Hi, all
I have Polycom SP300 phones. My extension range is 1xx, so I added
corresponding entry to the digitmap.
By some reason this does not affect on-hook dialing. If I have phone
off-hook all is ok. dial extension 102 for example and it connects.
if phone is off-hooh, however, I have to
how to configure the g729 with 2 channels in iax.conf.
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I also experienced this problem and the first thing that really helped out was
changing the timing in the span line of the zaptel.conf. Change it to look
like this (see below) and see if it helps out. I got the error much less after
doing this and eventually got rid of the error completely
Maybe this is rather a hardware question, but I am posting it on this
list because the probability of someone else of you having tried this is
greater here than other places I can think of.
I have an ISDN card that is setup in NT mode using the zaphfc driver in
bristuff, and I got it working
hello
i am using ast-rad-acc.pl from portaone connected with
asterisk manager.
my (%cdr) = @_;
$cdr{'CALLERID'},
$cdr{'DNID'},
these are empty
why these two variables are not working on hangup
any comments
thanks
Kamran Ahamd
__
Do You
As far as I know, the dialplan autodialer only works when the phone is off
hook. This of course allows for nonstandard numbers to be dialed without
regard to the digitmap. I, for example have lots of *XX numbers like *69
and *82, but if I wanted to dial *8 for a pickup I just dial *8 and then
Hello,
I'm currently using OpenSER as REGISTER server and Asterisk for the call
routing. Do i need the OpenSER nathelper module if i want to offer
(mostly) automatic NAT traversal to my users or does Asterisk have the
same functionality?
It seems that the nathelper module should be able to
Hi, all
Can someone point me to a good resource on IAX?
I am trying to do two things at the moment:
1. I want to learn
2. I want to conenct MozPhone to my * (wiki does not have much on it)
3. I want to connect two * servers at different locations.
I have looked at asterisk wiki and dis not
Has anyone implemented call waiting on the server side for calls to SIP
phones? I.e. where only one call is delivered to the phone, and the
called party hears a tone for subsequent calls, and they can press a key
sequence to switch between them, the switching being done on Asterisk
rather than
Carl Andersson wrote:
Maybe this is rather a hardware question, but I am posting it on this
list because the probability of someone else of you having tried this
is greater here than other places I can think of.
I have an ISDN card that is setup in NT mode using the zaphfc driver
in
Hi,
What could cause:
Got SIP response 406 Not Acceptable back from 10.0.0.10
10.0.0.10 = Hardware FXS
And are there any probable solutions?
Regards,
Dave Walker
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On 16/07/05, Anton Krall [EMAIL PROTECTED] wrote:
Also, both asterisks have notransfer?yes and I get this
-- Attempting native bridge of IAX2/[EMAIL PROTECTED] and
IAX2/voipjet-9
Why? Seems its not taking the notransfer into account.
native bridge is not the same as transfer.
--
Thanks Peter.
Any other takers on the list on this one ?
Kindest regards
David Wilson
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Hi
As a general note: if you want to start a new thread, don't reply to an
existing message: Write a new message. Otherwise your message will
appear as a reply and be buried somewhere down a thread that nobody
cares about.
On Sat, Jul 16, 2005 at 08:27:56PM +1000, Rudolf Ladyzhenskii wrote:
Hi,
On Fri, Jul 15, 2005 at 09:52:19AM +0100, Angus Comber wrote:
Hello
I am trying to get Asterisk to work with the Junghanns Quad BRI ISDN card. I
am progressing slowly!
Problem I am now experiencing is as below.
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
How can I disable that native bridge stuff?
The scenario I have here is this.
The main asterisk is behind a nat firewall and is routing port 4569 to that
asterisk.
The remote asterisk is also behind a nat and firewall.
Both asterisk are connected thru an openvpn and they can see each other
I understand that the X100P card is matched to a 600 ohm impedance but the
UK BT phone system is not (I haven't been able to find much information on
the impedance of the UK system).
Has anyone come up with an easy way to match the impedance between the two
so the X100P can work in the UK?
Steve,
The X100P card works ok in UK (I have 3 at the moment). The only problem I
encountered with it was when I had my SKY box connected to the same line.
This caused random hangups.
Apart from that the card works ok and the UK callerid patch is fine for
detecting the BT ids.
