft and how much better
the
world will be when it happens :) Maybe I'm using the wrong search
terms.
They will all tie
I believe the OP wants to use GSM handsets as extensions, like running
your own localized GSM network. That's not the same as using a GSM
terminal to connect Asterisk to the cellular network.
Correct!
IP Access makes such products.
http://www.ipaccess.com/products/nanoBTS.htm
That looks
I believe the OP wants to use GSM handsets as extensions, like
running
your own localized GSM network. That's not the same as using a GSM
terminal to connect Asterisk to the cellular network.
Correct!
IP Access makes such products.
http://www.ipaccess.com/products/nanoBTS.htm
I'm setting up an asterisk server to allow our PBX to make calls out via
VoIP, but when it calls out I get this message:
chan_capi.c: did not find device for msn =
(eg no msn)
Which would be correct because at that point I've only asked for an
outside line.
I'm using CAPI obviously, and my
Actually, for something like Asterisk, that has so many different
aspects, a Forum would be a much better idea. Then, each piece of
hardware can have its own category, along with an FAQ.
Please no. A forum might be okay if you have a nice fast web connection
and/or a bit of patience, and if
Hmm,
I was using 0.3.0 rc24, or the unstable branch. I see 0.2.0 is listed
as
'stable' so maybe I should have used that. Please do keep me informed
of
your progress.
Craig
After finally getting chan_misdn to load (missing #include to bitops.h
under Debian at least) it still won't load,
I've just found my first problem. /dev/mISDN was being created with the
wrong permissions...
Thanks
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of David Phelan
Sent: Monday, 20 March 2006 12:18
To: 'Asterisk Users Mailing
One thing that may help:
I use outlook rule to move all the messages into a folder.
Then outlook has a feature, instead of sorting by date, or subject,
you
can sort by conservation.
Me too. Except, for some reason it often misses the first message of the
conversation (especially if I'm
I have several locations, each connected by a Sonicwall VPN through
PPPOE DSL, with Snom 360 phones.
I've found that I have to tweak the Asterisk server MTU (inside one of
the firewalls) to get everything to work just right. Set the server
MTU too low, and the Snom phones don't communicate
Steve,
Excellent explanation.
In a nutshell, it might be better to just use a phone that can
automatically switch between GSM and WiFi. Of course, that's limited
to
handful of handsets.
I haven't done any sort of research, but I've been told that GSM+DECT
phones are available, and while
manufacturers (nokia, Panasonic, sonyericsson) but their web sites are
absolutely pathetic to the point being useless (or maybe I'm just in a
bad mood today :)
Thanks
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Harper
Sent
Not GSM/DECT but GSM/Wifi phones are available - This is not a
recommendation, I don't like what I've seen.
It strikes me as really strange that GSM/Wifi would be available while
GSM/DECT is not so much. DECT is a voice technology, while wifi isn't.
Still... there's a lot about the world I
Appreciate the replies everyone -- really
I'm wondering if I should be using zapHFC with my Junghanns card
instead
of
qozap? Everyone always mentions zaphfc -- mostly I guessed because
they
are
using a zaphfc-compatible card - but *maybe* I should try that
instead
of qozap???
And
I was given the challenge recently of creating a LAN-LAN bridge
between
two
buildings several
hundred metres from each other, using only existing Cat 3 wiring and
without
having to resort
to an expensive and finicky 5 Ghz wireless link. I was able to create
a 90
megabit link for
about
yeah that's what came up before when I asked the list about this a
couple
months ago. The concensus was that in the case of a lightning strike
or
what
have you the 24 awg copper would immediately fry and would not
transmit
too
much of the current sustained. of course my neat little trick
Turning on hyperthreading may have changed the way interrupts are
routed. Were you using the same kernel (eg SMP kernel even with
hyperthreading disabled)? The BIOS may have configured things
differently too if you disabled it there.
