Hello,
is it possible to set an IP address for RTP different than the one used for
SIP?
I want to use asterisk behind a sip proxy (opensips), but I was thinking if
I could avoid having to run rtpproxy on the sip proxy server and let
asterisk itself take care of it. So that:
Asterisk SIP address
Paul,
I have kind of a related question.
asterisk-1.8.4-summary.txt does not always properly link specific
patches to issues. For example, revision 307509 is associated with issue
18542, and it is not reflected in the summary. There may be more like this.
I tried to report this inconsistency
On 5/11/2011 10:15 AM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of isr...@gmail.com
Sent: Wednesday, May 11, 2011 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial
[This is my last post in this thread - as I really CBA anymore!]
Wow! You really don't see it do you? Fair enough. I thought you were
just playing along with my 'ego baiting' game - but it seems I hit the
mother load of all ego's here.
Apologies to all 'watchers' - but this was intended as a
Dear all,
Could you teach me minimum required Asterisk modules, application and etc to
install PSTN GW+SIP Client functionalities in a PC as shown below? I have
already downloaded astersik-1.8.3.3 and dahdi-linux-complete-2.4.1.2+2.4.1 in
the PC.
Analog telephone --- PSTN GW + SIP Client ---
This should be interesting, a double header Friday at 12 Noon EDT,
session 2 at 1PM EDT.
1) Pascal Doré, Media5corp. Pascal will talk about what they've been
up to in the year since his last visit. Thanks to the Asterisk mailing
list and VoIP community, their Media5fone was able to fix its g722
But it is all true. Don't rage on me if you are upset by my
accomplishments.
It was to establish the conditions I work under and the fact that I can and
do solve almost all of my setbacks myself in very disparate setups, from
VSAT, poorest countries in the world, war, explosions, getting shot
This is beginning to turn even more unpleasant than the original breach of
netiquette which prompted the discussion -- like one of those fights which
starts with a raised voice, escalates to fisticuffs and then weapons, and by
the time innocent bystanders are getting injured nobody can even
On Thu, May 12, 2011 at 4:28 AM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
This is beginning to turn even more unpleasant than the original breach of
netiquette which prompted the discussion -- like one of those fights which
starts with a raised voice, escalates to fisticuffs and then
PS 42 is the answer, now what is the quesstion. :)
What is the difference between a bird?
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Hi list,
Is there any way by which we can put multiple calls into hold with asterisk.
like A to B.
then C to B and A on hold.
then D to B now C ,A on hold like wise..
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
--
On Thu, 12 May 2011, virendra bhati wrote:
Hi list,
Is there any way by which we can put multiple calls into hold with asterisk.
like A to B.
then C to B and A on hold.
then D to B now C ,A on hold like wise..
It's more a phone issue that asterisk. Just get a multi-line phone
(GXP2000,
Hi David,
When I was testing 1.6.1 for high volume channels, I couldn't get over 1000
channels / 40 CPS without the load average spiking up due to io wait. I
switched back to 1.4 and I can go to 3000 channels / 75 CPS with no io wait
and a load average in the 1s. It seemed like it was caused by
Hi ,
I am using Cisco 7940/60 phone.
Is this okay or we need another phone for that. plz suggest me '
On Thu, May 12, 2011 at 3:52 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Thu, 12 May 2011, virendra bhati wrote:
Hi list,
Is there any way by which we can put multiple
In asterisk CLI do pri show spans. The fact the card is in RED alert
means the hardware does not see the pri line connected to the card.
I probably made a mistake in copying / pasting. pri show spans was showing
something like :
PRI span 1/0: Provisioned, Up, Active
Calls can enter, I see
On Thu, 12 May 2011, virendra bhati wrote:
On Thu, May 12, 2011 at 3:52 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Thu, 12 May 2011, virendra bhati wrote:
Hi list,
Is there any way by which we can put multiple calls into hold with
asterisk.
like A to B.
then C to B and A
Hi,
this is my first post to mailing list, so sorry in case i'm doing something
wrong.
when i want to count concurent calls from particular user, i dont use any
cron jobs or counters
in dialplan, run query on cdr, something like:
SEELCT dst, calldate, IF(action = 'substract', @count := @count -
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Nicolas Ross
Sent: Thursday, May 12, 2011 6:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trying out a
Jared,
Thank you for that information!
Has anyone else had an experience like this?
On 12 May 2011 20:25, Jared Geiger compuw...@gmail.com wrote:
Hi David,
When I was testing 1.6.1 for high volume channels, I couldn't get over 1000
channels / 40 CPS without the load average spiking up
Show us the output of a failed call with pri debug enabled on that span.
