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
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
Hi,
By the sip trace is very difficult to tell because the SIP messages are fine.
Try to enable all codec, and if possible copy and paste your asterisk sip
configuration for this peer.
Enviado do meu telefone Android usando o Symantec TouchDown (www.symantec.com)
-Original Message-
F
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
check your logs /var/log/asterisk/full -- make sure your verbosity is
set high enough to do you good and you wll probably find the answer.
Pat Collins wrote:
> Perhaps assigned as a test number somewhere along the line?
> Are these ISDN, SIP, IAX calls?
> There are MANY smart people on this list
Perhaps assigned as a test number somewhere along the line?
Are these ISDN, SIP, IAX calls?
There are MANY smart people on this list.
Maybe sharing the relevant configs and traces is a good place to start???
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
You might get a better response on the FreePBX forum. (FreePBX adds pre-built
dialplan elements onto standard asterisk. This forum is more for Asterisk)
But some suggestions:
SSH to your PBX
enter the Asterisk CLI
set verbose to 10
Call into the problematic number
...and watch where the call i
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
ment” but no PRIs.
>
>
>
> A lot of this equipment was available by the pound a few years back.
>
>
>
> Cary
>
>
> --
>
> *From:* José Pablo Méndez Soto [mailto:aux...@gmail.com]
> *Sent:* Wednesday, November 24, 2010 8:34 PM
> *T
Thanks Cary,
What happens is, the Telco won't allow the small company to resell the ISDN
connections, meaning, they bought the trunks and DIDs, then sold dialing
plans to route incoming calls through the PRIs out the Internet. This is not
the issue though. We definitely have to migrate to an SS7 c
I am not sure where you are and what legal conventions are involved.
Are you saying the Telco (and legal restrictions) say you cant send calls
to the internet via the AS5300 but you can if Asterisk does it directly?
What is the logic in that?
Or are they saying your Telco to Asterisk tru
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
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:
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
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
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
> 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
Paulo,
I am using them also. All call traffic to and from them must be in e164 format.
So your calls have to look like this, +15615551212 or +011 for international.
They do not let you set the caller name, but will let you set calling number
and that also needs to be e164 format.
Jonn
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
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
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paulo Pinheiro
Sent: Wednesday, January 02, 2008 11:21 AM
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
"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
It seems that the "other end" is having an issue authenticating you. I have
seen lots of switches act up with asterisk if you don't tweak you settings in
sip.conf just right. Do a SIP debug and have a look at te INVITE request.
- Original Message -
From: Ondrej Polívka
To: asteris
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
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,
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
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
-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
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
>
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
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.
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
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
This is what I got:
May 12 11:12:53 VERBOSE[1376]: -- Starting simple switch on 'Zap/4-1'
(Note that the line went dead on the calling phone before this next stuff
ever appeared)
May 12 11:13:01 WARNING[1376]: CallerID returned with error on channel
'Zap/4-1'
May 12 11:13:01 VERBOSE[1376]: -- Ex
Thanks I will give that a try.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John covici
Sent: Thursday, May 12, 2005 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Incoming calls picked-up then simply hange
Soren,
I tried the variable UserByAlias=no and it worked for me. Thank you very much
for this note !
Selon Soren Rathje <[EMAIL PROTECTED]>:
[EMAIL PROTECTED] wrote:
Good to hear I am not alone.
Actually, I am using the Nufone's h323 module. Still this creates the
problem. I had a braod look at th
[EMAIL PROTECTED] wrote:
> Good to hear I am not alone.
>
> Actually, I am using the Nufone's h323 module. Still this creates the
> problem. I had a braod look at the code and it seems that it is not
> possible that incoming calls go to other places than "general"
> context (I am not sure I underst
onday, February 07, 2005 5:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] incoming calls in h323 do not come to right
dialplan
I am also facing the same problem, running on chan_h323 and
CVS-HEAD-12/17/04-15:07:40.
Anybody managed to solve this?
I am also facing the same problem, running on chan_h323 and
CVS-HEAD-12/17/04-15:07:40.
Anybody managed to solve this?
Cheers
On Mon, 7 Feb 2005 07:10:55 -0800, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hello,
> I am moving topic from asterisk-dev list to asterisk-users list. Did anyone
>
I have asked the forum several weeks ago, but no one responded.
I have found out that, [user] selection is working only
With h323 (nofune) channel, not with oh323 from inaccess networks.
I might be wrong, but with h323 it works.
R.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAI
> OK I have 12 phone lines connected to 3 digium TDM04B cards on the same
> server. I must do the following thing :
>
> The first 10 lines will be use by one company and the 2 left by another
> one. For outgoing calls it's quite easy I just create 2 different group
> and let them dial on a dif
On Thu, 2005-02-03 at 07:07, Martin Roy wrote:
> OK I have 12 phone lines connected to 3 digium TDM04B cards on the same
> server. I must do the following thing :
>
> The first 10 lines will be use by one company and the 2 left by another
> one. For outgoing calls it's quite easy I just create 2
Martin,
You would want to put them into different contexts (in the zapata.conf).