I hope this
Hello,
I am following up on a previous mail of the same subject at
http://lists.digium.com/pipermail/asterisk-users/2005-June/110617.html
In a nutshell I have connected my asterisk behind a Siemens HICOM 118E
for a small call center application. The external PSTN calls will land
in HICOM 118E
Sure, I have more than 18 tunnels to manage here, and the only blocking
effects are thuse that I have volontary encoded .
;-)
I believe that Peter has missed something in the VPN parmeters themselves or
not correctly understood how are his IPtables onto this two IPSec secure
gateway...
Peter,
http://www.google.com/search?q=asterisk+iax
-Original Message-
From: Rudolf Ladyzhenskii [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 16, 2005 5:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Beginners question -- IAX
Hi, all
Vonage Softphone service works with Asterisk. Search this list for more
details.
-Original Message-
From: Michael Stearne [mailto:[EMAIL PROTECTED]
Sent: Friday, July 15, 2005 10:21 PM
Subject: Re: [Asterisk-Users] Vonage to IAX DID to IVR = Poor DTMF
Does Vonage work with
For our hotel application, we don't want to have to write 50 voicemail
entries, is there a way to do a voicemail macro in the same way as a
standard extension macro?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo
For our hospitality system, voicemail management is an issue. I looked
at vmail.cgi and it works for the user, but I need a higher level
management capabikity, i.e., flush all email from extensions 1XX
(Apartment1) when a guest checks out.
Is there anything like that or does anyone want to
Hi,
Can anyone please point me in a direction as to how to set up these 2
pci cards with AAH 1.1?
I have (am still) googling left, right center - but haven't found a
definitive guide yet.
The centos based setup lacks any of the tools I know (insmod, modprobe
) so it is time consuming
[EMAIL PROTECTED] takes only 15 minutes to install in a
Xeon 2.8. However downloading 700mb ISO file could
take all night.
But I guess that it worth it, because it is very easy
to manage, however I can not make my Cisco 7910 work.
--- Sergio Chersovani [EMAIL PROTECTED] wrote:
Javier Chia ha
Hey everybody,
I've been trying to recreate a paging unit that we have in house that
basically, when a user calls extension 44, it records their message.
When they hang-up, it plays a notification tone and then plays back the
message. I thought this should be easy, I have a sound card in
I have been trying to get this to work. I monitor the spool directory
and no call file is created. Am I missing something here?
My Config
Voicemail.conf
[general]
externnotify=/usr/local/bin/vm-notify.pl
[root@ root]# ls -lat /usr/local/bin/vm-notify.pl
-rwxrwxrwx1 root root
Asterisk Users,
I am reposting to the Asterisk-Users list what I saw
on the Asterisk-Biz list by Mr. Jeff Grammer,
GOD BLESS HIM!
I am in the Hamptons today, trying to whew a client,
and he took a look at my Level3 Partner pricing,
and laughed, as his rates were better than mine!!!
To top
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
This is where you have several sipura-841 SIP phones for
Hi,
My Asterisk CVS is apparently not doing much (other than keeping SIP
IAX2 registrations alive and doing some ZAP calls (without
echo-cancellation), but slowly the memory is filling up, so much so that
100m virtual memory is used up within 12 hours and I have to restart the
asterisk
My telco doesn't provide Disconnect Supervision or Polarity Change. So I
figured I have to detect hangups with busydetect=yes in zapata.conf.
I tested it. When the telco sends a busy tone * detects it and hangsup.
So far so good. The problem is the telco doesn't always send a busy
after
Here's t
link:
http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG
ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588
The bottom line is that they compare retail VOIP providers like Comcast
Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their
methodology seems sound.
Hiya,
I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports
updates. The port won't compile I just get this.
chan_zap.c: In function `pri_dchannel':
chan_zap.c:8391: error: structure has no member named `cause'
chan_zap.c:8886: error: structure has no member named
Did you do a make clean? I just, as in 1 hour ago, successfully
installed 1.0.9 using the port on FreeBSD.
Darren Wiebe
[EMAIL PROTECTED]
Mark Ackroyd wrote:
Hiya,
I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports
updates. The port won't compile I just get this.
