Hi,
please always add asterisk version to your query.
I managed to run internet radio (that streams MP3) within asterisk.
Minor change is nescesarry to make it work with random MP3s.
My Dialplan:
exten = _X.,n,Answer()
exten = _X.,n,MP3Player(http://stream.m-1.fm/m1/mp3)
$ cat /usr/bin/mpg123
Hi, I had to choose between an 8 port FXS device from Cisco/Linksys (the sipura
3000) and a similar device from Grandstream. A look on the Grandstream's forums
had me scratching my had, so much people with problems, frequently needed
restarts, etc.
The next thing, the Cisco/Linksys seems to be
Hi,
we have and Asterisk server connected to a Patton Smartnode 4638 with
4 BRI.
We configured 4 SIP account on Patton (1001, 1002, 1003, 1004).
The system is fully functional, but we have a problem to recognize
incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on
Patton) or
Il giorno 30/apr/10, alle ore 10:01, A.Santoro ha scritto:
Hi,
we have and Asterisk server connected to a Patton Smartnode 4638 with
4 BRI.
[...]
Hi Eco,
I think the problem is in your sip.conf.
Have you tried setting insecure=port,invite in the sip.conf for each
sip account?
Bye,
2010/4/30 A.Santoro n...@ecoricerche.it
Hi,
we have and Asterisk server connected to a Patton Smartnode 4638 with
4 BRI. [...]
Hi Eco,
I think the problem is in your sip.conf.
Have you tried setting insecure=port,invite in the sip.conf for each sip
account?
Bye,
Carlo
--
--- On Fri, 4/30/10, Raimund Sacherer r...@runsolutions.com wrote:
Hi, I had to choose between an 8 port
FXS device from Cisco/Linksys (the sipura 3000) and a
similar device from Grandstream. A look on the Grandstream's
forums had me scratching my had, so much people with
problems,
On 30 Apr 2010, at 09:41, Vieri wrote:
As far as having an internal fan for cooling, I don't know if that's actually
better... In general, these devices shouldn't need to rely on mechanical
cooling which tends to fail in time (sure, you can open the case and replace
it but that's extra
Hi all,
How can I implement a full-featured Call-Waiting behavior on the Asterisk level
(e.g. I don't want to relay on end-equipment capabilities)?
I found it very strange that such a basic feature is not built-in in Asterisk
(and I've googled a lot in search for this).
Here is what I need:
In my previous company we bought about 30 Grandstream GXP2000 phones.
The build and design quality of those phones were terrible (not to
mention firmware bugs).
Speakerphone and headset ports were unusable.
The external powersupply would only last a year or two before it failed.
The screen was
Hi all the people in the list!
I'm new on this list, this is my first post.
I configured asterisk 1.6 with freepbx 2.7 and dahdi to send faxes with
t38modem conected to hylafax as a sip extension of asterisk.
Everything is supposed to be configured fine, the faxes start sending, but
at the middle
Hi,
I have IAX trunks between Asterisk servers. They receive calls on ISDN cards
and Dial() through the IAX trunks to the primary Asterisk server where all
the SIP phone extensions are registered.
The IAX trunk settings are something like this (all servers have this identical
except for the
Hello
I've got small PBX (30 simultaneous connections) based on asterisk
(1.6.2.6), which uses Stargate 2N ISDN to GSM gate.
It runs ok for day or two, but then I get:
dahdi: HDLC Receiver overrun on channel TE4/0/1/16 (master=TE4/0/1/16)
in my kernel logs, in asterisk i get:
pri show spans
Hi!
calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton)
or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming
from SIP/1004.
Read up on how Asterisk does user/peer matching in sip.conf on inbound
calls: With all users/peers having the same IP and hostname
i'm having the same problem with one of my call centers located in Egypt..
although the ip-phones are located on a Dedicated Leased Line yet calls drop
out of the blue.almost an identical setup as yours..provider located in France
(data center) my server located in Sweden (data center) both on
Hello,
I consider buying three GrandStream GXW4024 and connect 72 analogue
phones to asterisk
Do you have any feedback how well it works with Asterisk ? I am on a
budget, do you have other recommendation for similar setup that get into
same budget - connect around 70 analogue phones to
Tim Nelson wrote:
Greetings all-
I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks
rather interesting. Has anyone used one? Where did you purchase it? Pricing?
