Hello
I had this codec negotiation with oh323
call. i used G723 codec and the provider had G729 as first priority. In this
situation what ever number i dial i used get "No one there to answer the call".
As soon as i changed my codec to G729 the call went through but had other
problems,
Digging around on google the other night, I found an example sip.conf for
cisco 7960's with multiples lines, but have quickly forgotten where it was,
and didn't bookmark it.
Would someone be kind enough to post a link to this if they know where it's
at, or post an example sip.conf that they
How come * says 1010 is BUSY in the trace below? I would have guessed
UNAVAILABLE since 1010 is not logged on/registered.
That's what has been programmed in the asterisk code and has been that way
since the beginning of time. I don't like it either, but I'm not a programmer
and can't change it.
On Sat, 2004-02-21 at 09:06, Costa Tsaousis wrote:
callgroup= ; UP
pickupgroup= ; UP
Q4: Since a user cannot accept calls, why to setup call pickup for
him/her?
Sorry, haven't used or checked call groups. Anyone else?
No answer on this yet...
I use pickup groups just fine (using
G'day,
I was wondering if anyone knows if its possible to have conference calls
distributed between multiple * servers? (as opposed to having conferences
on just one server)
I'd imagine this would be helpful in reducing latency with international
calls.
Thanks,
Andrew Griffiths
Use the outgoing call feature of asterisk to have the servers join each
others conferences. It's very simple.
[EMAIL PROTECTED] wrote:
G'day,
I was wondering if anyone knows if its possible to have conference calls
distributed between multiple * servers? (as opposed to having conferences
on
On Sun, Feb 22, 2004 at 08:21:18AM -0500, John Fraizer wrote:
Use the outgoing call feature of asterisk to have the servers join each
others conferences. It's very simple.
Okay, that makes sense, thanks.
Thanks,
Andrew Griffiths
___
hi everybody,
just ran into trouble...
I place an outgoing call from my zap (hfc in
NT-mode) via chan_capi. the I transfer the call to a SIP-phone (x-lite
orgs budgetone). if the called person now presses any key on his phone my
asterisk segfaults :(
any ideas?
anyone??!
Thanks Eric. I'll configure my system for IAXTEL today and try it
Have a great week end
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Saturday, February 21, 2004 8:11 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Calling SIP
Hi,
Did GrandStream Voip-phone and HT-286 Analog adapter talk using GSM 6.01 Codec ??
In tests using asterisk this codec it's the best for my kind of connection, did both Hardwares use this
codec to talk ?
In page they don't mention this.
Thanks alot.
Ps. ILBC ?? Talk too ??
Thanks
Hi -
I have a problem with DTMF dialling when in conjunction with a
Grandstream SIP phone.
Problem: My SIP phone is set to 'early send' (coz I don't like waiting
too long after dialling!)...
I want to dial a long series of DTMF carrier access digits before the
digits dialled from the phone
What is the best or simplest method
to connect 4 fax
machines into a
* system?
Fax
-
ATA-186 -
Switch - * Server - VoIP or PSTN
Fax - * Fax Server with TDM 400P -
Switch - * Server - VoIP or PSTN
Would like a dedicated # on the T1
to do direct to the fax machine.
Would love your
First, please no HTML email. Not only does it more than double the size of
your messages, it is plain bad etiquette for mailing lists.
Second, do not reply to a post, erase everything and then type your message
-- it breaks threading in a horrible way and buries your question in the
middle of
There is no support in Asterisk for FoIP. You might use alaw or ulaw g.711
if youre on the same LAN as your *. Otherwise its best effort transmission
and not really reliable
Regards,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Barry Fawthrop
I've started the
integration of * with my PBX and I need to get a good understanding on the tones
sent by it on the AA/VM port.
What is the best way
to do that. I could create an AGI but I'm afraid to loose some
information
Thanks
Jacques
I was wondering how difficult it would be to add a 2-3 sec delay before
the Name or Extension is said. Some of our customers who call in are
complaining that when they search for an employee by name by the time
they have put there phone back to there ear the name has already been
said.
Hope my
Tim Robinson wrote:
Problem: My SIP phone is set to 'early send' (coz I don't like waiting
too long after dialling!)...
Turn off early send and then hit the SEND button when you're done dialing.
You don't have to wait on the phone to decide you're done dialing.
John
1st question
I am able to receive calls to Asterisk via BRI, but when I attempt to
dial out from Asterisk to ISDN I always get
-- Executing Dial(SIP/1001-3fb0, Modem/g1:6658218) in new stack
-- Called g1:6658218
-- Modem[i4l]/ttyI1 is busy
-- Hungup
Yes, this does solve the problem, and it is what I have done to get
round the problem. It is not an elegant solution though, as the
equivilent - overlap sending - works just fine using a Digium E1 card.
I think it is indicative of a buffering problem in the Asterisk X100P
driver that needs
it appears that Asterisk does not support groups using isdn4linux. You
need to put in the full channel name instead of Modem/g1.
At least, this worked for me anyway.
