Hello Phil,
On Saturday, April 23, 2016, 11:11:29 PM, you wrote:
> Actually, this is now sorted. It turns out the latest recommended
> configs on the A&A wiki had peer vs. user confusion. On correcting
> this, all was well.
I'm glad you found it. It look me a while to track down that problem
whe
On Sat, 23 Apr 2016 22:45:32 +0100
Julian Beach wrote:
> Hello Phil,
>
> I have a couple of lines with A&A, and I have not been having any
> problems recently. When I have had similar problems in the past, it
> has been an issue with the SIP config. I originally had a number of
> contexts set up
Hello Phil,
On Saturday, April 23, 2016, 12:19:15 PM, you wrote:
> I have checked that the username and password in my config agree both
> ends, and have even tried changing them.
> The bulk of my calls come in on A&A, so I am obviously trying to find
> out what has gone wrong. No-one else is se
I have service with both VoIPtalk.org and Andrews & Arnold (aa.net.uk).
VoIPtalk calls are unauthenticated and reach me fine, but Andrews &
Arnold calls are authenticated. The last call I successfully received
was on Tuesday afternoon. Initially, A&A were for some odd reason not
sending calls to my
On 8/21/15 6:45 PM, Technical Support wrote:
I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring at the handset. I captured the SIP traffic and see
that my M65 is replying with an "488 not acceptable here". From what I
read this is usually codec related but
-
From: Technical Support [supp...@telium.ca]
Received: sexta-feira, 21 ago 2015, 19:46
To: asterisk-users@lists.digium.com [asterisk-users@lists.digium.com]
Subject: [asterisk-users] Incoming calls get 488 error
I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring
I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring at the handset. I captured the SIP traffic and see
that my M65 is replying with an "488 not acceptable here". From what I
read this is usually codec related but both asterisk and the M65 are set
for ulaw a
Hello:
I'm newbie in asterisk, please help me.
My context is as follows:
192.168.4.2 --> Asterisk 11.13.1 complied from source
192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway
When I call from a GSM cell phone, my TG100 GSM gateway answers and
dials extension (configured as a hotline on
On Saturday 19 Jul 2014, Norman Molhant wrote:
> I tried many things on our FreePBX box and found out
> the problem seems somehow linked with the customer's
> extension (or phone number), not his inbound route
> (changing the latter has no effect on the problem).
>
> Creating a new extension with
:43 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] incoming calls fall into echo test mode
>
> Hello all,
>
> Weird trouble here:
> we have 60-some happy subscribers on a FreePBX box, each with its own phone
> number, with no problem at all, exce
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant
Sent: Saturday, July 19, 2014 10:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] incoming calls fall into echo test mode
Hello all,
Weird trouble here:
we have 60-some happy subscribers on a FreePBX box, each with
is being misrouted in the dialplan
From: asterisk-users-boun...@lists.digium.com
on behalf of Norman Molhant
Sent: Saturday, July 19, 2014 10:43 AM
To: Asterisk Users List
Subject: [asterisk-users] incoming calls fall into echo test mode
Hello all
Hello all,
Weird trouble here:
we have 60-some happy subscribers on a FreePBX box,
each with its own phone number, with no problem at all,
except for one (and only one) subscriber who has this
problem: his outgoing calls are ok, but when someone
dials his phone number (be it from our network or fr
Matt,
We are located on Costa Rica and so far there's just 1 TELCO running the
industrym with the CAFTA treatment the carrier had to open for
interconnection but they get to define the ground rules for the
interconnection.
They are arguing ISDN is and "end customer" circuit and you cannot use it
Just out of curiosity, what country are you in?
I agree with the others in this thread, this seems very bizzare that the
telco requires you to do SS7 for dialup connections. I would ask them for
specifics about the "legal" issues with what you are doing - it sounds to me
like they are just trying
hat is the Telco’s problem in doing what the customer was doing before?
>
>
>
> Feel free to correspond directly if you want to.