Using different contexts you can slice and dice up your channels to your
hearts content.
You would then be able to have an S,1,Answewr in that context.
Jim
-Original Message-
From: Martin Roy [mai
Martin Roy wrote:
OK I have 12 phone lines connected to 3 digium TDM04B cards on the
same server. I must do the following thing :
The first 10 lines will be use by one company and the 2 left by
another one. For outgoing calls it's quite easy I just create 2
different group and let them dial on
Yes, the normal phone is the phone that is producing the quarter
ring. Asterisk doesn't see anything. The phone gets what appears
to be a normal dial tone.
Thanks,
Jon.
On Friday 28 January 2005 11:11 pm, Greg Blakely wrote:
> Tip side open on the analog line? Have you taken a butt set or no
Tip side open on the analog line? Have you taken a butt set or normal
phone and attached it directly to the outside line to see if you get
dial tone?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jon Gabrielson
> Sent: Friday, January 28, 200
On Tuesday 28 December 2004 04:32, C F wrote:
> Just a note on this. I tried using an external device with the TDM400
> configured as 4 FXO to ring even with asterisk. But no matter how I
> configured it, asterisk always picked up. and the external device
> didn't ring (just the first ring for Call
-- Forwarded message --
From: C F <[EMAIL PROTECTED]>
Date: Tue, 28 Dec 2004 13:21:45 -0500
Subject: Re: [Asterisk-Users] Incoming Calls
To: Rich Adamson <[EMAIL PROTECTED]>
I didn't try Dial but I did try wait and it didn't help. I'll try dial
and see
Not sure why it didn't work for you unless we are talking about two
different things. It does work for me and has been working just fine
for over a year now.
> Just a note on this. I tried using an external device with the TDM400
> configured as 4 FXO to ring even with as
Just a note on this. I tried using an external device with the TDM400
configured as 4 FXO to ring even with asterisk. But no matter how I
configured it, asterisk always picked up. and the external device
didn't ring (just the first ring for CallerID to come in).
On Mon, 27 Dec 2004 07:04:00 -0600
> Here is where the problem is.
>
> When the call comes in, it will be ringing on 2 of the FXO ports,
> and all the other phones in the office. I would like various / all
> the IP phones to ring, however asterisk must not answer the call
> while that is happening or else the normal extension wou
This is still a nasty design flaw (bug) in Asterisk.
IAX is similarly bugged.
I can only ask you to wait a little bit longer until I post the solution.
Ian Chilton wrote:
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it see
Hi,
> For incoming calls, Asterisk matches peer's on IP, meaning that the
> first peer it finds will match. This is the *last* one you have in
> sip.conf. The context given in that peer must have *all* extensions you
> need for incoming calls, which is the extension at the end of the
> registe
For incoming calls, Asterisk matches peer's on IP, meaning that the
first peer it finds will match. This is the *last* one you have in
sip.conf. The context given in that peer must have *all* extensions you
need for incoming calls, which is the extension at the end of the
register= line in the
way...it's alright now.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.
Sent: Tuesday, March 02, 2004 10:08 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Incoming calls.
OK...first off than
OK...first off thanks for the responses. However, I'm still having the
same sort of issue. I've looked through the two places that you
gentlemen suggested, and am still having the problem. Here is the error
message that I am receiving:
"-- Starting simple switch on 'Zap/1-1'
Mar 2 09:58:18 WAR
Hi Mark,
its sets the context in zapata.conf.. put a context= somewhere
above where you define the channels
then in your extensions.conf, create the context if you've not already, start
with a exten=> _.,1,Dial(blah...) then modify from that to suit your setup
Steve
On Mon, 1 Mar 2004, Mark M
Mark,
Zaptel is where it is told to go.
I have mine set to incoming, and a context of incoming in my
extensions.conf.
My configs are here http://www.codepipe.com/id25.htm without any UID's or
PWD's.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Beh
any IAX
packets coming in on my network interface to even start worrying if it's
an incoming config issue.
Next...
p.s. thanx anywayz..
Jamie
> -Original Message-
> From: Paul Cheng [mailto:[EMAIL PROTECTED]
> Sent: Friday, 12 September 2003 4:00 PM
> To: [EMAIL PROTECTED
Don't know if this will help, but have you seen:
http://www.junghanns.net/asterisk/page12.html
?
On Friday, September 12, 2003, at 03:06 AM, Jamie Carl wrote:
Hey all,
I was playing around with IAXTEL last nite and have outgoing calls
working a treat. I'm sure I woke a few people up in the
>When an incoming call is attempted Asterisk displays the following:
>
>Warning [131081]: chan_sip.c Line 1991: (__transmit_response): Unable to
>determine sequence number from ''
I've been seeing this problem within the last little while (sorry I
can't be more specific).
This probably isn't goi
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