It will be difficult, since the PRI is in use on our old asterisk box.
I will have to get to the colo at night, to avoid disrupting calls
during the day.
Is there any other thing that I should collect ?
--
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mayamatakeshi
Sent: Thursday, May 12, 2011 12:58 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Different IP addresss for SIP and RTP
Hello,
is it possible
Hi!
Here's a user with mobile phone - however why does it treat this as ERROR ?
I have a log full of that ---
-- Registered SIP '0010106' at 212.93.100.181:3698
[2011-05-12 16:07:57] NOTICE[30258]: chan_sip.c:19679
handle_response_peerpoke: Peer '0010106' is now Reachable. (212ms /
On 11-05-11 06:36 PM, Skyler wrote:
Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd
like to take a look at it for sure. The dial plan example Leif replied with
is pretty much what I was thinking, just didn't have a clue how to go about
it. ;)
You could also look
On 11-05-11 09:31 PM, Jose P. Espinal wrote:
Download links on the website have not been updated (asterisk.org)
Oops sorry! I will fix that right.. now!
Leif.
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Hey Guys!
I am looking ConfBridge for 1.8 version of asterisk. How could i obtain and
install with 1.8 ?
-S
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Holly Cow! Its there already sorry i thought it will only comes with 1.10. We
are using meetme since last 5 year do you think confbridge is better then
meetme ? just need your suggestion
/usr/lib/asterisk/modules/app_confbridge.so
From: satish...@hotmail.com
To:
On 05/12/2011 09:37 AM, satish patel wrote:
Holly Cow! Its there already sorry i thought it will only comes with
1.10. We are using meetme since last 5 year do you think confbridge is
better then meetme ? just need your suggestion
/usr/lib/asterisk/modules/app_confbridge.so
The app_confbridge
Thanks Kevin,
Good to know. Different mean features vise or performance ? Do you think it is
a good idea to replace meetme with confbridge in current 1.8 or i should wait
for 1.10 ?
-S
Date: Thu, 12 May 2011 09:50:12 -0500
From: kpflem...@digium.com
To: asterisk-users@lists.digium.com
Hello Everyone,
I wonder if someone could share a manual about using SIPp for Asterisk's
testing.
I'll be gratefull
Regards,
Elder Arohuanca
Lima - Peru
On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe zac.wo...@gmail.com wrote:
Sipp looks pretty good! I don't know how I missed this one. This
Hello,
is there some way to make Asterisk light up a certain light on an IP-phone ?
Like MWI, the message waiting indicator can light up if there is voicemail.
Could this light, or even other lights (like BLF-buttons) be used to
give a visual notification to the user ?
For example : if a
Hi all,
A week or so down the list, i read that not many people were using
realtime on an Asterisk18, so i had this afternoon a go at it...
[sorry for the inconveneant line-wraps]
First i did:
mysql create database asterisk;
mysql grant all on asterisk.* to 'voipadmin'@'localhost' identified by
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, May 12, 2011 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Light indicator managed by Asterisk
On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Hello,
is there some way to make Asterisk light up a certain light on an IP-phone ?
Like MWI, the message waiting indicator can light up if there is voicemail.
Could this light, or even other lights (like
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote:
Hello,
is there some way to make Asterisk light up a certain light on an
IP-phone ?
Like MWI, the message waiting indicator can light up if there is
voicemail.
Could this light, or even other lights (like BLF-buttons) be used to
On Thu, May 12, 2011 at 10:07 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *mayamatakeshi
*Sent:* Thursday, May 12, 2011 12:58 AM
*To:*
On 05/12/2011 06:58 PM, Andrew Latham wrote:
On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Hello,
is there some way to make Asterisk light up a certain light on an IP-phone ?
Like MWI, the message waiting indicator can light up if there is voicemail.
Many thanks to all that replied. I'm going to test out the
suggestions/scenarios and I'll post back with what worked for me.
S.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Thursday, May 12, 2011 6:29 AM
To:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, May 12, 2011 12:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light
On 05/12/2011 07:12 PM, Carlos Chavez wrote:
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote:
Hello,
is there some way to make Asterisk light up a certain light on an
IP-phone ?
Like MWI, the message waiting indicator can light up if there is
voicemail.
Could this light, or even
On 05/12/2011 07:24 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, May 12, 2011 12:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, May 12, 2011 12:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, May 12, 2011 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jonas Kellens
Sent: Thursday, May 12, 2011 1:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jonas Kellens
Sent: Thursday, May 12, 2011 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light
Hi
I've spent two days trying to solve this issue but to no prevail and I'm
hoping to get some help.