Operational issues?
http://spidermux.com/
Tim Nelson
Systems/Network Support
Rockbochs Inc.
On Fri, 30 Apr 2010 10:39:12 +0200, Carlo Dimaggio
jaasmail...@gmail.com wrote:
2010/4/30 A.Santoro n...@ecoricerche.it
Hi,
we have and Asterisk server connected to a Patton Smartnode 4638 with
4 BRI. [...]
Have you tried setting insecure=port,invite in the sip.conf for each sip
account?
Hi, all
how to get hold event in asterisk.
is it possible, when user1 put on hold in queue moh1 file played.
when call transfer to agent and answered agent put hold at that time
moh2 file played ?
I have used asterisk 1.4 version.
Regards,
--
Bhrugu Mehta
Sr. S/W Engineer (DD)
VOIP,Telephony
Am 30.03.2010 20:56, schrieb Richard Kenner:
You need promiscredir set to yes on sip.conf
And then what do I do in the dialplan? I.e., what context is the
redirect number interpreted in? Google searches on this issue show
inconsistent and contradictory information.
I usually set the
The disconnect is RECEIVED by Asterisk. So there is a problem with the
other party.
You are sending FACILITY - maybe the other party does not like FACILITY
and hangs up.
IIRC there is a setting in zapata.conf to enable/disable FACILITY.
regards
klaus
Am 10.04.2010 21:46, schrieb bruce bruce:
Hi,
I have one in stock - got it from a client who wanted to get rid of all
his old IT equipment.
Looks strange, did not have enough time to play with it Tried it
once, looked hard to configure.
It stays unused in the storage room.
Peter
On 29.4.2010 10:20, Tim Nelson wrote:
Greetings
On Fri, 30 Apr 2010 14:16:14 +0200, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton)
or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming
from SIP/1004.
Read up on how Asterisk does
I have played with one before, it worked quite well. (Until somebody
fried it by accident).
Joachim
Peter wrote:
Hi,
I have one in stock - got it from a client who wanted to get rid of all
his old IT equipment.
Looks strange, did not have enough time to play with it Tried it
once,
Thanks. Yeah, that was the issue. I was requesting RLT and it wasn't turned
ON with the provider. Your mentioned solution fixed it.
-Bruce
On Fri, Apr 30, 2010 at 9:59 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
The disconnect is RECEIVED by Asterisk. So there is a problem with the
On Fri, Apr 30, 2010 at 5:26 AM, Peter peterp...@aboutsupport.com wrote:
I consider buying three GrandStream GXW4024 and connect 72 analogue
phones to asterisk
I recommend against that product. I have two that now sit on a shelf
due to bad call quality, echo issues, and random one way
I don't think you are actually hitting the time out. Comment out the
set timeout line I think the results will be the same. Which tells me
the timeout is not kicking in.
On 4/29/10, Brendan Sterne bren...@callvine.com wrote:
Greetings,
I'm trying to continue to do some processing after a
If you use zap then asterisk already does it. With sip the phones will
not tell asterisk about the hook flash. However you can play around
with dynamic features and assign a key that will mimic hook flash.
Injecting the beep sound might be hard though. Playing a different
ring to 2nd caller based
2010/4/30 Motiejus Jakštys desired@gmail.com
Hi,
please always add asterisk version to your query.
I am using Asterisk 1.4.17~dfsg-2ubuntu1.1 on an Ubuntu 8.04.4 server.
I managed to run internet radio (that streams MP3) within asterisk.
Minor change is nescesarry to make it work
Andrew Latham wrote:
Are you guys talking about the Asterisk Cookbook Because that
could be released in the next 20 years at this point...
