Hi there
This has happened to me before when I changed the tone duration, it was
too long and the PSTN receiver no longer understood the tones, but it
seems unlikely nothing has changed.
The cli or logs should show you whats happening, something will be
blocking the call, either the group
Hi.
I have 3 lines connected to a SangomaA200. All was working normally,
but 3 days ago we experiment this issue.The outside callers can
call us normally; but when we try to do an external call, just receive
the message all circuits are busy.
Only happen with outgoing call. Incomming
And now they've gone back and reactivated that protocol change, breaking
chan_gtalk again. Applying the small patch from
https://issues.asterisk.org/jira/browse/ASTERISK-18301 got things
operational again. Let's see how long this one goes before they go back to
the old protocol version.
--
On 11-08-21 10:36 AM, Paul Belanger wrote:
On 11-08-21 02:54 AM, Jeremy Kister wrote:
On 8/20/2011 12:46 PM, Paul Belanger wrote:
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
confirmed on asterisk 1.8.6.0-rc1
pre-patch behavior: ring-no-answer
On Mon, Aug 22, 2011 at 1:42 PM, Paul Belanger pabelan...@digium.comwrote:
I've just reverted this patch, it seems google is still making changes to
the protocol.
Yep, looks like they also reverted their protocol change. Going back to
pre-patch chan_gtalk.c makes things work again.
Wish
On 8/20/2011 12:46 PM, Paul Belanger wrote:
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
confirmed on asterisk 1.8.6.0-rc1
pre-patch behavior: ring-no-answer
post-patch behavior: expected
--
Jeremy Kister
http://jeremy.kister.net./
--
On 11-08-21 02:54 AM, Jeremy Kister wrote:
On 8/20/2011 12:46 PM, Paul Belanger wrote:
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
confirmed on asterisk 1.8.6.0-rc1
pre-patch behavior: ring-no-answer
post-patch behavior: expected
Thanks
Fixed endless ringing on outgoing gtalk calls for me. Asterisk 1.8.5.0.
-Vladimir
On 8/20/2011 11:46 AM, Paul Belanger wrote:
Afternoon,
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
I've seen a few reports around the web and like to get more
We're a VoIP provider essentially competing with our local incumbent
Telco, and a sizeable number of our customers use satellite internet.
As a result, these customers never have ping times less than 500ms,
and are often exceeding 2500ms.
I manually apply a patch to the Asterisk source
Have you tried SIP session timer values in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Tuesday, June 28, 2011 9:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Tuesday, June 28, 2011 9:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Outgoing calls get dropped on high-latency
connections.
We're a VoIP provider essentially competing with our
Nicholas,
Sorry I don't know, but are your calls working okay ?
Depending on the verbosity level being set, I see warning
msgs all the time, that I ignore.
Frequently, an upgrade to the next release of the same
major version also eliminates the warning msgs.
If you are really concerned,
Asterisk 1.4.29 or so.
access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any range
1 2
access-list _dmz_acl extended permit udp 10.129.42.0 255.255.255.0 any eq
5060
But yes, all your feedback worked. I didn't need to port-forward any
incoming ports, only
I'm trying to move my Asterisk deployments under a Virtual IP address and
now remember why I dislike this. My primary Asterisk system is now behind a
firewall in private address space. My question is what ports are needed to
be opened just for the purpose of placing outgoing calls. I would have
On Sun, Jan 3, 2010 at 9:14 PM, Nicholas Blasgen
nicho...@refractivedialer.com wrote:
I'm trying to move my Asterisk deployments under a Virtual IP address and
now remember why I dislike this. My primary Asterisk system is now behind a
firewall in private address space. My question is what
Nicholas,
you haven't specified which version, which does make
a lot of difference.
1.6.x can easily traverse NAT. If you are only making
outbound calls, you shouldn't need to forward 5060.
Unless you have a special NAT that is blocking
outbound connections, the SIP.conf settings
Hello! I am looking for a configuration sample to authenticate outgoing
calls. The idea is that each user have a password to dial any number. I was
reading about Asterisk cmd Authenticate, Disa, etc. But I dont know how use
this tools when I have running freepbx. Thanks for any idea.
