I have a box running [EMAIL PROTECTED] 1.0 and I noticed today when I
went to change some settings to do with the DID’s
that it is no longer detecting the different lines.
I have a Digium 4 port line card
and Im pretty sure that the DID’s
used to work when I used fxs_ks signaling on the
On Tue, June 21, 2005 23:07, Kristof Hardy said:
> Francesco Peeters wrote:
>> I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the
>> pricing for these in Europe, so I'd like to hear from people here
>> whether
>> that is a reasonable price for them?
>
> Prices I know are around
I have the same problem.
seems that tdm400b is not working on CVS HEAD
On 6/18/05, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> did you udate first?
>
> - Original Message -
> From: David Romero
>
> To: Asterisk-Users@lists.digium.com
> Sent: Friday, June 17, 2005 9:36 AM
> Subje
Damon Estep wrote:
I assume the bandwidth is being donated or something, but surely someone
would be willing to donate reliable bandwidth as the knowledge hosted on
the site (which is also donated!) is worth way more than the bandwidth.
Sure it's the bandwidth? If the wiki is loaded, I see "Se
The VoIP Connection wrote:
It's here:
http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/gxp2000_config.txt
Very interesting, it wasn't available at Grandstream's site :-)
Thanks! I will adjust some things on the page now I have the new
template.. http://voip-info.org/tiki-index.php?pa
You could use Agentcallbacklogin instead - the queue will call them when
a call comes in, but they are free to make outbound calls in the meantime.
Julian.
Damon Estep wrote:
Is there a way for a logged in agent (hearing music on hold) to initiate
an outbound call without logging out of the qu
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Callum
> McGillivray
> Sent: Tuesday, June 21, 2005 6:38 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Echo Issues
>
>
> Hi all,
>
{clip}
> Some of the calls we are receiving have
I would also donate some bandwidth…….
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, June 21, 2005 9:34
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
voip-info.org unreliable lately?
I
Hey dave - could I have a quick squizz at your zapta.conf ? Also, what
phones / card / server specs are you running ?
How many users ? We have 20.
If anyone else can offer any suggestions they would be greatly appreciated.
David Phelan wrote:
HI Callum,
I am going thought a similar thing h
Hi all
could somebody help me how to install and setup H323 i
would like to connect asterisk box with huawei/cisco,
but i still dont understand about installing h323 on
asterisk
thaks
__
Do you Yahoo!?
Yahoo! Mail - Helps protect you from nasty
Em (20:59:21), Asterisk Users Mailing List - Non-Commercial Discussion
escreveu:
>On Tue, Jun 21, 2005 at 08:09:52PM -0300, Alessandro wrote:
>>
>> Does Somebody know why no load modules to FXS? I used zaptel-1.0.7
>> version.
>
>What color are the modules?
>Post your /etc/zaptel.conf
I would be willing to donate some bandwidth for it. We already donate bandwidth for the Asterisk CVS mirror./bAsterisk.com/Cluecon.comOn Jun 21, 2005, at 5:19 PM, Damon Estep wrote: Anyone have any insight as to why voip-info.org has been up and down all day, and more importantly unreliable for th
>hi,>>can you someone post tftp template for
gxp-2000?>like >http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txt>>thanks>>--->Marek Cervenka>===
It's
here:
h
HI Callum,
I am going thought a similar thing here with a site that I setup about 6
weeks ago...
There doesn't seem to be any Pattern to it at this stage.
I am *trying* to get the end users to keep a log of calls with echo to see
if it is a specific channel Problem, Inbound/outbound etc.
If I come
The company that is sponsoring Cluecon and founder of Asterlink have
allowed me to post and keep posting many of the kewl and useful apps
we have done so the Asterisk Community can get more cool apps and
toys to play with. I have just posted a pretty good example of how
to use res_js. A
Dear Asterisk Community,
Does your company provide inbound 800# origination? If so, please read
this message and e-mail us a quote for monthly co-lo hosting of our
asterisk server and per-minute inbound 800# origination.
The Prostate Cancer Research and Education Foundation (PC-REF) is a
non-p
Hey everyone,
I am a new user to asterisk, and have encoutered a problem making outgoing calls through my OEM Digium FXO PCI Card.
