Vincent Li wrote:
> Hello,
>
> I just had a meeting about a pilot project going on in our University, The
> project manager has done some research in the past year and concluded that
> Asterisk can not scale well to large user base like 10,000 users, thus
> Asterisk is not fit for large Universit
On Mon, 16 Mar 2009 23:00:32 -0700
Michael Higgins wrote:
> I have an asterisk server at home. I'd like to test one just
> installed elsewhere.
>
And did succeed just after emailing, of course. :(
Sorry for the noise!
--
|\ /|| | ~ ~
| \/ ||---| `|`
I have an asterisk server at home. I'd like to test one just installed
elsewhere.
Both servers are behind firewalls. I can see the session start in CLI, my
congratulations is apparently playing and RTP is being sent.
Hearing no audio. Can send key presses and see audio playing changed. "Peer
a
Hello'
I am at the same situation as you. I also work at a university and we have
over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot.
I am using a realtime users database and the main problem is that Aaterisk
does too mcuh database access to inquire for the currently reg
Dear Sir,
I am intrigued by your ideas and would like to subscribe to your
quarterly newsletter as well as your annual seminar and leadership
conference.
--
Sent from mobile device
On Mar 16, 2009, at 10:49 PM, drew einhorn
wrote:
> The plastics industry says polycarbonate bottles are sa
Sorry, It has absolutely nothing to do with this list. It was
intended for my wife
and was accidentally sent to the wrong address. I really hope I have not
offended folks that I really want to answer the on topic questions I am asking
on this list.
I'm very, very sorry.
On Mon, Mar 16, 2009 at
Tilghman Lesher wrote:
> On Monday 16 March 2009 21:49:53 drew einhorn wrote:
>
>
>
> What does this have to do with Asterisk?
>
>
I was thinking plastic bottles are just todays version of cups on a string.
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On Mon, Mar 16, 2009 at 8:45 PM, Jason Aarons (US)
wrote:
> Is the feature you are implementing Single Number Reach?
>
> They dial a number and you call another number (Verizon Cell Phone) trying to
> connect them to the user? But the problem is Verizon answers with the silly
> out of reach mess
On Monday 16 March 2009 21:49:53 drew einhorn wrote:
>
What does this have to do with Asterisk?
--
Tilghman
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To UNSUBSCRIBE or update options visit:
The plastics industry says polycarbonate bottles are safe.
http://www.bisphenol-a.org/about/faq.html#g
I'm sure Maggie and here friends would say ALL plastic bottles are
very dangerous.
This lady seems to be at a reasonable middle ground.
http://trusted.md/blog/vreni_gurd/2007/03/29/plastic_water
Is the feature you are implementing Single Number Reach?
They dial a number and you call another number (Verizon Cell Phone) trying to
connect them to the user? But the problem is Verizon answers with the silly out
of reach message? I've never seen where the PSTN carrier lets you re-direct
th
John Novack wrote:
> drew einhorn wrote:
>
>
>> Maybe I don't understand this suggestion.
>>
>> I think your suggestion applys to my sip phones/atas,
>> but they are not the problem.
>>
>> The problem is that when Verizon's network notices the the cell phone
>> is currently not on their netwo
Which if you follow my solution will still ring to the other phones/devices.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Use the M option to accomplish this (I'm 1.2 here) if you use 1.4/1.6
then there might be an easier solution, not sure.
On Mon, Mar 16, 2009 at 8:58 PM, drew einhorn wrote:
drew einhorn wrote:
> Maybe I don't understand this suggestion.
>
> I think your suggestion applys to my sip phones/atas,
> but they are not the problem.