>
>
>
> Cary Fitch
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> aste
..@lists.digium.com] *On Behalf Of *José Pablo Méndez
> Soto
> *Sent:* Wednesday, November 24, 2010 7:31 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Incoming calls through SS7 for data
> modemtransmissions - possible??
>
>
>
> Hello,
>
> W
: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming calls through SS7 for data
modemtransmissions - possible??
Hello,
We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem
Hello,
We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.
The Telco they were buying the trunks to discovered this configuration and
restricted th
On Thu, 18 Nov 2010, Flavio Miranda wrote:
Looking to dahdi show channles , I realized that all the trunks was in
the same context. So, I have changed this and everything works!
That's why I prefer to work from what Asterisk parsed the file as, not
what the poster thinks :)
--
Thanks in a
to Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
> Date: Thu, 18 Nov 2010 11:53:26 -0800
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Incoming calls
>
> On Thu, 18 Nov 2010, Flavio Miranda wrote:
>
> >
On Thu, 18 Nov 2010, Flavio Miranda wrote:
> I'd like that each analog trunk of my TDM410p was received in different
> extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each
> trunk in a different context and in my extensions.conf, under [default]
> I put such contexts and an espec
Hi all,
I'd like that each analog trunk of my TDM410p was received in different
extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a
different context and in my extensions.conf, under [default] I put such
contexts and an especific estension to answer it. therefore,
Hi all,
After a lot of trouble with a TE110p working with mfcr2 , brazil variant,
everything looks great,but I can not go out of my calls.
When I try I receive the following log:
== Using SIP RTP CoS mark 5-- Executing [33220...@local:1]
Dial("SIP/4804-001a", "DAHDI/g11/33220567,,T"
21 dec 2009 kl. 12.00 skrev jonas kellens:
> My SIP-provider sends my a SIP-invite like this :
>
> INVITE sip:329298y...@80.xx.xx.69:5060 SIP/2.0
> Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c
> Max-Forwards: 70
> From: ;tag=f395877e02bf8eb2fd8f5a0e
> To:
> Call-ID:
My SIP-provider sends my a SIP-invite like this :
INVITE sip:329298y...@80.xx.xx.69:5060 SIP/2.0
Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c
Max-Forwards: 70
From: ;tag=f395877e02bf8eb2fd8f5a0e
To:
Call-ID: f395877e02187250fd8f5...@80.xx.xx.68
CSeq: 1 INVITE
User-Agen
Hello,
i have one question regarding incoming SIP INVITES.
I have a testbed where i have 5 extnsions : 6001 - 6005
Domain : domainA.com
Then i have configured a sip trunk, where my PBX registers to a foreign SIP
Proxy.
All is working fine, until following scenario:
Incoming call from [EMAIL PRO
Thank you,
yes, I changed the PCI Slot and it's the same,
I get a used card from a customer with 2 FXO, same REV, that board was
working on the customer server, put it on mine, and stop working.
I put my board on his server and the board is working perfectly.
I had not test outgoing calls on tha
Hi,
I need libpri, because I have a TE110P E1 with a PRI ISDN service.
2008/7/15 Matt Watson <[EMAIL PROTECTED]>:
> On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote:
> > After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading
> libpri,
> > zaptel, the incoming calls to a TDM400P
On July 14, 2008 08:24:33 pm Jose Flores Galicia wrote:
> After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
> zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
> working.
THis isn;t going to fix your problem... but just FYI, you don't need to
install
Hi Jose -
> After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
> zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
> working.
>
> The board is working, I tested in another server with the 1.2.13 asterisk
> version.
> Also changed the pci slot where the
After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri,
zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop
working.
The board is working, I tested in another server with the 1.2.13 asterisk
version.
When a call is incoming, I do a ztmonitor to check the rx
Le vendredi 20 juin 2008, RoLaNd RoLaNd a écrit :
> Hi all,
>
> i've recently acquired a callcentric account.
>
> i've perfectly setup my sip.conf and extensions.conf to make outgoing
> calls.