Michael Graves wrote:
Here's t
link:
http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG
ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588
The bottom line is that they compare retail VOIP providers like Comcast
Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their
I agree with you but not 100% with them. An IP to Ip call on an ATA flat
out is better . Now don't even get me started about cellular. My Service
dosen't drop calls in the middle of conversations. VoIP is a notch
better than Cellular.
Michael Graves wrote:
Here's t
link:
7910 works fine wiz asterisk but you can not transfer calls, for that
reason I will sell mine if somebody is interested...
jl
2005/7/16, Javier Chia [EMAIL PROTECTED]:
[EMAIL PROTECTED] takes only 15 minutes to install in a
Xeon 2.8. However downloading 700mb ISO file could
take all night.
On Friday 15 July 2005 16:54, Peter Osborne wrote:
You can alway use google's cache. Use site:www.voip-info.org when
searching or type the full URL into google and click on the cached version.
Pete
On 15 July 2005 4:36 pm, Damon Estep wrote:
Does anyone have a mirror of this running?
Yes,
Hi the list :-)
Wondering why I can't bridge two X101P FXO cards to forward an external
call from a first X101P to another analog telephone outside my house
throught a seconf X101P.
[VACATION]
exten = s,1,Answer
exten = s,2,Dial(Zap/3/ww0161417888),120
exten = s,3,Voicemail(u1001)
exten =
Hi all,
on sourceforge.net I added the fixup release 0.5.4 of
chan_capi-cm driver.
The changes from 0.5.3 to 0.5.4 are:
- fixed 'group' setting according to Asterisk defaults.
- use SetCallerPres(prohib_not_screened) instead of CallingPres(32) for CLIR.
- full CallingPres support added.
- use
Good Day list,
Does
anyone know if it is possible to setup asterisk such that it passes DTMF Tones
through from One channel to the next transparently.
I
have a situation where asterisk is answering the phone on Channel 1 (first
channel of a PRI) and then bridges this call to
On Sat, 2005-07-16 at 10:12 -0700, Michael D Schelin wrote:
I agree with you but not 100% with them. An IP to Ip call on an ATA flat
out is better . Now don't even get me started about cellular. My Service
dosen't drop calls in the middle of conversations. VoIP is a notch
better than
1) Trying to create a browser based Click-to-Call feature for * that
appears to the user as a hands free call on Cisco 7960 phones (SIP). If I
use the Action: Originate function, the phone does not auto answer, but
rather rings and if answered initiates the call. If I manaually change
the line
Steve Gladden wrote:
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
This is where you have several
Hi,
for some weeks now I have been unable to make calls via my voicepulse
connect IAX account?
When I attempt the console looks like this:-
rt*CLI
-- Executing Dial(SIP/2008-cf55,
IAX2/NBhXX:[EMAIL PROTECTED]/12124565900) in new
stack
-- Called NBhX:[EMAIL PROTECTED]/12124565900
Hi:
I live in a country where calls from landline phone to a mobile phonesis more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to thepstn and mobile phones. I want to save money by transferingmobile calls througha
On Sat, 2005-07-16 at 11:55 -0700, chawki hammoud wrote:
Hi:
I live in a country where calls from landline phone to a mobile
phones is more expensive than mobile to mobile. I have FXO card
connected to the landline. All the calls from IAX goes through this
interface to the pstn and mobile
Hello
try to setup a gsm gateway
it will do what you want
best regards
Thierry
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de chawki
hammoudEnvoyé: samedi 16 juillet 2005
20:55À:
Asterisk-Users@lists.digium.comObjet: [Asterisk-Users]
Asterisk
Hi,
Anyone to know how to run an Outbound Proxy to solve the NAT
problem? I saw the FreeWorldDialup (FWD) are using a SER on port 5082. I have
tried to configure SER with nathelper/rtpproxy. Anyway I still can nothave
a callfromSIP UA behind a NAT but in same time it works perfect with
the
Generally speaking one works against one's own best interests when he
reminds the group that he has been posting on a topic repeatedly without
anyone answering.
What you are asking for is not reasonable; it's not the way Asterisk
works, and there is in my mind (and I'll bet in the minds of
Lists skrev:
On Friday 15 July 2005 16:54, Peter Osborne wrote:
You can alway use google's cache. Use site:www.voip-info.org when
searching or type the full URL into google and click on the cached version.