>
> The problem is that when Verizon's network notices the the cell phone
> is currently not on their network, they pick up the call and answe
I may not need a 1 to 1 ratio of phones to sip channels Mostly I'm
trying to be a bit conservative on my estimates to leave plenty of
room for expansion and growth. I'm hoping that I won't need to do
much transcoding (but I shouldn't as long as the phones and the ITSP
use the same codec). a f
I have a possible suggestion -- don't consider the call answered
unless someone types a 1 or something -- makes the dial plan more
complex, but it should work pretty well.
on Monday 03/16/2009 drew einhorn(drew.einh...@gmail.com) wrote
> On Mon, Mar 16, 2009 at 6:47 PM, C F wrote:
> > Good luc
At 19:57 3/16/2009, Eric Fort wrote:
>I'm looking to install a basic asterisk system for my church with:
>
>8 inbound sip channels
>8 sip handsets
>basic voicemail
>room to grow (maybe doubling each of the above)
>
>
>What would be a recomended system as to needed processor and memory?
>
On Mon, Mar 16, 2009 at 6:47 PM, C F wrote:
> Good luck having Verizon change that.
> In the meantime why don't you try implementing a call screen feature
> so that the call is not considered answered until a key is pressed by
> the one answering? That way the caller will still hear ringing until
I'm looking to install a basic asterisk system for my church with:
8 inbound sip channels
8 sip handsets
basic voicemail
room to grow (maybe doubling each of the above)
What would be a recomended system as to needed processor and memory?
Thanks,
Eric
__
On Mon, Mar 16, 2009 at 5:43 PM, Jason Aarons (US)
wrote:
> Nextel does that, pickups up after x rings and says 'The Nextel subscriber
> you are trying to reach is unavailable, please try your call again later".
>
> I'm not sure what Verizon or Nextel called this "feature" or what advantage
> is i
Good luck having Verizon change that.
In the meantime why don't you try implementing a call screen feature
so that the call is not considered answered until a key is pressed by
the one answering? That way the caller will still hear ringing until
the one answering presses that key.
On Mon, Mar 16,
VB wrote:
> If you using cisco why don't you use fax on/off ramp it works quite well.
> Then you can do with the fax file whatever you want.
>
> >From other point of view I did connected 1.6.0.6 with spandsp-0.0.5 to PRI
> and receivefax seems to be working ok. The connect speed is low somewhere
>
> I was looking at the aastra 9133i, however I was informed that this phone is
> no longer supported. What are good phones around the $100 - $125 price
> point? (Need POE)
I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet,
support PoE and works with 2.5mm headset.
$110 at voipsu
Nextel does that, pickups up after x rings and says 'The Nextel subscriber you
are trying to reach is unavailable, please try your call again later".
I'm not sure what Verizon or Nextel called this "feature" or what advantage is
it for the carrier to play it versus just letting it ring forever.
Hi,
Is the following behaviour a bug or a feature ?
Using bristuff-0.4.0-RC3d.tar.gz, the call file thereafter produces :
[Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:267 apply_outgoing: At least
one of app or extension (or keyword message/pdu) must be specified, along
with tech and dest in fil
Hi,
I'm having a problem with Verizon Wireless,
I'm hoping someone here knows the right way
to phrase the trouble report so it gets to someone
at Verizon who can solve the problem.
We have DIDs that simultaneously ring on
voip lines, and Cell numbers.
Verizon voicemail is turned off.
Every thin
Dear all,
I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.
The system there will have a max of 12 concurrent calls to PSTN
provided via an ADSL/SDSL link to our VoIP provider i
SIP wrote:
> I believe SNOM 300s do PoE (might have to check that, though) and are
> around $100. We've little experience with them, but we use an office
> full of Snom 320s, and we're nothing but pleased with them. Good
> speaker, good handset, lots of excellent options. And reasonably priced.
2009/3/16 Olivier
> Hi,
>
> I'm rather new to this domain so I may be doing stupid things without being
> concious of that.
>
> I've got a Patton MATA I'm trying to setup as T.38 fax adapter.