Well, I had the same problem and had to debug. In fact for some reason, and
it's a bit hackward, incomin
Hi all,
i've recently acquired a callcentric account.
i've perfectly setup my sip.conf and extensions.conf to make outgoing calls.
but the problem is with incoming calls! when i call in, asterisk doesnt even
see the incoming call!
how is tht possible!
please see the following my config:
sip
100
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Incoming calls not being answered by asterisk
>
> The first thing to do is type "sip debug" on the console and place the
> call from the Sipura. If you get a bunch of SI
Ciao Roand
I think you should buy a book and do some reading to build up your
knowledge.
but in the meantime try something like this in the dialplan
(extensions.conf)
exten => PSTN,1,Answer() ; Answer inbound calls or internal miss-dials
exten => PSTN,2,Playback(silence/1)
exten => PSTN,3
The first thing to do is type "sip debug" on the console and place the
call from the Sipura. If you get a bunch of SIP messages flashing down
your console you know the call is reaching Asterisk and it's most
likely going to be an issue authenticating the call or a problem in
your dial plan.
If no
Hello all,
ive got the following setup currently:
__Sipura 3102-PSTN
|
Lan |
|
|__asterisk
i configured both asterisk and pstn to be able to receive/make calls through
each other using sip of course..
but the thing is i want asterisk that when it receives an inc
I am trying to configure Asterisk for BSNL, india network.
I have successfully configured it for outgoing calls.
When any outside number make any call to trunk then it receives the call
properly but when the call is disconnected by inside extension then outside
phone does not get a busy tone.
Ast
> Hi John, I have copied your changes in the Peer Details section of
> the trunk set up
then I went ahead and added the DID number in the
> Income Routes but still did not work. I tried the number alone and
> also tried adding the + sign in front of it. Do you think we should
> have any changes in
User Details section of the trunk set up?
Thanks much,
Paulo
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonn R
Taylor
Sent: Wednesday, January 02, 2008 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls
ere is no auth used.
Jonn
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro
Sent: Wednesday, January 02, 2008 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls
Hi Jose, I apologize for th
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro
Sent: Wednesday, January 02, 2008 10:21 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming Calls
I am having a problem that I would like to verify if someone could help...I am
using
o
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jose P.
Espinal
Sent: Wednesday, January 02, 2008 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls
Hi Mr. Paulo,
Could you please explain this situation in a more
> Hi Paulo,
>
> Make sure your DID number is in the e.164 format, ie, +15551234567.
> I had the same issue with bandwidth.com and that fixed the problem.
>
>
> HTH,
>
> Zaheer
Zaheer is right. Everything from bandwidth is 164 format. So you need the
+15551234567 in the dial plan as well as in y
AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Incoming Calls
I am having a problem that I would like to verify if someone could help.I am
using bandwith.com as my SIP TRUNK provider. When I place the phone number
in the DID number field ( using Elastix) it gives me an error
Hi Mr. Paulo,
Could you please explain this situation in a more detailed way to see
how can we help you?
Regards,
Paulo Pinheiro wrote:
I am having a problem that I would like to verify if someone could
help...I am using bandwith.com as my SIP TRUNK provider. When I place
the phone numb
Paulo Pinheiro wrote:
>
> I am having a problem that I would like to verify if someone could
> help…I am using bandwith.com as my SIP TRUNK provider. When I place
> the phone number in the DID number field ( using Elastix) it gives me
> an error message stating the phone number I dialed is not i
I am having a problem that I would like to verify if someone could
help...I am using bandwith.com as my SIP TRUNK provider. When I place
the phone number in the DID number field ( using Elastix) it gives me an
error message stating the phone number I dialed is not in service. When
I leave the DID n
"Glare" that's what it's called, if the number you advertise as your
business number is zap/1 then use zap/G1 to dial out, otherwise use
zap/g1 to dial out. This will reduce but not eliminate the problem.