Gustavo
Gustavo A Gonzalez wrote:
Hello! I am looking for a configuration sample to authenticate
outgoing calls. The idea is that each user have a password to dial any
number. I was reading about Asterisk cmd Authenticate, Disa, etc. But
I don’t know how use this tools when I have running
exten
Hello list,
How could I limit the outgoing calls for one trunks easily?
Thanks
VoipCrazy
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Register Now:
try put calls into groups using GROUP() function and check call limit
with GROUP_COUNT()
voip crazy wrote:
Hello list,
How could I limit the outgoing calls for one trunks easily?
Thanks
VoipCrazy
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Hi all,
I have a problem with my asterisk box and an X100P FXO card. I am able to
place outgoing calls from my SIP phone (Cisco 7940) to any external number
using my PSTN line, but when I call my PSTN line from my cell phone, the
Cisco doesn't ring (and no message appears in the Asterisk CLI).
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put sip debug peer 101 I have this lines:
*CLI sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI
-- SIP read from 10.0.0.9:5060:
INVITE
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put sip debug peer 101 I have this lines:
*CLI sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI
-- SIP read from 10.0.0.9:5060:
INVITE
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put sip debug peer 101 I have this lines:
*CLI sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI
-- SIP read from 10.0.0.9:5060:
INVITE
there as well.
Regards,
Roelof Dijkstra
Network Engineer EMEA
Compuware Europe BV
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Josu Lazkano Lete
Sent: Tuesday, May 08, 2007 1:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] outgoing calls
thank you very much!
it works
- Original Message -
From: Dijkstra, Roelof
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, May 08, 2007 1:53 PM
Subject: RE: [asterisk-users] outgoing calls
Hello Josu,
In you're sip.conf you have the 2
Can you put a protocol analyzer on your E1 and send us the layer 3 trace?
John Treble
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hello.
I have a Digium TE110P in my server asterisk, have connected a PRI of the
PSTN. the incoming calls work correctly, but when attempt to make calls
outwards does not work and it leaves an error to me like the following one:
*-- Channel 0/1, span 1 got hangup request
Apr 11 22:43:45
Hi list,
I've been trying all kinds of things for hours but I keep ending up
with nothing, so I was hoping to get some help.
Because I could not get it to work i'v completely reset to the default
configuration, except for sip.conf
If I call my number I get the DEMO talking to me so I know
I currently have Asterisk 1.2.7.1 and the Sangoma A200 w/ 6 FXO ports and HW Echo canceller. I have outgoing calling setup to use a group so that if one channel is busy it goes to one of the other channels. What's weird is that when I dial an outside number, sometimes it goes through and other
Some central offices cannot immediately accept digits. Try preceding
your dial string with a 'w'.
Andrew Berman wrote:
I currently have Asterisk 1.2.7.1 http://1.2.7.1 and the Sangoma A200
w/ 6 FXO ports and HW Echo canceller. I have outgoing calling setup to
use a group so that if one
Adding the w seems to work. Thanks Michael.--AndrewOn 5/15/06, Michael Welter [EMAIL PROTECTED]
wrote:Some central offices cannot immediately accept digits.Try preceding
your dial string with a 'w'.Andrew Berman wrote: I currently have Asterisk 1.2.7.1 http://1.2.7.1 and the Sangoma A200 w/ 6 FXO
On Mon, 2006-03-13 at 21:13 +0100, Christoph Eicke wrote:
On Monday 13 March 2006 20:47, Dave Hope wrote:
Hello all,
With some help from people in #asterisk on freenode, I've managed to get
incoming SIP calls working.
Outgoing calls however are however a different matter. My whole
On Monday 13 March 2006 20:47, Dave Hope wrote:
Hello all,
With some help from people in #asterisk on freenode, I've managed to get
incoming SIP calls working.
Outgoing calls however are however a different matter. My whole working
(incoming calls only) SIPgate configuration can be found
Hello all,
With some help from people in #asterisk on freenode, I've managed to get
incoming SIP calls working.