Rgds
Tim
Tomica Crnek wrote:
1st question
I am able to receive calls to Asterisk via BRI, but when I attempt to
I did this also, but the result is the same
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Robinson
Sent: Sunday, February 22, 2004 7:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 2 questions about ISDN BRI
it appears that Asterisk
Hi!
I am able to receive calls to Asterisk via BRI, but when I attempt to
dial out from Asterisk to ISDN I always get
-- Executing Dial(SIP/1001-3fb0, Modem/g1:6658218) in new stack
-- Called g1:6658218
-- Modem[i4l]/ttyI1 is busy
-- Hungup 'Modem[i4l]/ttyI1'
Change
Hi Jiri,
Been there. We switched from INFO to RFC2833 for this same reason.
Take a look at:
http://bugs.digium.com/bug_view_page.php?bug_id=0001033
Not only retransmissions are affected but out of order packets too.
This behaviour can be partly blamed on the RFC:
In addition, the INFO
Thanks for stripmsd=0, it helped! It supports groups also.
It is one US Robotics card
[chan_modem.so] = (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_aopen.so = (A/Open (Rockwell
Chipset) ITU-2 VoiceModem Driver)
== Loading
On Sun, 22 Feb 2004, Andrew Kohlsmith wrote:
First, please no HTML email. Not only does it more than double the size of
your messages, it is plain bad etiquette for mailing lists.
Second, do not reply to a post, erase everything and then type your message
-- it breaks threading in a
I am having trouble getting SIP phones to register with Asterisk. I know
that the phone can register with FWD and I have used tcpdump to see the
registration packets arrive at the Asterisk server, but nothing goes back.
How should I attack the problem?
What debugging tools exist to try to
- Original Message -
From: James Golovich [EMAIL PROTECTED]
James Golovich wrote:
I don't explicitly disable echocancellation on the channels I use for fax,
and zaptel always seems to detect the tone to disable echo cancellation
from the fax. I send/receive all my faxes over IAX2
Rich Adamson wrote:
How come * says 1010 is BUSY in the trace below? I would have guessed
UNAVAILABLE since 1010 is not logged on/registered.
Sounds right to me.
That's what has been programmed in the asterisk code and has been that way
since the beginning of time.
Is that right? I was afraid it
On Sun, 22 Feb 2004, Barry Fawthrop wrote:
- Original Message -
From: James Golovich [EMAIL PROTECTED]
James Golovich wrote:
I don't explicitly disable echocancellation on the channels I use for fax,
and zaptel always seems to detect the tone to disable echo cancellation
Hi,
anyone here running SNOM phones with TAPI integration with Outlook?
Any other hardware phone with some TAPI integration?
rgds
pos
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
I have the latest as well. Not sure why it's not working.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Boehnlein
Sent: Sunday, February 22, 2004 12:23 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Agents / ackcall
On Sat, 21 Feb 2004,
On Fri, 20 Feb 2004 20:12:36 +0100, Jean-Marc V. Liotier wrote:
On Fri, 2004-02-20 at 17:48, Warren H. Prince wrote:
we're in the process of building a business application, and are
reviewing providers as well. While we haven't selected one, or found
one we're really happy with, our
George Pajari wrote:
I am having trouble getting SIP phones to register with Asterisk. I know
that the phone can register with FWD and I have used tcpdump to see the
registration packets arrive at the Asterisk server, but nothing goes back.
How should I attack the problem?
What debugging tools
Use the outgoing call feature of asterisk to have the servers join
each
others conferences. It's very simple.
Sorry, I am not quite sure what is the outgoing call feature. Would you
please elaborate a bit.
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
Hi!
I am having trouble getting SIP phones to register with Asterisk. I know
that the phone can register with FWD and I have used tcpdump to see the
registration packets arrive at the Asterisk server, but nothing goes back.
Sounds like your SIP client is behind NAT and you need nat=yes in
On Mon, 23 Feb 2004, dkwok wrote:
Use the outgoing call feature of asterisk to have the servers join
each
others conferences. It's very simple.
Sorry, I am not quite sure what is the outgoing call feature. Would you
please elaborate a bit.
Asterisk has a few different ways to
I have 5 Hyuandi pstn handsets and wondering it is possible to plug into
tdm400p or ata-286?
Perhaps a general question whether handsets from PSTN pbx can be reused
with Asterisk?
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
___
Asterisk-Users
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 22, 2004 8:52 PM
Subject: Re: [Asterisk-Users] SIP extension busy when not available ??
Although the current logic does not require a sip phone to register, it
would
seem like
If you can plug the phone into a normal line coming from your Telco (and
make and receive calls) then the phones will work with Asterisk.
On Mon, 2004-02-23 at 02:05, dkwok wrote:
I have 5 Hyuandi pstn handsets and wondering it is possible to plug into
tdm400p or ata-286?
Perhaps a general
I am beginning a project to integrate * with a Cisco AS5350 gateway for
inbound/outbound termination. If anyone has experience with this, what
channel type would you recommend? H.323, SIP or MGCP?