http://www.telos-systems.com/techtalk/gldefs.htm#Glare
On 10/18/07, Gustavo Gonzalez <[EMAI
Hello I have a question about incoming calls on TDM400P cards. I want to
know why an incoming call appear in a sorpresive way on a phone that I
pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
channels )that are free. I have four lines connected to my TDM400P card and
when
j Polívka
To: asterisk-users@lists.digium.com
Sent: Saturday, August 18, 2007 3:25 PM
Subject: [asterisk-users] incoming calls in SIP
Hi, when I try to call in it tells me: NOTICE[11664]: chan_sip.c:10637
handle_request_invite: Failed to authenticate user "585415198" ;tag=as18abefe8
Hi, when I try to call in it tells me: NOTICE[11664]: chan_sip.c:10637
handle_request_invite: Failed to authenticate user "585415198"
;tag=as18abefe8
Can someone help me out of this? I have Asterisk 1.2 on the Ubuntu 7.04.
Outcoming and internal calls functions well. Thanks
sip.conf:
[general]
c
your problem is that you need to handle this in your dialpan to achieve
which DID has been dialed! look for SIPGETHEADER application on asterisk,
you shoul look for variable "to" where it comes the DID
On 11/27/06, Frederico Madeira <[EMAIL PROTECTED]> wrote:
I have an asterisk box registering
I have an asterisk box registering 100 numbers on a voip provider.
Numers are: 2546.1000 to 2546.1099
My problem is that every incoming call arrived to number 2546.1099 that is
the last number to register on voip provider. The correct is call arrive in
destination number.
See this exaple:
I call t
Jay R. Ashworth<[EMAIL PROTECTED]> Wrote on: 9/20/2006 4:00 PM:
> On Wed, Sep 20, 2006 at 08:48:17AM -0400, joea, j4computers wrote:
>> Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card
>> will have two FXO and two FXS modules.
>>
>> Two incoming analog lines, which need to be t
On Wed, Sep 20, 2006 at 08:48:17AM -0400, joea, j4computers wrote:
> Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card
> will have two FXO and two FXS modules.
>
> Two incoming analog lines, which need to be treated as distinct
> entities. Meaning, for example, line 1= company1,
Just delving into asterisk, using trixbox 1.2 and a TDM400p. The card will
have two FXO and two FXS modules.
Two incoming analog lines, which need to be treated as distinct entities.
Meaning, for example, line 1= company1, line2=company2, or line 1= home line,
line2=business line. In my limi
ch 2006 4:15 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p > > > Paul C wrote: > >> I am running Asterisk
1.0.9
and have been running all my calls through a > >> VSP over a IAX2
> When a friend calls, I would like for him to enter a 4 digit password
> in order to access to a sub-menu, if no password is entered, then the
> welcome msg is said ...
>
> Any hints on how to do that ??
In your incoming-rtc context, define an extension (let's say 1234)
exten => 1234,1,Authentica
Hi,
I run an asterisk server. The configuration is very basic.
Here is my question :
When someone calls my phone line, which is connected to an FXO card,
asterisk is answering using the context :
; Incoming calls goes to this default context :
[incoming-rtc]
include => postes-sip
;
exten => s,1,
ling List - Non-Commercial Discussion'"
Sent: Sunday, March 05, 2006 6:52 AM
Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p
I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of
the site. It is sending CRC errors )to Telsta, drops all
aul C" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 01, 2006 5:15 PM
Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p
> > Paul C wrote:
> >> I am running Asterisk 1.0.9 and have bee
: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Sunday, March 05, 2006 7:52 AM
Subject: RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p
> I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of
> the site. It is s
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul C
Sent: Wednesday, 1 March 2006 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110p
> Paul C wrote:
>> I am running Asterisk
Paul C wrote:
I am running Asterisk 1.0.9 and have been running all my calls through a
VSP over a IAX2 trunk however we have recently purchased and connected a
TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make
outgoing calls via it fine, however incoming calls are droppe
Paul C wrote:
I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped a
I am running Asterisk 1.0.9 and have been running
all my calls through a VSP over a IAX2 trunk however we have recently
purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through
Optus. I can make outgoing calls via it fine, however incoming calls
are dropped after a f
-Commercial DiscussionSubject: RE:
[Asterisk-Users] Incoming Calls Getting Crossed - Weird
It is not the firmware but a setting. "Call Join on
Xfer (2 calls)"
Make
sure that is is set to OFF.