> Whenever I connect a fax machine (Dell MFP1815dn) or a phone to it, I can
> successfully send a fax or
On Mar 16, 2009, at 3:53 PM, SIP wrote:
> David Ruggles wrote:
>> I was looking at the aastra 9133i, however I was informed that this
>> phone is
>> no longer supported. What are good phones around the $100 - $125
>> price
>> point? (Need POE)
>>
>> Thanks,
>>
>> David Ruggles
>> CCNA MCSE (NT
Danny Nicholas wrote:
> Sounds like a personal preference to me. Here is the Wiki for SipX.
> http://en.wikipedia.org/wiki/SipX
>
> Reading this, it's just another flavor of the same medicine. Both are
> open-source with Commercial support available.
>
I'd contend that the business model says
I don't know how good Asterisk's GR.303 support, but you could use DLCs as
well. However, that's a lot of complexity and (seemingly) immature
functionality liability to achieve the same end you'd get with a channel
bank. The only benefit is that DLCs are specifically for oversubscription,
wherea
I'll second that.
On Mon, 16 Mar 2009 18:48:10 -0400, C F wrote:
> Channel Banks would be the way I would do it.
>
> On Sun, Mar 15, 2009 at 3:12 AM, Duncan Turnbull
> wrote:
>> Hi All
>>
>> I am looking at a replacement for a hotel PBX which requires at least 60
>> analogue extensions.
>>
>>
David Ruggles wrote:
> I was looking at the aastra 9133i, however I was informed that this phone is
> no longer supported. What are good phones around the $100 - $125 price
> point? (Need POE)
>
> Thanks,
>
> David Ruggles
> CCNA MCSE (NT) CNA A+
> Network Engineer Safe Data, Inc.
> (910) 285-
Channel Banks would be the way I would do it.
On Sun, Mar 15, 2009 at 3:12 AM, Duncan Turnbull wrote:
> Hi All
>
> I am looking at a replacement for a hotel PBX which requires at least 60
> analogue extensions.
>
> I tend to use Sangoma equipment but haven't tried this many analogue
> extensions
Polycom
On Mon, Mar 16, 2009 at 6:24 PM, David Ruggles wrote:
> I was looking at the aastra 9133i, however I was informed that this phone is
> no longer supported. What are good phones around the $100 - $125 price
> point? (Need POE)
>
> Thanks,
>
> David Ruggles
> CCNA MCSE (NT) CNA A+
> Network
The Asterisk Development Team is proud to announce release of Asterisk 1.4.24,
and is available for immediate download at http://downloads.digium.com/
In addition to other bug fixes, this release candidate fixes several crash
issues, and resolved some remaining issues related to call pickup and ca
I know there has been better uptime than this reported, but I figured
I'll share it anyhow:
@pbx:~# uptime
18:39:07 up 621 days, 9:40, 2 users, load average: 0.00, 0.00, 0.00
pbx:~# cat /proc/cpuinfo
processor : 0
vendor_id : CentaurHauls
cpu family : 6
model : 10
mo
I have a weird problem with call using my T1 card. I can make calls fine
using my analog and IP phones, but when I try to initiate a call using a
.call file, I get the following error
-- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
-- Requested transfer capability: 0x00 -
We have used SNOM 360s, @ about $200, but just tried some Grandstream
GXP2000. I like the 360s but the Grandstream is only $79.00, has four
lines, good speaker phone, and will use a $10 cell headset.
YMMV. But it works, and the price is right.
Cary Fitch
-Original Message-
From: a
Your sip.conf should look like this
sip.conf
[procall]
type=peer
username=XX
secret=XX
context=default
and extensions.conf
[default]
exten = 246463,1,Dial(SIP/8003)
you must also have a sip user for 8003 in your sip.conf like
[8003]
type=friend
username=XX
secret=XX
context=outgoing
And don
Mmm, $100-$125 What? USD? CAD? AUD?
