On 10/13/2010 12:09 AM, Paul Belanger wrote:
On Tue, Oct 12, 2010 at 8:08 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
[Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto
UDP socket destined for public_ip:2049
Is something failing, or is this just
Define aa confrence room num and Syntex is like...
Macro(conference-enter,${EXTEN})
On Mon, Oct 11, 2010 at 6:14 PM, Daniel Knoll dan...@danielknoll.de
wrote:
Hey,
i forgot to ask, how can i get the user number from a caller he is in a
conference, i don't find a variable to us this
Hi,
I have simulated “Chan phone” driver according to my own driver code and I
am able to make internal and external [trunk] Asterisk calls.
Only issue I am facing is with hangup in ringing state of incoming call.
(1) Make a call from external X-lite to FXS and FXS is in ringing state
now
On Tuesday, October 12, 2010 05:31:46 pm Tilghman Lesher wrote:
On Tuesday 12 October 2010 08:51:15 Oguzhan Kayhan wrote:
Hello,
I am using 1.6.2.9-1+b1 asterisk.with cdr_mysql.
Everything seems workging correctly except cdr logs.
It fills up all data when a call established except src
Hello,
I have a default installation of asterisk 1.6.1.9-2
When i create a user in users.conf via asterisk-gui,
calls, voicemail etc works.
But if i create a user realtime (and my realtime caching is available too)
i can see the realtime user with sip show peers.
But, my local dial rules does not
Check sip_buddies table for the correct context entry.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-13 5:31 AM, Oguzhan Kayhan oguzh...@bilkent.edu.tr wrote:
Hello,
I have a default installation of asterisk 1.6.1.9-2
When i create a user in users.conf via asterisk-gui,
calls, voicemail
Hi,
Which DTMF mode do people mostly use?
I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for
feature usage), the tones arent repeated to the end user.
So if I call a company that has a menu system, I can't use the menu.
Thanks
Dan
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Deneen
Sent: Tuesday, October 12, 2010 9:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sound file debug
On
It depends upon whether you are receiving DTMF or sending, and whether you
are using a VoIP protocol or using DAHDI/Zaptel.
Could you explain a bit what type of setup you have?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-13 9:15 AM, Dan Journo d...@keshercommunications.com wrote:
Hi,
It depends upon whether you are receiving DTMF or sending, and whether you
are using a VoIP protocol or using DAHDI/Zaptel.
Sorry about the lack of info.
It's a simple SIP only setup. A handful of sip phones, an asterisk server, and
a sip provider.
The DTMF signals from the sip phones are
Then that likely means your phone have the correct dtmfmode, but the link
between you and the provider doesn't.
Make sure both you and the provider are using the same dtmfmode. My
experience shows that sometimes it's also between your provider and THEIR
provider, and sometimes reporting the
I just tried this:-
[test_calls]
exten = 555,1,Answer()
exten = 555,n,SendDTMF(12345)
exten = 555,n,Playback(beep)
I dialed 555 on the sip phone, nothing was heard, and then a beep...
It seems that Asterisk isn't sending DTMF. Its only able to receive.
Thanks
Dan
--
Then that likely means your phone have the correct dtmfmode, but the link
between you and the provider doesn't.
Just carried out another test to see if my provider was working properly:-
exten = INCOMINGDDI,1,Wait(1)
exten = INCOMINGDDI,n,Answer()
exten = INCOMINGDDI,n,SendDTMF(12345)
If I
Can you send us the SIP config of the sip provider (in sip.conf), removing
appropriate passwords and static IPs of course.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, October 13, 2010 10:22 AM
Can you send us the SIP config of the sip provider (in sip.conf), removing
appropriate passwords and static IPs of course.
[provider]
type=friend
host=removed
username=removed
fromuser=removed
secret=password
context=incoming_calls
dtmfmode=rfc2833 also tried auto.
disallow=all
allow=gsm
* The provider has confirmed that they support rfc2833 or inband with
the right codecs.
Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?)