I've configured Asterisk as a SIP client, running on OpenWRT on an embedded
device with onboard FXS and ATA. Asterisk is connecting to an external SIP
provider on the Internet who in turn
Eric Wieling wrote:
pbx*CLI core show application minivmmwi
Core show application minivmmwi
core show function DEVICE_STATE
Both of these must be a 1.6.x or newer, I have neither under 1.4
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Check out
http://kb.smartvox.co.uk/index.php/asterisk/sip-extensions/shared-voicemail-part2/
Date: Thu, 12 May 2011 14:38:46 -0400
From: supp...@drdos.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Light indicator managed by Asterisk
Eric Wieling wrote:
pbx*CLI
Correct.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Doug Lytle
Sent: Thursday, May 12, 2011 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light
Hi all,
I would like to know what are core modules that are used for
asterisk?
can anyone help me regarding this...
with regards,
viswavardhan
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You can, using device_state function (I use asterisk 1.6.2.X)
Here is a example for a conference... when sombody enter to conference a
light up on my aastra phone:
exten = s,1,Set(DEVICE_STATE(Custom:confer)=INUSE)
exten = s,n,Meetme(5000)
exten = s,n,Hangup
exten =
Hi,
look if you have res_config_mysql.so module instaled on your asterisk.
On CentOS /usr/lib/asterisk/modules
Regards
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dan, elder,
I have played with scripts to generate calls and track their
completion, email me off-list if you have questions.
daveC
Daniel - Asterisk wrote:
Hello Everyone,
I wonder if someone could share a manual about using SIPp for
Asterisk's testing.
I'll be gratefull
Regards,
Hi all
Usually I build asterisk from source, but recently have been doing a
couple of test installations with packages from the Digium repository.
About how long does it take to get from new release announcement into the
Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the
rpm
On 05/12/2011 02:40 PM, Cassius Smith wrote:
Hi all
Usually I build asterisk from source, but recently have been doing a
couple of test installations with packages from the Digium repository.
About how long does it take to get from new release announcement into the
Digium RPM repository?
On 05/12/2011 07:12 PM, Carlos Chavez wrote:
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote:
Hello,
is there some way to make Asterisk light up a certain light on an
IP-phone ?
Like MWI, the message waiting indicator can light up if there is
voicemail.
Could this light, or even
How to reload only agents.conf ?
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New to Asterisk? Join us for a live introductory webinar every Thurs:
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Thursday, May 12, 2011 3:28 PM
To: asterisk-users
Subject: [asterisk-users] how to reload agents.conf ?
How to reload only agents.conf ?
[Danny
Guys!
I am running 1.8 on production we have one PRI and 50 extensions. since last
few days its working fine but today some how server load get high 194 % CPU and
when i did asterisk -r i got CLI but no out put for any command. I check logs
and nothing interesting there.. I am not using any
On 12/05/11 9:31 PM, Steve Totaro wrote:
PS 42 is the answer, now what is the question. :)
Heh, that might be one example where top posting would make sense ;-)
--
Cheers,
Matt Riddell
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
hi:
I report my issue as issue 19628.
it is fixed and I run asterisk 1.8 in production now.
thanks a lot for your help!
Regards,
tbskyd
2011/5/11 d tbsky tbs...@gmail.com:
hi:
ok I will create a bug report. and I found I still need
prematuremedia=no in asterisk 1.6.2.18.
hi:
sorry. the issue number is 19268. not 19628.
sorry about that!!
Regards,
tbskyd
2011/5/13 d tbsky tbs...@gmail.com:
hi:
I report my issue as issue 19628.
it is fixed and I run asterisk 1.8 in production now.
thanks a lot for your help!
Regards,
tbskyd
2011/5/11 d
Hello Folks,
What could be producing the following warnings on console, after an
installation from source (Asterisk 1.4.41):
[May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module 'res_musiconhold.so':
/usr/lib/asterisk/modules/res_musiconhold.so:
somewhere along the way, i noticed incoming calls from google voice are
no longer working on my asterisk 1.8.3.2 system.
When the call comes in, asterisk immediately prints on the console:
== Spawn extension (google-in, s, 2) exited non-zero on
'Gtalk/+12153930924-f947'
[May 12 22:47:18]
On 5/12/2011 11:08 PM, Jeremy Kister wrote:
[May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote
peer reported an error, trying to establish the call anyway
I found the problem, and I am sending in a bug report :)
if anyone is interested, the issue is 19286 (i'll be
Hello,
After installing Asterisk from source in Slackware 13.1, I get the
following error:
Error loading module 'res_config_odbc.so':
/usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol:
ast_odbc_clear_cache
Then a core dump.
If I change the /etc/asterisk/modules.conf in order
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