I'm not sure, but you may be able to keep hyperthreading on in
Ummm... you can probably ignore most of what I said in my last email. I
just noticed that you said your machine has two physical processors, so
even with ht disabled you will still need an SMP kernel.
I'll pay more attention next time :)
James
-Original Message-
From: [EMAIL
I assume you mean this:
http://en.wikipedia.org/wiki/ISAC
but maybe you are referring to one of the controller chips on BRI
adapters?
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Trond G. Andersen
Sent: Thursday, 4 May 2006
I think it's most likely that it's a mail loop caused by a brain dead 'change
of address' script.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Alberto Sagredo
Sent: Saturday, 13 May 2006 17:00
To: asterisk-users@lists.digium.com
Hi,
I am investigating getting a wifi VoIP phone because its may be a
better
option than an ATA and a cordless phone..
Does anyone have any experience with the whats out there??
Do they support things like WPA etc??
I have heard the battery life can be a problem.. Is this the case?
On 22:32, Thu 18 May 06, Craig Guy wrote:
From the picture on the web site it looks like it uses a cologne
chipset.
Any idea if these cards will be available in Australia?
Can't you just order them from the digium website?
Or is digium not shiping to Australia?
To legally connect any
Last time I checked with Telstra about 3 months ago, at 7 channels (eg
3.5 BRI services), a 10 channel E1 service (OnRamp10 from Telstra) is
cheaper than BRI in terms of monthly line rental (a fair bit more
expensive to install though). So if you actually need 4 ISDN ports / 8
lines to connect to
Also you can use the unstable branch of debian, all things are near
ok, from the asterisk core to the kernel.
Bye
It may have been 2 years since I worked with Debian on production
systems, but in my experience there are alot of unstable packages in
unstable. So it's a bad advice to run
I have a hfc-usb adapter connected to a Samsung PBX, and am having a bit
of trouble.
With capi, using 'immediate' mode, I can get the dialplan working just
fine, except that there is some sort of interference in the audio, and
I'm not quite sure how to describe it except that it's sort of like
Just noticed that the poor audio quality only occurs when the call goes
out via SIP. If I go straight to voice mail or something it's fine.
Does that ring any bells?
Thanks
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James
Further to my previous email, I have definitely established that the
audio gets choppy only when the path includes sip and capi.
PAP2 to Asterisk to MyNetFone to PSTN is fine.
PAP2 to Asterisk MOH is fine.
PBX (via capi) to Asterisk MOH is fine
PBX (via capi) to Asterisk to PAP2 is choppy
PBX
FYI, I was having problems getting chan_misdn to work, it just wouldn't
get the extension in immediate mode. chan_capi got the extension okay
but the audio quality was awful.
In the end, I put a Wait(0.01) before Answer in the incoming mISDN
context, then DISA(no-password|sip_provider_out) and
My dial plan as shown below is,
[capi-in]
exten = s,1,Dial(Sip/123,20)
exten = s,2,Voicemail(123)
exten = s,3,Hangup
I believe I should be able to receive calls with the above.
With immediate = yes then you should.
I have also tried the following, and i get the same problem and debug
Does anyone know of a hardware adapter that can take ISDN BRI frames
(I.430) and encapsulate them in Ethernet (any form, but TDMoE would be
really cool), in much the same way that the redfone does for PRI?
(yes I have asked this before in looser terms, but it was a while ago :)
Would anyone find
this very well, but where is the BRI voip gateway?
Hawk
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-
[EMAIL PROTECTED] On Behalf Of James Harper
Sent: Tuesday, June 06, 2006 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT?
maybe it will fit for you? if yes, i think you can work with the
following
budget:
via epia board ~85$
mini itx case (small size!) ~85$
ram ~20$
DiskOnChip (or HDD) ~20 - ~50
HFC BRI ~50$
so globally ~300-350/side
you can also go for patton
Do you mean receiving traffic on 2 BRI lines (2 channels spread on 2
separate ports) connected to 2 differents boxes so that one line or
box
failure wouldn't affect incoming calls ?