Outgoing calls however are however a different matter. My whole working
(incoming calls only) SIPgate configuration can be found here. [1]
When I uncommon what's in there, nothing
On Sun, 2005-12-11 at 16:18 +0200, Warren Burstein wrote:
I'm running asterisk 1.0.9 with TDM400B's for both internal and external
lines.
I put in the macro that dials outside lines an AbsoluteTimeout(36000),
never expecting it to happen. But it does, a few times a month.
I've noticed
I'm running asterisk 1.0.9 with TDM400B's for both internal and external
lines.
I put in the macro that dials outside lines an AbsoluteTimeout(36000),
never expecting it to happen. But it does, a few times a month.
I've noticed an odd thing, it seems that it usually happens twice in a
row
Warren Burstein ha scritto:
What is frustrating is that the cdr file shows the dst as T rather
than as the phone number dialed. I realize that AbsoluteTimout causes
it to jump to the T extension, but it would help to know who the user
dialed (asking a week later isn't going to get any
Simone Cittadini wrote:
Warren Burstein ha scritto:
What is frustrating is that the cdr file shows the dst as T rather
than as the phone number dialed. I realize that AbsoluteTimout
causes it to jump to the T extension, but it would help to know who
the user dialed (asking a week later
I am trying to route my calls through an outside IAX provider. I
am having a problem with which codec to use. The only way I have
successfully been able to make an outgoing call is if i do:
disallow=all
allow=g729
in the sip.conf file (for my phones) and the iax.conf file. The second I add
On Nov 23, 2005, at 11:14 AM, Michael wrote:
I am trying to route my calls through an outside IAX provider. I am
having a problem with which codec to use. The only way I have
successfully been able to make an outgoing call is if i do:
disallow=all
allow=g729
in the sip.conf file
HI!
I configured asterisk to send all outgoing calls to our Gateway. I noticed
when asterisk sends call to gateway that he represents all calls as
asterisk and not as callerID(number of sjphone client registerd to
asterisk).
Can anyone give me an example of such configuration?
Thank you
Hello,
I am new to asterisk (i started to try tree days ago) and i have managed
to setup asterisk to fit my needs so far exept one thing.
I want to setup asterisk as an SIP ISDN Gateway. As I said things are
going well so far. I can make calls from SIP phones to other SIP phones,
and incoming
Just found out my self :-D. Problem was a syntax change in 0.4.0pre1
version of chan_capi.
All the resources I found on the web were written with the old syntax.
Matthias
Am Donnerstag, den 02.06.2005, 17:01 +0200 schrieb Matthias BXhm:
Hello,
I am new to asterisk (i started to try tree days
Greeting!,
I read somewhere that without cdr, Mysql etc it is possible to take
outgoing-call-logs to a text file. (I am not sure please). is it really
possible ? if so, how do I do it? any links to refer?
Thank you.
Kumara
___
Asterisk-Users mailing
Yes I tried the rx and tx values, but no luck there. Then I removed
everything from this line, adsl, fax, etc. and left only asterisk and
still not working. Then I tried the following to get the dialtone and
dial digits myself
exten = _9,1,Dial(${TRUNK}/)
And that didn't work either. I also
I tried that and it didn't work. Then I decided to use a different phone
line. I had not thought about this before, it just didn't occur to me.
And everything worked fine. The phone line that doesn't work is my ADSL
line. Wall to splitter, one side going to ADSL router the other going
into a
Hi Mehmet,
On Tue, May 03, 2005 at 11:20:44AM -0400, You wrote:
I tried that and it didn't work. Then I decided to use a different phone
line. I had not thought about this before, it just didn't occur to me.
And everything worked fine. The phone line that doesn't work is my ADSL
line.
I can't seem to be able to make outgoing calls with X100P card. I can
receive calls fine and it picks up the line and sends the tones, but the
telco doesn't recognize them. While the tones are sent I continue to
hear the dial tone on the line when I pick up a parallel. I also cannot
dial from
On Mon, May 02, 2005 at 01:02:34PM -0400, Mehmet Tolga Avcioglu wrote:
I can't seem to be able to make outgoing calls with X100P card. I can
receive calls fine and it picks up the line and sends the tones, but the
telco doesn't recognize them. While the tones are sent I continue to
hear
I am in Turkey. I imagine this is due to incorrect zone information, but I
can't seem to be able to find the correct values for Turkey. I tried
guessing them with no luck.