I've scoured the archives to see what channel type would be the most
stable but haven't found a
I am having an issue with registering SIP client w/ NAT. I have set
this up before on other boxes... But for some reason this one is not
acting the same... I have attached a sip debug from the registration...
For what ever reason it does not appear to be setting up the nat session
correctly
Take a look at sample.call in in the source tree. It explains it all.
dkwok wrote:
Use the outgoing call feature of asterisk to have the servers join
each
others conferences. It's very simple.
Sorry, I am not quite sure what is the outgoing call feature. Would you
please elaborate a bit.
On Sun, 22 Feb 2004, Matthew B Marlowe wrote:
I have the latest as well. Not sure why it's not working.
I've updated the patch to add support for customer preackannounce
messages, coinfigureable in the agents.conf file.
The latest patch can be downloaded from:
I use SIP to/from our AS5300... H.323 didn't work well for me...
Also... be aware the the AS53XX access servers will NOT do VoIP to VoIP
calls... it will do PSTN-to-VoIP or VoIP-to-PSTN just fine.. Your Asterisk
configuration will become quite intricate especially with authentication
and
Submit it to http://bugs.digium.com as a feature
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Boehnlein
Sent: Sunday, February 22, 2004 8:08 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Agents / ackcall
On Sun, 22 Feb 2004, Matthew B
You should make the preackannounce option in the queue.conf file so you
can set a different sound for each queue.
call from tech support, press # to accept
call from sales, press # to accept
(although this works great for me)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Sun, 22 Feb 2004, Matthew B Marlowe wrote:
You should make the preackannounce option in the queue.conf file so you
can set a different sound for each queue.
call from tech support, press # to accept
call from sales, press # to accept
(although this works great for me)
I thought
Well, good job. I just implemented it and it works. So I'm happy.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Boehnlein
Sent: Sunday, February 22, 2004 8:43 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Agents / ackcall
On Sun, 22 Feb
On Sun, 22 Feb 2004, Greg Boehnlein wrote:
file is in the channel code. I don't know enough about Asterisk
Architecture or design philosphy to know wether what I'm doing is breaking
protocol, but since it is open source, people are free to comment and
expand on it, or write something
I would like to bring this subject up again.
Is there anyone with experience of implementing these type of systems?
/Chris
On Wed, 28 Jan 2004, Christopher Arnold wrote:
Hi all!
i would like to implement a One2One application.
Basically i want one or two queues hooked up against
A quick rundown of my setup...
Norstar FXS - X100P- Asterisk (Starlight) - IAX2 - Asterisk (Voipsrv)
- Cisco 7940 SIP
Now what I've done is to setup a simple menu system on my Asterisk (Voipsrv)
to allow the caller to select which extension to ring off Voipsrv. The main
benefit being that I can
If they are analog POTS type sets then any ATA (Sipura or Grandstream)
should work.
If they are a Digital set thats special for the PBX, then they
are a good door stop ;)
Basicly if you can plug it into a Tip/Ring analog line and it
works then it should with a ATA. Some sets use the yellow/blk
SIP with G.711 for local lines or SIP with gsmfr for long disctance
(slow connection).
I used AS5350 several months with *, then i recieved my E100Ps and moved
the Cisco to History
(where all proprietary solutions must go).
P.S.
The Cisco is for selling for around 11K Euro (New, used in rack 4
ngrep.
There is some patch for better displaying from iptel, that works grat
George Pajari wrote:
I am having trouble getting SIP phones to register with Asterisk. I know
that the phone can register with FWD and I have used tcpdump to see the
registration packets arrive at the Asterisk server, but
hmm, this pages must be fixed. Looks terrible on all NGlayout based browsers
Philipp von Klitzing wrote:
Hi there,
please comment and adjust or enhance as you find appropriate:
http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning
Typical questions asked on the mailing
Ok, after much stuffing around with the configs to sort it, I've narrowed
the problem down to DTMF passing from the Norstar extension as being what
breaks my setup.
If I'm on a call with someone on a Norstar extension from my system, and
they press a key, I hear a split second of the DTMF signal
Hi all Sorry for the last post! Not enough sleep combined with inattention
caused me to reply to the wrong message.
Sean
-Original Message-
From: Anton Tinchev [mailto:[EMAIL PROTECTED]
Sent: Mon 2/23/2004 12:25 AM
To: [EMAIL PROTECTED]
I was able to resolve the issue... Me being stupid...
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Posted At: Sunday, February 22, 2004 7:45 PM
Posted To: Asterisk User Group
Conversation: Sip Register Fail - NAT
Subject:
Just a silly question, but can the asterisk box ping/contact the client?
(IE, is the routing correct on the * box?)
How can I get asterisk to indicate why it is ignoring SIP REGISTER requests?
P.S. - netstat shows that someone is listening on port 5060
On Sun, Feb 22, 2004 at 09:12:47PM -0800, [EMAIL PROTECTED] wrote:
Just a silly question, but can the asterisk box ping/contact the client?
(IE, is the routing correct on the * box?)
(just to clarify, I mean the natted IP address..)
___
60 matches
Mail list logo