Basically, I dial a number from my SIP Phone, say
9-410373
Then it picks up my line, and dials
410-777-737..3
or
4-107(jumbled together)-7
Hi all,
We have just installed a * server (CVS head) with a TE110P card and a
IDSN20 line, we are using the GXP-2000 handsets running the latest
firmware (.9).
Some of the calls we are receiving have echo at the our end (we can hear
ourselves speak).
We have a traditional ISDN telephone sy
On Tue, Jun 21, 2005 at 08:09:52PM -0300, Alessandro wrote:
>
> Does Somebody know why no load modules to FXS? I used zaptel-1.0.7
> version.
What color are the modules?
Post your /etc/zaptel.conf
--
Mike
___
Asterisk-Users mailing list
Asterisk-User
yOn Tue, 21 Jun 2005, Matthew Boehm wrote:
> We have a TE110P (single span PRI) and are having tons of echo on all calls,
> both incoming and outgoing. We didn't have any echo at all yesterday and
> nothing in any of the configs has changed.
>
> All of all calls follow this pattern:
>
> Cisc
I actually do not need DNS at all as I refer to all hosts via IP
addresses but Asterisk still seems to need DNS perhaps to do reverse
lookup or something like that..
On 6/22/05, Eric Bishop <[EMAIL PROTECTED]> wrote:
> I actually do not need DNS at all as I refer to all hosts via IP
> addresse
On Tuesday 21 June 2005 18:16, Matt King wrote:
> We are open to suggestion on this issue, so if you've got a way forward I'd
> love to hear about it, but in the meantime I'd like to repeat my request to
How about not crossposting this? It really does not belong in -dev. (Nor
really -users, but
>
> This should work, taken in account the username/pass are
> correct and the hostnome is the one they provided you as SIP
> registry server.
> What looks odd to me is the last two lines.
> Why are you first disallowing all codecs and in the next
> line allowing them all again ? You should either
I answered my own question. I just had to dig a little deeper on the
lists.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Tuesday, June 21, 2005 5:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-User
Which software pack to you have for the IP600? Sip.ld, bootrom, etc...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Johann
|Sent: Tuesday, June 21, 2005 11:54 AM
|To: Asterisk Users Mailing List - Non-Commercia
What does the Zombie mean in this line from the CLI?
== Spawn extension (daymenu, s, 1) exited non-zero on 'SIP/700-fe13'
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
__
Hi all,
Does Somebody know why no load modules to FXS? I used zaptel-1.0.7 version.
/var/log/messages:
Jun 21 19:06:15 darthvaden kernel: Zapata Telephony Interface Registered on major 196
Jun 21 19:06:16 darthvaden kernel: PCI: Found IRQ 9 for device 02:09.0
Jun 21 19:06:16 darthvaden kern
Eric Rees wrote:
I have googled this and come up empty. Has anyone had any problems
compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am
getting when I run make.
app_nv_faxdetect.c: In function `nv_detectfax_exec':
app_nv_faxdetect.c:210: error: structure has no member na
Is there a way for a logged in agent (hearing music on hold)
to initiate an outbound call without logging out of the queue?
We want sales agents to be able to make outcalls when there
is no callers in queue, but still be logged in to get new inbound calls if they
come in.
?
__
We are using * as a voicemail server for a Cisco Call Manager and we
occasionally get the same thing. It can't be callerID in our scenario
since we are using Cisco PRI gateways and there wouldn't be any CallerID
tones on that. It occurs when someone calls a number and it rolls to
the voicemai
Anyone have any insight as to why voip-info.org has been up
and down all day, and more importantly unreliable for the last month?
I assume the bandwidth is being donated or something, but
surely someone would be willing to donate reliable bandwidth as the knowledge
hosted on the site (w
I'm loosing IAX registration after few hours and can not make call using
IAX protocol, reloading configuration does not help. Internet
connection stays up not a problem.
When I reboot the firewall IAX goes through just fine.
What to look for?
--
#Joseph
_
> md5 instead of plaintext?
Doesn't asterisk take care of this automatically with SIP? I have
other providers that use md5 and they all respond with a 401 challenge
and then asterisk generates the md5 and uses the realm given it in the
401.
> Also, I think I just seen a change in the last day o
Hello again :-(
I have a problem with chan_unicall. If I have two simultaneous incoming or
outgoing calls, they sound broken because cpu load goes to 99%. Also with one
call, the cpu load goes to 99%. Seems like device /dev/zap/channel is busy
after 5 or 10 seconds , and chan_unicall does not write
> I'm getting 403 forbidden errors when attempting to register to a
> certain provider. I've tried just about every combination of
> configuration settings I can think of with no luck. Following is what
> I would think should work (and one of the settings I have tried).