If you're willing to a little bit more, I'll strongly recommend Polycom
IP 430. We're using them and they
are absolutely painless (well, except the initial package of 100 of
those which were heavy and caused
some back pain ;p)
Singer
David Ruggles wrote:
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE)
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 da...@safedatausa.com
__
The inbound was working well suddenly stopped working I want all calls made
to the number should answer the extension 8003
On Mon, Mar 16, 2009 at 3:49 PM, Danny Nicholas wrote:
> Just to read this right – you are trying to take an inbound call from
> 888xxx and transfer it to your sip
Hi,
I've been playing with T.38.
I observed that mostly but not always, it's the "calling endpoint" that
reINVITE the other party to drop current SIP/G711 session and start a new
T.38.
But sometimes, it's also the callee party that reINVITE the calling party.
Which is the "standardized" or most
Just to read this right you are trying to take an inbound call from
888xxx and transfer it to your sip extension 8003?
If so,
Are you able to make internal calls to 8003?
Can you transfer other calls to 8003 (exten => s,1,Dial(SIP/8003) )
?
_
From: asteris
nothing the problem persitem
On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee wrote:
> are you sure calls from this provider are going to context 'default' ?
>
> sip.conf
> [procall]
> type=peer
> username=XX
> secret=XX
> context=default
>
> 2009/3/16 Bayardo Sanchez
>
>> i create inbound number
Sounds like a personal preference to me. Here is the Wiki for SipX.
http://en.wikipedia.org/wiki/SipX
Reading this, it's just another flavor of the same medicine. Both are
open-source with Commercial support available.
In the 3 month's I've been reading this forum, there have been discussions
o
Hello,
I just had a meeting about a pilot project going on in our University, The
project manager has done some research in the past year and concluded that
Asterisk can not scale well to large user base like 10,000 users, thus
Asterisk is not fit for large University environment.
The project
At 22:22 3/13/2009, Matt Riddell wrote:
>On 14/03/2009 10:29 a.m., Doug wrote:
>> At 16:10 3/10/2009, Matt Riddell wrote:
>> >On 7/03/2009 4:58 a.m., Klaus Darilion wrote:
>> >> Hi!
>> >>
>> >> What are the typical ways to work around the 64 groups limit?
>> >
>> >What we actu
If you using cisco why don't you use fax on/off ramp it works quite well.
Then you can do with the fax file whatever you want.
>From other point of view I did connected 1.6.0.6 with spandsp-0.0.5 to PRI
and receivefax seems to be working ok. The connect speed is low somewhere
between 2400-9600 but
Bayardo Sanchez wrote:
> in my extension.conf i set :
>
> [default]
> exten = 1246463,1,Answer(SIP/8003)
This should be:
exten => 246463,1,Dial(SIP/8003)
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
are you sure calls from this provider are going to context 'default' ?
sip.conf
[procall]
type=peer
username=XX
secret=XX
context=default
2009/3/16 Bayardo Sanchez
> i create inbound number but i calling and send this error:
>
> [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_i
Hi,
I'm rather new to this domain so I may be doing stupid things without being
concious of that.
I've got a Patton MATA I'm trying to setup as T.38 fax adapter.
Whenever I connect a fax machine (Dell MFP1815dn) or a phone to it, I can
successfully send a fax or talk to the other end.
Whenever I
1246463 is not the same as 246463. Note the missing 1
If you want to match what is being dialed then your extensions.conf
should look like this:
[default]
exten = 246463,1,Answer(SIP/8003)
Bayardo Sanchez wrote:
in my extension.conf i set :
[default]
exten = 1246463,1,Ans
>> On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez <
>> bayardo.sanc...@gmail.com> wrote:
>>
>>> [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
>>> Call from '101396_procall' to extension '246463' rejected because
>>> extension not found.
On Mon, 16 Mar 2009, Bayardo
On Mon, 16 Mar 2009, Olivier wrote:
> 2009/3/16 Gordon Henderson
>
>>
>
>> On Mon, 16 Mar 2009, Marco Sambo wrote:
>>
>>> Hi,
>>> I have a question. How can I configure my sip.conf to make a SIP phone
>> busy
>>> on incoming and outcoming calls? I explain my problem.