DTMF tones in Turkey are the same as the standards everywhere. The other
signalling tones are different (such as dial
Hello,
I have a problem concerning outgoingcalls on my
Asterisk box, with a E1 Digium card. I manage to
receive call on the E1 with no problem and can
transfer to internal ip phones. But the problem
happens when calling from the internal to outside.
Here are my config files :
zaptel.conf :
On Tue, 5 Apr 2005, cereal killer wrote:
I have a problem concerning outgoingcalls on my
Asterisk box, with a E1 Digium card. I manage to
receive call on the E1 with no problem and can
transfer to internal ip phones. But the problem
happens when calling from the internal to outside.
Here
Thank you Peter, it works perfectly now.
I have a problem concerning outgoingcalls on my
Asterisk box, with a E1 Digium card. I manage to
receive call on the E1 with no problem and can
transfer to internal ip phones. But the problem
happens when calling from the internal to outside.
Here
Hi,
here what i have:
[2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004]
Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 onRedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL.
Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following
Hello All,
I have multiple SIP accounts from two different providers, and I'm
wanting
to balance our outgoing calls based on extensions. I thought the
following
would work, but it's sending all calls through the last SIP account
listed
(extensions.conf)
exten =
Put the exensions in different contexts.
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, September 23, 2004 7:48 PM
Subject: [Asterisk-Users] outgoing calls, based on caller extension
Hello All
I have a hard time to forward all my outgoing calls to my SIP
connection.
For incoming calls I set up the registry and it is working perfectly
How can I tell asterisk to forward all calls beginning with 9
to
Gate.Sipserver.com
User : userxx
Password : pwxx
Best regards,
Han
Hi !
I'd like to connect phpgroupware to asterisk: when a user click on a phone
number, his phone rings and he gets connected to the number he just clicked.
I've tried by putting various files in /var/spool/asterisk/outgoing, without
results (we are using SIP phones + CAPI channels).
Is there a
: Monday, December 15, 2003 3:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Outgoing calls for a fancy address book app
Hi !
I'd like to connect phpgroupware to asterisk: when a user click on a phone
number, his phone rings and he gets connected to the number he just clicked.
I've tried
mattf wrote:
Hello,
just use the manager conduit:
Telnet to asterisk server ip_address to port 5038 (as long as you have a
login set up in manager.conf)
and send the following(what is between the dash lines):
---
Action: Login
Username: user
Secret: pass
Action:
On Mon, 2003-12-15 at 14:02, Ludovic Drolez wrote:
Hi !
I'd like to connect phpgroupware to asterisk: when a user click on a phone
number, his phone rings and he gets connected to the number he just clicked.
I've tried by putting various files in /var/spool/asterisk/outgoing, without
Hi!
I'd like to connect phpgroupware to asterisk: when a user click on a phone
number, his phone rings and he gets connected to the number he just clicked.
I've tried by putting various files in /var/spool/asterisk/outgoing, without
results (we are using SIP phones + CAPI channels).
This
Hello guys,
Software: Asterisk From CVS, Linux RH9
No telephony hardware.
I have a SIP service provider (addaline.com) and
(at this time) one Windows SIP (soft)phone.
Problem:
Both from console and from the sip phone if I dial
an autside number it doesn't work.
With sip debug I seen that
I searched the archive and only found references to using
Asterisk as a calling card platform.
What I would like to do is:
1. Dial some extension, say 666 (mom, go figure..)
2. I would like Asterisk to then call the local access number for the
calling card, enter the PIN after some delay,
On Mon, 2003-07-21 at 16:14, John Sutter wrote:
I searched the archive and only found references to using
Asterisk as a calling card platform.
What I would like to do is:
1. Dial some extension, say 666 (mom, go figure..)
2. I would like Asterisk to then call the local access number
66 matches
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