> Rather then list every
What Linux version are you using?
There is an ebuild on Gentoo
--
#Joseph
On Tue, 2005-06-21 at 16:15 -0500, Eric Rees wrote:
> I have googled this and come up empty. Has anyone had any problems
> compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am
> getting when I run make.
>
>
Hello Adam, Matt King <[EMAIL PROTECTED]> writes:
I am familiar with the OSI definitiion. I've read it again, but I
can't work out exactly how asking for permission contravenes this
definition.
Then Adam wrote:
"6. No Discrimination Against Fields of Endeavor
The license must not restrict
Sebastian Silva wrote:
To see the entire log, in logger.conf:
full => notice,warning,error,debug,verbose
then:
"tail -f /var/log/asterisk/full"
in other console run asterisk, you will see all log output in the
previous console and why asterisk stops.
Sebas,
The full file did not show an
You can use the astxs util in the contrib/scripts folder.
astxs -install app_valetparking.c
if your asterisk src isn't in /usr/src/asterisk then export ASTSRC=/
path/to/asterisk
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jun
Hi,
On 12:52, Tue 21 Jun 05, snacktime wrote:
> I'm getting 403 forbidden errors when attempting to register to a
> certain provider. I've tried just about every combination of
> configuration settings I can think of with no luck. Following is what
> I would think should work (and one of the set
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Matthew
> Boehm
> Sent: Tuesday, June 21, 2005 1:13 PM
> To: Asterisk Users
> Subject: [Asterisk-Users] Digium Card: Echo, Echo and more Echo
>
>
> We have a TE110P (single span PRI) and are having tons o
Hi, all
I'd installing asterisk 1.0.7 with spanDSP 0.0.2pre18+astfax 1.0 on Fedora
core 2. Fax reception (using RxFax) is working well. I have problems when
sending a fax (it's an image in TIFF G3 format, using TxFax) composed of 2
parts/pages to a fax machine on PSTN, only receive first page but
Francesco Peeters wrote:
I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the
pricing for these in Europe, so I'd like to hear from people here whether
that is a reasonable price for them?
Prices I know are around 99 EUR, incl VAT. But if you ask me, depending
on how many yo
Clock source will be important here. For phase one, you should probably
set asterisk to time from the PBX since the PBX is likely timing from the
T1 circuit. At phase two, you will likely want to reverse this having
your PBX clock from the Asterisk system and having Asterisk clock from the
telco
> >> |Rich is indeed correct, Asterisk does not yet support multiple
> >> |registrations for a single peer entry. Thus when you register
> >> |the previous registration is discarded and the new one is
> >> |used. Thus like he said, the last one that registered gets the call.
> >
> > And asterisk wi
Matt King <[EMAIL PROTECTED]> writes:
> I am familiar with the OSI definitiion. I've read it again, but I
> can't work out exactly how asking for permission contravenes this
> definition.
> 2) OrderlyCalls MAY NOT be used to provide or augment call queuing without
> the prior written permission
it would be very helpfull (IMHO) if you post the output of the
Asterisk console with a high verbosity level. Also, show us how the
important code in your extensions.conf
best regards
On 6/21/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> I have two asterisk machines. One of them has a Digium b
I have googled this and come up empty. Has anyone had any
problems compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am
getting when I run make.
app_nv_faxdetect.c: In function `nv_detectfax_exec':
app_nv_faxdetect.c:210: error: structure has no member named
`cid'
app_n
Sorry, I don't have that info at hand. Unless you can tell me how to
look at it from the command line.
Orlando Guitián wrote:
Andres:
Can you end me the service tag as i would like to have a replicated
config.
Thanks
From: Andres <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED],Asteris
hmmm Caller ID ? that sounds like a modem as a quick
burst
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
CityTechs.NetSent: Tuesday, June 21, 2005 2:58 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk
answers with high pitch sound
Hi,I've google
Unfortunatly it won't compile under 1.0.7 :(
I have uncommented #define AST_10_COMPAT but I don't see any usage of it in
app_changrab.c.