>>> When SIP phone receive a
On Mon, 16 Mar 2009, Bayardo Sanchez wrote:
> [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
> Call from '101396_procall' to extension '246463' rejected because
> extension not found.
>
> but the extensin existed
I run 1.2 so the command syntax may be different...
You should be able to get support from the people who sold you the card.
You need to configure 2 files (I'm looking at an old system, so they
have
the zaptel style names).
My files are below - the thing to note is the span 1,1,0,
the second 1 tells you that the span is a timing source, extern
in my extension.conf i set :
[default]
exten = 1246463,1,Answer(SIP/8003)
On Mon, Mar 16, 2009 at 12:06 PM, Pascal Bruno wrote:
> Do you have an extension set for 246463 in your extensions.conf?
>
>
>
>
>
> On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez <
> bayardo.sanc...@gmail.com
Hi,
I'm using an Asterisk box with zap channel as a gateway between PSTN and
an alarm receiver system. The alarm system uses Contact ID protocol.
My problem is that the negotiation fails and I think that the problem is
that "kissoff tone" is cut and the transmitter doesn't recognize it.
Maybe th
Hello,
I am trying to install my E1 card to make a conection with an Ericsson
MD-110 PBX.
I installed dahdi drivers as:
dahdi_hardware
pci::04:08.0 wcte12xp-d161:8000 Wildcard TE121
ran dahdi_genconf and it created all my e1 ports.
On the other side i also configured the pbx to communic
Do you have an extension set for 246463 in your extensions.conf?
On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez
wrote:
> i create inbound number but i calling and send this error:
>
> [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
> Call from '101396_procall' t
i create inbound number but i calling and send this error:
[Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
Call from '101396_procall' to extension '246463' rejected because
extension not found.
but the extensin existed
--
Bayardo Sánchez García
Web Developer - Inter
I've just determined that it IS happening on my box, but why?
I did a packet capture using tcpdump on this very same box and it shows the
correct invite while sip debug shows the wrong values. here's what I see in
wireshark:
No. TimeSourceDestination Protocol
2009/3/16 David Ruggles :
> Is it possible to control the light on a programmable button without the blf
> option? I'm using a programmable button to turn call recording on and off
> and I would like the light to indicate the status.
>
> Thanks,
>
9133i phones are pretty much obsolete, and are not
I'm not getting inbound audio from bandwidth.com. Their engineer said the
invite that they're sending me looks like this:
INVITE sip:+15129616...@67.198.16.18:5060;transport=udp SIP/2.0.
Record-Route: .
Record-Route: .
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0.
Via: SIP/2.0/UD
Hi Tim,
I've made a test with 2 Asterisks and the 2 consoles showed me an HTML
packet sent and one received. This does not work with the SIP protocol.
The idea was to understand what was it for (I suppose someone did it for
some purpose...), then how to use it to improve our solution (es: open
2009/3/16 Gordon Henderson
>
> On Mon, 16 Mar 2009, Marco Sambo wrote:
>
> > Hi,
> > I have a question. How can I configure my sip.conf to make a SIP phone
> busy
> > on incoming and outcoming calls? I explain my problem.
> > When SIP phone receive a call and then I try to call that phone, I find
I have a customer with an Asterisk 1.4 system (r144238 - between 1.4.22-rc5
and 1.4.22 released). It uses SIP to connect to the PSTN via a provider who
is on the same LAN as the box (it is co-located at the provider). They also
have about 20 SIP phones as extensions that connect to the box over the
Is it possible to control the light on a programmable button without the blf
option? I'm using a programmable button to turn call recording on and off
and I would like the light to indicate the status.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7
Paulo Santos wrote:
> I managed to do 10 calls per second, lasting 5 seconds each.
10 or 5, I can't remember...
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On Mon, 16 Mar 2009, Marco Sambo wrote:
> Hi,
> I have a question. How can I configure my sip.conf to make a SIP phone busy
> on incoming and outcoming calls? I explain my problem.