--
_
-- Bandwidth and Colocation Provided by
Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?)
This is from the sip.conf for the provider:
allow=gsm
allow=ulaw
This is from the sip extension:-
alaw,ulaw,gsm
--
_
-- Bandwidth and Colocation Provided
I would suggest first to make sure that asterisk is receiving DTMF fine from
your IP devices/phones. Do you have a test IVR where you can dial and press
digits and verify that asterisk is responding?
Once you are sure that asterisk is receiving DTMF fine, then you should ask
your provider what
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, October 13, 2010 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DMTF Mode
Are you using the
Anyone used the new Grandstream GXP-21XX series phones. We have been
testing these phones and like what we see. We are looking for a greater
cross section of testing before we roll them to production. Any feed back
would be appreciated. We are talking with Grandstream engineering and they
are
I would suggest first to make sure that asterisk is receiving DTMF fine from
your IP devices/phones. Do you have a test IVR where you can dial and press
digits and verify that asterisk is responding?
Made a quick IVR, and its working for both sides of the asterisk (between the
provider and
We are testing the innomedia ATA's to possibly replace our current line up
of ATA's that we are using. Has anyone used their product? What is their
track record on stability, voice quality, DTMF talkoff, T.38
Thanks
Bryant
From: Zeeshan Zakaria
Based on this, your call is probably getting to the provider as ulaw (the
alaw is thrown out since it isn't in both selections; if you are in U.S. you
don't need the alaw). Try the call with higher debug (at least 5) and verify
which one is being selected.
debug 5 doesnt give me any info
On Wed, Oct 13, 2010 at 10:12 AM, Dan Journo
d...@keshercommunications.com wrote:
How can I tell if Asterisk is sending the tones through to the provider? I
need to find out whether its something I'm doing, or something the provider
is doing.
You need to enable DTMF logging (logger.conf) and
How can I tell if Asterisk is sending the tones through to the provider?
You need to enable DTMF logging (logger.conf) and debug an incoming /
outgoing call.
Can you understand this? I can see the DTMF signals coming in. I pressed 5 on
the normal phone line, and then I pressed 8 on the sip
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, October 13, 2010 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DMTF Mode
Based on this, your call is
Could features.conf be preventing asterisk from repeating the DTMF tones?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Typically Grandstream 21XX and 20XX is all we've deployed in the past and
have had great success with them. I occasionally ( and I mean rarely ) get
complaints about calls when on speaker phone, but I think thats more user
error than anything else, i've been using them for a couple years now and
I'm on 1.4.30 and this is what I get using debug 5
-- Accepting AUTHENTICATED call from 192.168.xx.xx:
requested format = ulaw,
requested prefs = (ulaw|gsm|alaw),
actual format = gsm,
host prefs = (slin|gsm|ulaw|alaw),
priority = mine
Strange. I dont get
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, October 13, 2010 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DMTF Mode
Could
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, October 13, 2010 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DMTF Mode
I'm on 1.4.30 and
What is your featuredigittimeout value?
Not used. So default 1000ms.
I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw) and
its started working in a fashion.
The DTMF tones keep getting stuck. I press a number on the sip phone, and the
other party hears a tone. But
I have followed http://www.asterisk.org/AsteriskNOW-1.5-QuickStart
I have installed asterisk in virtualBox for now.
I am able to login in to console.
Now if I want to create a simple PBX in my local network.
like
I have 5 machine in my network.
I am thinking of assigning each a soft phone.and
I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw)
and its started working in a fashion.
The DTMF tones keep getting stuck. I press a number on the sip phone, and
the other party hears a tone. But every few tones, it gets stuck and they
hear a long tone of about 3
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, October 13, 2010 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DMTF Mode
What is
Dan Journo d...@keshercommunications.com wrote:
Based on this, your call is probably getting to the provider as ulaw (the
alaw is thrown out since it isn't in both selections; if you are in U.S.
you don't need the alaw). Try the call with higher debug (at least 5) and
verify which one
Since alaw is sort of making it work, either your SIP provider or your
other party is not in U.S.