The Asterisk Cookbook probably won't ever be released unless someone else wants
to step up and start it. We (as in the authors of Asterisk: TFoT) had
Hi,
I have a B400P BRI card with point-to-point connection (signalling:
bri_cpe) with this dmesg: http://pastebin.com/sXrRt1yM
When i restart asterisk server, the card cannot connect to the telco,
the control led flashes red. If I unplug the cable between the ISDN nt
and the card and
- Lyle Giese l...@lcrcomputer.net wrote:
Tim Nelson wrote:
Greetings all-
I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux.
It looks rather interesting. Has anyone used one? Where did you
purchase it? Pricing? Operational issues?
http://spidermux.com/
Tim
CF,
When I comment out the timeout the call continues as expected. I
believe the timeout is kicking in.
Can anyone point me to an example where TIMEOUT(absolute) is used as a
general timer, where the call continues after the expiry? I'm not
sure which extension to use T or t. I've tried
Hi!
This clears all my doubts, is not my configuration problem.
As I said, you could think about creating 4 different SIP gateways on the
Patton with 4 differing SIP ports. I don't know if the Patton will handle
4 gateways - but it might.
We have 4 trunk and 4 company in our office, I was
2010/4/30 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Hi!
This clears all my doubts, is not my configuration problem.
As I said, you could think about creating 4 different SIP gateways on the
Patton with 4 differing SIP ports. I don't know if the Patton will handle
4
Hi All,
I've been lurking here for a while now, having only made a couple of
posts. I am starting a new hardphone project and was wondering if there
is some GPL'ed IAX source that I could start with. I've searched and
haven't come up with much beyond iaxClient. While iaxClient does give
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Bryan Jacobs wrote:
I wonder if all the cell providers let you do this?
I presume you mean turn off voice mail. I don't know, but the first
time I called Verizon to have it done the gal I spoke with said it
couldn't be done. So I said thanks,
On Fri, 30 Apr 2010, Barry L. Kline wrote:
Bryan Jacobs wrote:
I wonder if all the cell providers let you do this?
I presume you mean turn off voice mail. I don't know, but the first
time I called Verizon to have it done the gal I spoke with said it
couldn't be done. So I said thanks,
http://downloads.asterisk.org/pub/telephony/libiax/
That package is outdated AFAIK but is a start. You should be able to use
chan_iax in Asterisk as a reference to fix libiax and use it for your own
purposes.
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive,
Before posting let me mention that this doesn't happen with ALL destination on
this provider.. some destination doesn't face this problem .. but this is a
sample call
[K -- Executing [0020100324...@a2billing:1]
[1;36;40mDeadAGI[0;37;40m([1;35;40mSIP/58169-ac47fda0[0;37;40m,
in the SIP/2.0 180 Ringing, the SDP shows:
a=sendonly
this is hold by rfc 3264. then when the other end picks up, a new SDP is
probably sent with
a=sendrecv
I believe your server is acting correctly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of
then why is it happening on a few destinations on that particular provider?
Date: Fri, 30 Apr 2010 13:09:05 -0700
From: david.wh...@watchguard.com
To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Strange
I don't know in your particular case, but if I call a PSTN endpoint via my
provider, the SIP signaling is different than if I'm calling a remote SIP
endpoint. This is because PSTN gateways have to make decisions (about codecs,
eg) independently of the remote endpoints.
In other words,
It is irrelevant who hangs up, you want to just use DeadAGI in the h
extension
I wish that would be the case, but at least on 1.4 I see:
[Apr 30 14:59:38] -- Executing [...@master-route:1] DeadAGI(...) in new
stack
[Apr 30 14:59:38] WARNING[27845]: res_agi.c:2160 deadagi_exec: Running
Thanks a lot Kevin for the reply
From: Kevin P. Fleming kpflem...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, April 29, 2010 5:43:15 AM
Subject: Re: [asterisk-users] No change in payload.
44 matches
Mail list logo