If positive, do these providers price this service (2 ports - 2
channels)
at an intermediate level between
I'm a bit confused about exactly what isn't working... you have given
the asterisk receiving parts of extensions.conf, and say that when you
send a message from the phone to * you get a 'no data' message, but then
say that * is able to receive messages from the phone and that sending
to the phone
I tried putting in a delay like you suggested, but it had no effect.
How exactly did you put the delay in? It should be:
Answer
Wait(1) (or .5 or whatever - just play around with it)
SMS(...)
The original extensions fragment you posted didn't even have an Answer
in it. I'm not sure if that is
'sa' would appear to be the right option, as Asterisk in your case is
answering the call as the message center (the phone is the 'terminal
equipment')
Would the pap2 be doing anything funny like waiting for fax tones or
something before letting the tones go through?
What happens if you just pick
I want to have something for the kids to play with which just records
until silence is detected, plays back what was recorded, then repeats.
They are having fun with Echo() at the moment :)
I have mocked something up with:
exten = *93,1,Answer
exten = *93,n,Record(/tmp/echo:alaw|1)
exten =
Easy to do on the Linksys PAP2, if that helps. The functionality
probably depends on the make and model of the phone... maybe if you gave
those details as well?
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Is there a way to get a report from Asterisk on the quality metrics
(packet loss, delay, jitter) of at least the inbound component of a SIP
call?
Thanks
James
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
James Harper wrote:
Easy to do on the Linksys PAP2, if that helps. The functionality
probably depends on the make and model of the phone... maybe if you
gave
those details as well?
James
Fantastic, this may solve the problem In the mail I've just posted
(which hasnt' appeared yet
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote:
Ideally I would have liked the pap2 to have done the same as
'immediate'
when talking about fxo, capi, misdn, etc, but I couldn't get it to
automatically dial nothing. A '0' was the best I could do. If anyone
knows how to put
Are there any known problems with Cisco routers (Cisco 837) and SIP
sessions? I have been trying to track down a problem for about 3 hours
now and I think the Cisco router is the culprit!!!
I keep getting 488 Not acceptable here messages, which are apparently
normally the message you get when a
Of James Harper
Sent: Monday, 12 June 2006 00:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Cisco router and 488 Not acceptable here
messages
Are there any known problems with Cisco routers (Cisco 837) and SIP
sessions? I have been trying to track down
On Jun 11, 2006, at 8:15 AM, James Harper wrote:
Additionally, just to satisfy myself that I wasn't going mad I
changed
the port from 5060 to 5070 and now things are working, so something
is
definitely playing up on port 5060.
If you are behind a NAT perhaps two SIP devices are both
I'm looking at setting up an ISDN internet service for someone, and
she'd like to be able to do VoIP. The modem (230kbps serial and 2 POTS
ports) you get from the ISP can do DVO (Dynamic Voice Override) where
you can be online at 128kbits/sec (2 channels), but if a voice call is
detected (call
James Harper wrote:
Additionally, just to satisfy myself that I wasn't going mad I
changed
the port from 5060 to 5070 and now things are working, so something
is
definitely playing up on port 5060.
James
You probably have are behind NAT and your NAT device has a SIP ALG.
Changing
I now have another problem with sending messages back to the phone,
when I run:
smsq -o0198339100 -q101 --mttx-channel sip/phone1 --ud test
You need to run smsq as the Asterisk user, or else the file is created
with permissions that Asterisk can't read and/or move.
James
There is a spec for echo cancellation on PSTN called g.168. I believe
it's a
suite of tests which put the echo canceller through its paces and if
you
pass
them you are certified to conform to g.168. None of the echo
cancellers in
zaptel conform to this, whereas the Octasic, Tellabs and
I believe this is no longer be true with the new Native music on
hold...
Ah. I suspected as much, when my home server wasn't running any zaptel
for a bit I found that MoH was working fine, even though last time I
tried without ztdummy a while ago it was severely broken.
(The reason I
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