> When SIP phone receive a call and then I try to call that phone, I find it
> busy.
> When SIP phone make a call and
Shaun Ruffell wrote:
> John Millican wrote:
>> Well,
>> lsmod | grep hisax returns nothing
>>
>> plain lsmod:
>> Module Size Used by
>> dahdi_dummy22472 0
>> dahdi 215776 1 dahdi_dummy
>> crc_ccitt 18944 1 dahdi
>> af_packet
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a call and then I try to call that phone, I find it
busy.
When SIP phone make a call and I try to call that phone, I find it avaible
and it r
Gilles wrote:
> Hello
>
> I'd like to build myself an Asterisk server for SOHO use. Intel's
> D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very
> good deal, but I'm concerned about two things:
>
> 1. Will an A400P (from OpenVox, but supposed to be Digium-compatible
> http://ti
On Mon, 16 Mar 2009, Gilles wrote:
> Hello
>
> I'd like to build myself an Asterisk server for SOHO use. Intel's
> D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very
> good deal, but I'm concerned about two things:
>
> 1. Will an A400P (from OpenVox, but supposed to be Digium-compa
On Mon, Mar 16, 2009 at 11:29 AM, David Backeberg wrote:
> On Mon, Mar 16, 2009 at 6:05 AM, dubravko caric
> wrote:
>> fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 ->
>> SIP ATA (T38 enabled) -> fax
>>
>> My question is, how can I know if I'm really using T38? is T38 in
David Backeberg wrote:
On Sat, Mar 14, 2009 at 12:00 AM, Steve Underwood wrote:
Fully open-to-the-public FAX servers tend to get just get a lot of bad
calls, many of them wrong numbers, or voice users. FAX servers for
I've definitely seen that, and have been able to either identify th
John Millican wrote:
> Well,
> lsmod | grep hisax returns nothing
>
> plain lsmod:
> Module Size Used by
> dahdi_dummy22472 0
> dahdi 215776 1 dahdi_dummy
> crc_ccitt 18944 1 dahdi
> af_packet 57100 2
> snd_pcm_oss
Hello
I'd like to build myself an Asterisk server for SOHO use. Intel's
D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very
good deal, but I'm concerned about two things:
1. Will an A400P (from OpenVox, but supposed to be Digium-compatible
http://tinyurl.com/ck6nfu) fit with a P
On Mon, Mar 16, 2009 at 6:05 AM, dubravko caric
wrote:
> fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 ->
> SIP ATA (T38 enabled) -> fax
>
> My question is, how can I know if I'm really using T38? is T38 information
> coming to the other side (because of SIP to IAX conver
On Mon, Mar 16, 2009 at 12:10 AM, Steve Totaro
wrote:
> Again, if I am interpreting this correctly, he is not using SIP. A
> four port card 2fxo/2fxs means to me that he is not using SIP at all.
You are correct. I was confused. It is Zap (zaptel) channel
>
> If by card, you mean some kind of SI
Oh sorry, I wasn't clear.
The IAX protocol has a frame type for sending this URL info.
Skype has an attribute for it.
The intention is (I think) to be able to forward the URL for
the customer (in the corporate CRM system) to the agent
answering a call on a softphone.
Some of the IAX softphones
Hi,
As soon as I removed back line 266 as suggested by Peer Oliver, it worked.
Lines changed in /usr/src/bristuff-0.4.0-RC4-xr6/asterisk/pbx/pbx_spool.c :
/* Olivier
if (ast_strlen_zero(o->tech) || ast_strlen_zero(o->dest) ||
(ast_strlen_zero(o->app) && ast_strlen_zero(o->exten)) ||
(ast
Hi,
I'm currently using Asterisk 1.4.23.1, and I have a problem (also on
previous version).