SNOMS
are great ophone but 'features' like this drive me crazy.
Alex
Fr
beratoreSent: Monday, February 20, 2006 6:38 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Incoming Calls Getting Crossed - Weird
LMAO!
app_PatientDatingService
Yes I have all Snom 360's, are you thinking the problem
is
EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
LopezSent: Monday, February 20, 2006 6:14 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Incoming Calls Getting Crossed - Weird
You have stumbled across the new undocumented fe
You have stumbled across the new undocumented feature
app_MakeAFriend or posably app_MakeAConsultantCrazy. Look to see if you
have any SNOM phones,
Hey, I got a weird one
for you guys, I am running vanilla 1.2.4 and have all incoming
calls come in as SIP from teliax. Twice ov
Hey, I got a weird
one for you guys, I am running vanilla 1.2.4 and have all incoming
calls come in as SIP from teliax. Twice over the past week 2 callers who
have called in around the same time end up talking to each other instead of
going through the ivr or at some point during the IVR.
Hi there everybody,
We are running Asterisk 1.2.1 with a TE410P card attached to one
PRI ISDN line, and many SIP phones. Yesterday we ended up in a situation
where all incoming calls were giving the engaged tone. Every time some
tried to ring in we got:
Jan 4 14:56:32 WARNING[896] chan_za
Hi,
I'm not really sure if this helps you, but as far as I remember, the
diastring with chan_capi-cm-0.6 is not
"CAPI/g1/0299546476:b${EXTEN},30,r" but
"CAPI/g/[/]" or in your case
"CAPI/g1/${EXTEN}/b,30,r".
To set your CallerPresentation, use the SetCallerPres() in your
Dialplan, which is
uld
route the call to an internal extension, but when I ring the msn 0299546476,
I can't hear anything except a tone dropping the call.
Armin, I will appreciate if you can put me in the right direction?
Cheers
PolAus
From: Armin Schindler <[EMAIL PROTECTED]>
To: Esteban Guana-Jarrin
On Wed, 26 Oct 2005, Esteban Guana-Jarrin wrote:
> Can anyone please provide some help. I have installed an AVM fritz card on an
> asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card
> driver and
> chan_capi-cm-0.6. According to the installations guide I can now see that the
>
Can anyone please provide some help. I have installed an AVM fritz card on
an asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card
driver and chan_capi-cm-0.6. According to the installations guide I can now
see that the CAPI channel in asterisk is up,
*CLI> capi info
Contr1:
Hi Everyone,
Got a setup as follows:
Telco > Siemens HiCom 300E <> Asterisk1
Asterisk2 <> Siemens HiPath 4xxx
The solution works except for one problem. Incoming calls from the telco get
redirected to the Asterisk1 box with the correct extention, only if there is
Hi,
stupid question:
how can I let to call an extensions from outside?
Untill now, I've just the possibility to call our number and then,
after the system answer, dial the extension.
My sistem is like this:
SER -> internal extensions
Asterisk -> incoming/outgoing gateway.
FaberK
--
.:FaberK:.
On Fri, Jul 22, 2005 at 04:41:01PM +, salahssaid2.salah wrote:
> > From: Andres Tello Abrego <[EMAIL PROTECTED]>
> > Date: Fri, 22 Jul 2005 06:53:19 +
>
> > youa re using -v option multiple times at startup.
> > That message is perfectly fine.
And thus see quite a few messages that are n
llo Abrego <[EMAIL PROTECTED]>
> A: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Objet: Re: [Asterisk-Users] incoming calls
> Date: Fri, 22 Jul 2005 06:53:19 +
> youa re using -v option multiple times at startup.
> That message is perfectly fin
youa re using -v option multiple times at startup.