Complains about missing asterisk.h ( I think it should be #include
"../asterisk.h" )
It also complains about ASTERISK_FILE_VERSION() function, and about _
On Jun 21, 2005, at 11:48 AM, Denis Galvão - iSolve wrote:
On 21 de jun de 2005, at 14:18, Jay Milk wrote:
|Rich is indeed correct, Asterisk does not yet support multiple
|registrations for a single peer entry. Thus when you register
|the previous registration is discarded and the new one i
There is no clean or simple way to do that today. You're essentially
looking for a way to disassociate the exten number from the user name,
and allow users to roam from one phone to another. I'm certainly not
saying that is impossible at all, just saying its not a current
approach that can be easil
We have an old Telrad 128KSU PBX to which I am trying to connect asterisk in
the following manner:
Current Setup:
Telco-> T1->PBX
Desired Setup:
Telco-> T1-> Asterisk-> T1-> PBX.
I am first trying to setup the Asterisk -> T1->PBX part without disturbing
existing setup so I can get asterisk to
We have a TE110P (single span PRI) and are having tons of echo on all calls,
both incoming and outgoing. We didn't have any echo at all yesterday and
nothing in any of the configs has changed.
All of all calls follow this pattern:
Cisco 7960 -> Asterisk -> PRI
Here is my zapata.conf
[channe
I'm getting 403 forbidden errors when attempting to register to a
certain provider. I've tried just about every combination of
configuration settings I can think of with no luck. Following is what
I would think should work (and one of the settings I have tried).
Rather then list every combinaton
Hello All
i have 2 problems, please help me
1. How to implenment record call at called side.
i want to record the call by called press the DTMF key.
2. how to implement call out functions, for example: i create .call file and copy to /spool/outging, then when asterisk call out, i want that: wh
We have an old Telrad 128KSU PBX to which I am trying to connect asterisk in
the following manner:
Current Setup:
Telco-> T1->PBX
Desired Setup:
Telco-> T1-> Asterisk-> T1-> PBX.
I am first trying to setup the Asterisk -> T1->PBX part without disturbing
existing setup so I can get asterisk to
I apologize in advance for posting this yet again (3rd time actually). But I
have a little more data to share this time so bear with me.
I have Asterisk running on an internal IP address behind a Cisco Pix 515
with firmware version 5.2(3)
Here is the setup
Mitel SX200 PBX --- Asterisk --- Cisc
On Tuesday 21 June 2005 14:48, Denis Galvão - iSolve wrote:
> Is there a way to just register the phone when user pickup the phone!?
> In this way we can have two phones regitered with the same context.
How would you have asterisk know which IP to ring if nobody is registered
until the phone ring
Hello everyone,
In case you haven't seen it yet, a few of us coming back from Astricon
Europe have uploaded our pictures and created a page on the wiki:
http://www.voip-info.org/tiki-index.php?page=Astricon+Europe+Media+2005
Check it out, and I'll see you all in Anaheim.
P.S. - If all 10,00
>It might be possible to change the values slightly to judge their impact.
>I've not done the math, so not sure if changing the values has any real
>merit.
Yes, I think it does. I'm definitely going to try some tweaks there.
For anybody interested, the reference document that I am using can
be fo
Hi All,
I can get Grandstream 100 SIP phones for EUR 75. I'm not sure about the
pricing for these in Europe, so I'd like to hear from people here whether
that is a reasonable price for them?
TIA & BRgds
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704
If yo
Page cannot be found
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Joshua Colp
|Sent: Martes, 21 de Junio de 2005 08:16 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: SV: SV: [Asterisk-Users] Presence and IM?
|
|A
> >>>Has anyone used ground start on Asterisk? I am using a TE110P connected a
> >>>Adtran 750 channel bank FXO card. It appears that Asterisk is not setting
> >>>
> >>I'm testing with a TE110P and a Adit 600 with GS. So far, it appears to
> >>be working fine.
> >>
> >>I'm using CVS HEAD CVS-v1-0
Ok Steve,
Wich will be the version I will need to install to solve this problem??
Best Regards,
OBS: I am really in Brazil and I am using a R2 E1 from Embratel( Telco
company here in Brazil).
___
Asterisk-Users mailing list
Asterisk-Users@lis
Hi,
I've googled it and look in voip-info.org without any success.
Hope someone can point me to the right direction. I saw a couple
similar questions, but don't see any answers.