They arent in the US. Everything is in the UK.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Since alaw is sort of making it work, either your SIP provider or your
other party is not in U.S.
I dont understand why the codec should make a difference if im using rfc2833.
Could you clear that up for me?
--
_
-- Bandwidth
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, October 13, 2010 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DMTF Mode
Since
From what I read, the codec could be trying to switch from rfc2833 to inband
during the call, causing the stuck effect.
Any way to prevent that?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, October 13, 2010 10:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DMTF Mode
From
On Wed, 13 Oct 2010, Bryant Zimmerman wrote:
Anyone used the new Grandstream GXP-21XX series phones. We have been
testing these phones and like what we see. We are looking for a greater
cross section of testing before we roll them to production. Any feed back
would be appreciated. We are
According to the WIKI, changing rfc2833 to auto in sip.conf should do the
trick.
Didnt help. I'm contacting the provider to see if they have any ideas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Thanks to everyone who helped me on this.
Hopefully the provider can sort out the sticking tones now.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Gordon
Thanks for the reply. Grandstream has three new phones that will replace
the GXP-20XX series as some point. GXP-2000 - GXP-2100, GXP-2010 -
GXP-2110, GXP-2020 - GXP-2120. The GXP-2110 has been released the others
appear to be on the cusp of release. We have been testing the GXP-2110
On Wed, Oct 13, 2010 at 11:26 AM, Jigar Joshi jiga...@gmail.com wrote:
How to proceed?
I am very very newbie to asterisk.
pabelanger ~book
infobot [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN
0-596-51048-9) --- Order yours at
http://www.oreilly.com/catalog/9780596510480/ --- Free
Thanks Paul ,
I want some quick reference tutorials.
On Wed, Oct 13, 2010 at 9:58 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Wed, Oct 13, 2010 at 11:26 AM, Jigar Joshi jiga...@gmail.com wrote:
How to proceed?
I am very very newbie to asterisk.
pabelanger ~book
infobot
Hello Asterisk Community,
Is there a way to check in asterisk cdrs and extension forwarded?
I mean, i'm calling to a ISDN number, wich goes to extension 8222, but
this extension is forwarded to another one, the problem is that in
CDRs i am able to see the the first step of the call, but never
Hi
Check this out
http://spidermux.com/
On Mon, Oct 11, 2010 at 8:18 PM, Karim Davoodi karimdavo...@gmail.comwrote:
Hello,
I want to create channel bank in this case:
channel bank
|-|
|
ANI and CID are same in SIP some people use P-Asserted-Identity header to
send ANI , but that is not a standard specification just a workaround.
--
Thanks Regards,
Godson Gera
IVR FreeSWITCH Radius India http://godson.in/
On Tue, Oct 12, 2010 at 5:07 AM, JR Richardson
Hi,
I have an asterisk server sitting behind a pfsense firewall, I have
successfully configured pfsense for NAT traversal, and clients from the
internet can call clients inside the network of asterisk, as well as
other clients registered with this asterisk server on the internet.
The problem
Hi,
I am trying to set up two bords on my server: TDM410p(This on is ok) and
TE110p.
This is my system.conf
# Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER)
fxsks=1,2,3,4
# Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Sent: Wednesday, October 13, 2010 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] checking CDR
Hello
Hi,
(Following is for asterisk 1.4)
For the forwarded calls, you should see two entries in the cdr, and this is
because a forwarded call is actually two separate calls. You have to look in
the channel and dstchannel fields of the cdr to match the call ids of the
calls to figure out which calls
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Sent: Wednesday, October 13, 2010 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] checking CDR
Hello
The real question is are you having the phone forward the calls or is your
dial plan redirecting to outbound calling?