That message is perfectly fine.
ali kia wrote:
hi ;
our * handle good the outgoing calls but 4 incaming calls we have this
msg :
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2
(Ring/Answered)...
someone ha
hi ;
our * handle good the outgoing calls but 4 incaming calls we have this msg :
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??,
thx in advance,
CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir d
hi ;
our * handle good the outgoing calls but 4 incaming calls we have this msg :
Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)...someone have an idea ??,
thx in advance,
CaraMail met en oeuvre un nouveau Concept de Sécurité Globale à partir de
On 20:42, Sun 10 Jul 05, Rene Kluwen wrote:
> Same here,
>
> Audio quality is ok. SIP registration sucks. The helpdesk makes me believe
> that it is a problem on my side: "Your asterisk doesn't respond to a sip
> request in time". But I have no problems with any other provider, except
> with Budge
Same here,
Audio quality is ok. SIP registration sucks. The helpdesk makes me believe
that it is a problem on my side: "Your asterisk doesn't respond to a sip
request in time". But I have no problems with any other provider, except
with Budgetphone. I am not even getting a SIP request, so how do I
PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl
Try using insecure=very in your peer definition. That makes asterisk
not require authentication from your peer if comes from the ip address
give in host
Try using insecure=very in your peer definition. That makes asterisk
not require authentication from your peer if comes from the ip address
give in host=xxx.xxx.xxx.xx directive.
That helped me receiving calls from my sip provider, which had exactly
the same problem.
Julian.
On 7/10/05, Peter Ra
terisk Users Mailing List - Non-Commercial
Discussion
Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl
Long short,
Maybe X-Ten has an stun relay setup and Asterisk doesn't?
Rene Kluwen
Chimit
> (this time with subject)
>
> Hello,
>
> Im trying
On 19:16, Sun 10 Jul 05, Peter Raaijmaker wrote:
> (this time with subject)
>
> Hello,
>
> I?m trying to get Asterisk to accept incoming calls from budgetphone.nl.
> When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
> tone.
> I tried X-lite, which worked perfect, so
Long short,
Maybe X-Ten has an stun relay setup and Asterisk doesn't?
Rene Kluwen
Chimit
> (this time with subject)
>
> Hello,
>
> Im trying to get Asterisk to accept incoming calls from budgetphone.nl.
> When I dial my budgetphone nr on a PSTN KPN line it immediately gives a
> busy
> tone.
(this time with subject)
Hello,
Im trying to get Asterisk to accept incoming calls from budgetphone.nl.
When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
tone.
I tried X-lite, which worked perfect, so my modem (with nat) probably is not
the problem.
I did a sip deb
On Mon, 2005-06-06 at 15:25 -0400, David Sampson wrote:
> I have 2 4-port Digium FXS cards in my system. I would like to play a
> different recording based on which trunk rings. Any pointers?
>
> Thanks
>
>
This is really a no brainer if you read the documentation. Simple have
each c
I have 2 4-port Digium FXS cards in my system. I would like
to play a different recording based on which trunk rings. Any pointers?
Thanks
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1:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming calls picked-up then simply hanged-up
Are you sure you have context=from-pstn in your zapata.conf for the fxo
channels?
Julian.
On 5/12/05, fhunter <[EMAIL PROTECTED]> wrote:
> I d
12, 2005 1:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Incoming calls picked-up then simply
hanged-up
Yep. Check context and it point to from-pstn
Any other ideas.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PRO
Discussion
Subject: Re: [Asterisk-Users] Incoming calls picked-up then simply hanged-up
Are you sure you have context=from-pstn in your zapata.conf for the fxo
channels?
Julian.
On 5/12/05, fhunter <[EMAIL PROTECTED]> wrote:
> I don't think my first posting went thru.
>
> I
Yep. Check context and it point to from-pstn
Any other ideas.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M.
Sent: Thursday, May 12, 2005 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
Are you sure you have context=from-pstn in your zapata.conf for the
fxo channels?
Julian.
On 5/12/05, fhunter <[EMAIL PROTECTED]> wrote:
> I don't think my first posting went thru.
>
> I am trying to set up Asterisk for the first time. I am new to this.
> I am using [EMAIL PROTECTED]
> I have a
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