Sometimes, when I try to do an attended transfer to another internal with
default feature *2, Asterisk doesn't make it (it doesn't play
'pbx-transfer'). Sometimes on second time, Asterisk make transfer corre
Hallo Ralf,
das ist die Antwort von der Liste. Klingt etwas vage und nicht absolut
erfolgversprechend... Derzeit ist kein Upgrade auf Asterisk 1.6 geplant
und nach allem, was ich darüber bisher gelesen habe, kann die Umstellung
auch etwas größer werden (Wählplansyntax etc.). Beronet empfiehlt
derz
On Mon, Mar 16, 2009 at 8:49 AM, Vieri wrote:
>
> Hi,
>
> I am trying to understand why some of my call transfers fail.
>
> My scenario is as follows:
>
> Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
>
> Step1: PBX1 extension 101 calls PBX2 extension 102
>
> Step2: PBX2 extension 102 answ
Hi Tim,
it seems that using trunks is the right wayis this what you meant?
Tim Panton wrote:
> Use IAX :-)
>
> In principle chan_skype could also support it.
>
> T.
>
> On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote:
>
>> Hi,
>>
>> Does anybody knows where I can find some docs about how
Hi,
Trying to trace an asterisk hang on a production (it had to be didn't
it) system. The last thing before it crashed was
[Mar 16 12:32:42] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED
Polarity on channel 1, state 4
[Mar 16 12:54:34] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED
P
Hi Tim,
ok, but I think the big question is...what is the URL for? It seems I
need a special device...but which? What kind of device do you use?
Thanks.
Giorgio
Tim Panton wrote:
> Use IAX :-)
>
> In principle chan_skype could also support it.
>
> T.
>
> On 16 Mar 2009, at 10:51, Giorgio Incan
Hi,
I am trying to understand why some of my call transfers fail.
My scenario is as follows:
Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
Step1: PBX1 extension 101 calls PBX2 extension 102
Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension
103
Step3: PBX1
2009/3/16 Christophorus Laube
> Hi,
>
> does anyone of you have made it to get the ANI also picked up? I mean:
> if I fetch a foreign call to me by using the pickup application I want
> to see the callerID/ANI of the caller to the foreign extension. Is that
> possible and if yes - how do I achiev
dubravko caric wrote:
> Hi all,
>
> I have a question regarding using T38 for fax sending and here is my
> scenario:
>
> fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk
> #2 -> SIP ATA (T38 enabled) -> fax
>
> My question is, how can I know if I'm really using T38? is T38
>
MaxGao wrote:
hi,all
i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to
ReceiveFAX, link to a E1 (DE410P) using dahdi
this can receive the fax from E1 successfully, but i see many error
message in the log like this:
[Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called w
Use IAX :-)
In principle chan_skype could also support it.
T.
On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote:
Hi,
Does anybody knows where I can find some docs about how to make the
URL
parameter inside the Dial command work? I tried to make some tests
with
a sip phone without succ
Hi,
does anyone of you have made it to get the ANI also picked up? I mean:
if I fetch a foreign call to me by using the pickup application I want
to see the callerID/ANI of the caller to the foreign extension. Is that
possible and if yes - how do I achieve that?
Regards, Christophorus
_
Hi All,
Is this available on asterisk:
http://www.ietf.org/html.charters/simple-charter.html
what do i need to enable to support this. thanks
Regards,
Nhadie
___
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asterisk-users ma
Hi,
Does anybody knows where I can find some docs about how to make the URL
parameter inside the Dial command work? I tried to make some tests with
a sip phone without success: the sip debug shows no URL inside sip
packets. :(
Any hint appreciated. :)
Thank you
Giorgio
__
On Mon, Mar 16, 2009 at 09:14:48AM +0100, Rayed Bs wrote:
> Thank you for your attention;
> I have successfully installed junghanns (With BRIstuff) under a kernel 2.6
> fc6 and asterisk 1.2, but i can't do it with B410P in the same
> environnement(Problem with the kernel);but my real problem is in
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