Bryant
From: Zeeshan Zakaria zisha...@gmail.com
Sent: Wednesday, October 13, 2010 2:16 PM
To: Asterisk Users Mailing List -
Am 13.10.2010 19:50, schrieb Ahmed Ossama:
Hi,
I have an asterisk server sitting behind a pfsense firewall, I have
successfully configured pfsense for NAT traversal, and clients from the
internet can call clients inside the network of asterisk, as well as
other clients registered with
Hi list,
This is not necessarily an asterisk issue, but a lot of you guys know
way more then me, so I have a question:
someone at my company sets his phone to forward calls to his cellphone,
so someone calls our office, call is forwarded to his cell, and the
callerID that shows up on his cell
On Wed, 13 Oct 2010, Bryant Zimmerman wrote:
Gordon
Thanks for the reply. Grandstream has three new phones that will replace
the GXP-20XX series as some point. GXP-2000 - GXP-2100, GXP-2010 -
GXP-2110, GXP-2020 - GXP-2120. The GXP-2110 has been released the
others appear to be on the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Wednesday, October 13, 2010 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call forwarding callerID
Hi
On Wed, 13 Oct 2010, Gerard wrote:
This is not necessarily an asterisk issue, but a lot of you guys know
way more then me, so I have a question: someone at my company sets his
phone to forward calls to his cellphone, so someone calls our office,
call is forwarded to his cell, and the
Since alaw is sort of making it work, either your SIP provider or your
other party is not in U.S.
Its stopped working again. This is really unusual. I didnt change anything.
I decided to do a tcpdump, and I can clearly see the rfc2833 packets being
exchanged correctly.
Why should both parties
Hi,
I have some problem with my provider. While the sip registration is
successful, i intermittently encounter problem in dialing out. I receive 603
Declined error in my Sjphone client. The asterisk log shows line is
busy/congestion.
Appreciate if help or direction can be provided.
Thanks.
CK
On Wed, Oct 13, 2010 at 6:48 PM, asterisk asterisk aster...@ck-lee.com wrote:
Appreciate if help or direction can be provided.
21.6.2 603 Decline
The callee's machine was successfully contacted but the user
explicitly does not wish to or cannot participate. The response MAY
indicate a
Hi all,I'm planning a new Asterisk installation; the users want to duplicate the paging function they have with their current Panasonic hybrid system. They dial *3 and announce a held call on line 3, for example, and the announcements comes out of all the desktop phone speakers. I'm planning to
Hi Cassius,
Can`t help for SPA-942, but the Wiki had good info on the Polycoms. Use the
Wiki and you`ll do good. Two warnings:
1) It seems to me that the adhoc MeetMe room used by the page application
slows things down quite a lot. If you page and have a phone nearby, you`ll
hear
Hey all, sorry if this has been covered, but I've not found anything after a
couple hours' worth of googling. I can see (and I'm familiar with) all the
usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find any
reference to MySQL and the new CEL logging tool other than ODBC. Is
Thanks Mike - this does help. The setup will be a local server on the LAN, and hopefully have plenty of snort to handle the load (20-30 phones). I also am not quite ready to put out 1.8 for my users yet.Do you have a snippet of dialplan code you'd be willing to share to loop through a group and
On Wed, Oct 13, 2010 at 9:31 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Hey all, sorry if this has been covered, but I've not found anything after a
couple hours' worth of googling. I can see (and I'm familiar with) all the
usual MySQL addon apps once I install Asterisk 1.8.x, but I
On Wed, Oct 13, 2010 at 9:53 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Wed, Oct 13, 2010 at 9:31 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Hey all, sorry if this has been covered, but I've not found anything
after a
couple hours' worth of googling. I can see
On Wed, Oct 13, 2010 at 9:52 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Wed, Oct 13, 2010 at 9:53 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Wed, Oct 13, 2010 at 9:31 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Hey all, sorry if this has been
My SIP registration are name sort of like this : phonea-exten1, phone1-exten2,
etc. Makes it easy to loop, I can send you a snippet tomorrow. But you have to
know in advance all the SIP peer names.
Mike
From: asterisk-users-boun...@lists.digium.com
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