Fedora Core 2
2 X100P(clone) PSTN
Asterisk 1.07
Everything seems to be running fine, but on occasion, Asterisk answe
Andres:
Can you end me the service tag as i would like to have a replicated config.
Thanks
From: Andres <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asteris
Stijn,
On 21:17, Tue 21 Jun 05, Stijn Jonker wrote:
> Remco, Michiel & others,
>
> On 21-Jun-2005 0:03, Stijn Jonker wrote:
> > Remco,
> >
> > On 20-Jun-2005 23:08, Remco Barende wrote:
> >
> >
> On 21:11, Mon 20 Jun 05, Remco Barende wrote:
>
>
> >Are you sure that for the B
On 21 de jun de 2005, at 14:18, Jay Milk wrote:
|Rich is indeed correct, Asterisk does not yet support multiple
|registrations for a single peer entry. Thus when you register
|the previous registration is discarded and the new one is
|used. Thus like he said, the last one that registered gets
I'm having a problem with the callerID that the polycom IP600 phones are
displaying. I would like to modify the CIDName and leave CIDNumber as
exactly what the phone call came in as(provided they aren't hiding
callerID). Most of the calls will be going to the queue, but a few will
go directly
True, but also I want this to be the beginning of a small way Im trying to
figure out in order to make virtual extensions, for ex, people can move from
ext to ext and just login and route their exts... Using db, etc.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED]
Hi,
Is anyone successfully using Asterisk in India hooked up to the PSTN?
I have tried "defaultzone=us" and no tones would work at all when
calling the IVR,
but if i set "defaultzone=uk" most but not all of the buttons work.
Does anyone have any tips or tricks for getting TDM / PSTN connectivity
I've been debugging and tracing chan_unicall again, and I found the same
problem reported by Andres Maduro, in Asterisk-Dev list
here is my strace output:
...
22049 write(30, "\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0\0"...,
320) = 320
22049 write(1, "\r\33[1;30;40m-- \33[0;37;40m
Remco, Michiel & others,
On 21-Jun-2005 0:03, Stijn Jonker wrote:
> Remco,
>
> On 20-Jun-2005 23:08, Remco Barende wrote:
>
>
On 21:11, Mon 20 Jun 05, Remco Barende wrote:
>Are you sure that for the BRI outgoing callerid is allowed?
>>>
>>>
>>>Yep, with pain in the heart i rec
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Chris Modesitt
> Sent: Tuesday, June 21, 2005 9:35 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Best Echo Canceller.
>
>
> I know this is slight OT howeve
I released app_changrab.c lastnight really late... It includes a way
to hijack a channel and originate calls from the CLI.
/b
---
Keep Your Friends Close, But Your Enemies Even Closer...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I am having this exact problem today.
I have two Dell 1850's running Asterisk 1.07. Both had TDM400P cards
running just fine. I replaced the TDM400P in both machines with TE410P.
Server One works just fine with just a new modprobe. Server 2 does not
even see the card upon reboot.
Swapped car
I tried this with CVS Head as of today and I get:
-- Executing Dial("SIP/705-0c37", "SIP/g10/768||T") in new stack
Jun 21 11:21:09 WARNING[24473]: chan_sip.c:1742 create_addr: No such
host: g10
Jun 21 11:21:09 NOTICE[24473]: app_dial.c:977 dial_exec_full: Unable to
create channel of type 'SIP
I agree you but i read in the wiki about ast_data
Asterisk, SER and MWI:
I wish to send mwi to clients registered on ser db
I patch sources files by hand however compilation
failed:
c -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENT
Hello Adam,
Thank you so much for taking the time to write to me. I can
understand your concerns; let me see if I can address them.
Matt,
Sourceforge.net is exclusively for hosting software whose licensing
terms meet the OSI's definition of Open Source:
http://opensource.org/docs/defin
Actually SIP has the capability for it... For example, on Free World Dialup
which uses SER you can have up to 24 registered SIP devices to a single
account I believe, may be slightly smaller... But it's still a large number.
Thus when your number is rung, all registered SIP devices are contacted...
On 12:00, Tue 21 Jun 05, Anton Krall wrote:
> Can this hint system be used for gxp2000 phones or just for snoms?
>
Right now the gxp2000 doesn't support it. I heard rumours on
this list that Grandstream is planning this feature for some
future firmware. I'm waiting for it as well. Till that time
> Ok, so how are you guys coping with scenarios like this?
> Managers working in the office during the day or mid day and
> then in the afternoon, working remotely using their laptops?
Give them two extensions and ring them both. One's the hard-phone,
one's the soft-phone.
> |Rich is indeed c
This phone runs symbian, has a built in camera for video conferencing,
blah blah blah. Dunno yet if you have enough to make it a soft phone,
but odds are there is. Could be another gsm alternative to also do voip
[InfoWorld: Top News] Motorola adds Wi-Fi to 3G phone for NTT DoCoMo
http://www.info
But what do people do with large LCR rules... Build special contexts for
each peer/user and then include the main LCR context? This seems a
little cludgy.
Is there any way to have the dialplan context set the account for cdr
based on the accountcode defined in the sip.conf? At least this way I
Trey Scarborough wrote:
- Original Message - From: "Mark Johnson"
<[EMAIL PROTECTED]>
To:
Sent: Tuesday, June 21, 2005 8:56 AM
Subject: [Asterisk-Users] Cisco 7750
I have read of people attempting to do this, and I just wanted
everyone to know about what we've discovered about the
But what do people do with large LCR rules... Build special contexts for
each peer/user and then include the main LCR context? This seems a
little cludgy.
Is there any way to have the dialplan context set the account for cdr
based on the accountcode defined in the sip.conf? At least this way I
Outlook cut the subject... Damn MS..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Dave Cotton
|Sent: Martes, 21 de Junio de 2005 11:28 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] app_changrab.c
I tried with several iax softphones:
iaxcomm
idefix
iaxphone
and i have a problems that i do not have with SIP clients.
A calls B, B phone starts ringing, asterisk says that call has been
accepted, that is ringing but it is not yet answered. If B "picks up",
asterisk says that call has been an
Can this hint system be used for gxp2000 phones or just for snoms?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Joshua Colp
|Sent: Martes, 21 de Junio de 2005 10:03 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: S
Hi
I try to create a sip client with asterisk-api package,
I've a question:
I can create a channel sip that generate sip signaling with Class Channel
or with another
class ?
Thanks Ale
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To see the entire log, in logger.conf:
full => notice,warning,error,debug,verbose
then:
"tail -f /var/log/asterisk/full"
in other console run asterisk, you will see all log output in the
previous console and why asterisk stops.
Sebas
Ronald Wiplinger wrote:
I have now a very strange situa
I know this is slight OT however I have decided that I need to but in some
echo cancellers on my PRI's. I was wondering if anybody else was using a
hardware echo canceller capable of 24 T1's, how well it works and an
approximate price range:)
Thanks
Chris
__
I have two asterisk machines. One of them has a Digium board (server
A) and the other is simply using ztdummy (server B). Server A is
running on Debian and Server B is running Gentoo. Server A is running
Asterisk CVS-Nv1-0-7-06/01/05-01:27:25 and Server B is running
Asterisk 1.0.7.
The pr
On Tuesday 21 June 2005 10:38, Goolsby, Daniel S (Daniel) wrote:
> Could you just configure the extention to be a ring group instead of an
> actual extention, or ring queue.. then have his phone/laptop log in
> whenever he's at the office/coffee shop?
As someone else pointed out if you want to kee
On Tue, 2005-06-21 at 10:20 -0500, Anton Krall wrote:
> Where can We get it from?
>
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf Of
> |Brian West
> |Sent: Martes, 21 de Junio de 2005 09:11 a.m.
> |To: Asterisk Users Mailing List - Non-Commercia
I guess I would need to do something like that and mix with dialing 2
extension at the same time with dial(ext1&exte2)
Seems the easier way to do it for now.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich Adamson
|Sent: Martes, 21 de Junio de
Robert Rozman wrote:
Hi,
I'm getting unreliable dtmf recognition (it works fine for 4-5 digits,
errors (duplicates) on more), when transferred inband from gsm gateway
to NT port of quadbri under bristuffed Asterisk.
Since Asterisk is claimed to have good dtmf recognizer, I suspect
ther
On 6/13/05, Steve Davies <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I am using a number of snom190 phones, and an asterisk "gateway"
> server, and recently started experimenting with call transfers. The
> snom phones provide support for attended and un-attended call
> transfer, so I would rather use tha
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