Pete
On 15 July 2005 4:36 pm, Damon Estep wrote:
Does anyone have a mirror of
Hi everyone,
I have a number of SIP registrations going fine, but am trying to get a new
provider going, and they have no sample Asterisk SIP config. They have been
helpful, but keep falling back to the way they think packets should be
flowing,
and I've been trying to figure out how the Asterisk
Try terminating using 50 ohm resistors as suggested by this guide:
http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html
in chapter 2.2 (Connect ISDN telephones to your ISDN card.)
Best regards,
Jan Snelders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote:
Hi everyone,
I have a number of SIP registrations going fine, but am trying to get a new
provider going, and they have no sample Asterisk SIP config. They have been
helpful, but keep falling back to the way they think packets should be
I think FWD is using Jasomi's SBC to tackle NAT issues.
On 7/16/05, Shady [EMAIL PROTECTED] wrote:
Hi,
Anyone to know how to run an Outbound Proxy to solve the NAT problem? I saw
the FreeWorldDialup (FWD) are using a SER on port 5082. I have tried to
configure SER with
Hi,
Quoting Michiel van Baak [EMAIL PROTECTED]:
On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote:
The error on the console is:
Jul 16 11:29:20 NOTICE[3361]:-- Registration for
'[EMAIL PROTECTED]'
timed out, trying again
Jul 16 11:29:21 WARNING[3361]: Forbidden - wrong password on
Hey Kevin,
I managed to resolve this error after a week of pulling out my hair.
Here is what I did to resolve the error and a link below for further assistance.
1 - If you are not using 2.6 kernel, upgrade.
2 - Check your span line in your zaptel.conf. You should be receiving
Chris Mason (Lists) wrote:
Steve Gladden wrote:
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
This is
On 17:01, Sat 16 Jul 05, [EMAIL PROTECTED] wrote:
Hi,
Quoting Michiel van Baak [EMAIL PROTECTED]:
On 16:32, Sat 16 Jul 05, [EMAIL PROTECTED] wrote:
The error on the console is:
Jul 16 11:29:20 NOTICE[3361]:-- Registration for
'[EMAIL PROTECTED]'
timed out, trying again
I have an OpenBSD 3.7 gateway. This gateway run Asterisk.
I have two windows box which use X-Lite softphone, and each box connect
to Asterisk using this softphone (X-Lite).
Asterisk use the following configuration :
/etc/asterisk/sip.conf
; Phone #1
[Phone1]
type=friend
host=dynamic
nat=yes
On Sat, 16 Jul 2005 10:10:29 -0700, Bruce Ferrell wrote:
Michael Graves wrote:
Here's t
link:
http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG
ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588
The bottom line is that they compare retail VOIP providers like Comcast
This is not a problem. I do this and a bit more. The IAX protocol helps
quite a bit to go thru the NAT.
--
Ted Serreyn Phone:262-432-0260 Fax:262-432-0232
Serreyn Network Services, LLChttp://www.serreyn.com/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Sat, 2005-07-16 at 17:05 -0500, Michael Graves wrote:
I agree with others who have chimed in that IP-to-IP calls can sound
better than PSTN calls. I have a co-worker who has a SipGate account in
the UK. Calls to him via SipGate go out through my FreeWorldDialup
account. They sound great. So
trixter http://www.0xdecafbad.com wrote:
On Sat, 2005-07-16 at 17:05 -0500, Michael Graves wrote:
I agree with others who have chimed in that IP-to-IP calls can sound
better than PSTN calls. I have a co-worker who has a SipGate account in
the UK. Calls to him via SipGate go out through my
On Sat, 2005-07-16 at 16:12 -0700, Bruce Ferrell wrote:
It's sometimes called comfort noise... As far as I'm aware, it's only
done in VoIP.
I spent 15 years working with digital switches/T1 channel banks. I
guess it might have been built in and I just didn't know about it, but
we were
Hi,
I would like to implement a inexpensive home PBX with Asterisk. I have
an internal ISDN bus with 6 ISDN phones. I now thought, I connect a
Fritz card to my Mehrgerateanschluss (Point-to-Multipoint) supplied by
my provider and a second Fritz card to the internal bus. Will this work?
I have tried to update my CVS build from 29-6-2005 with a new one.
but now when i start asterisk in verbose mode it crash after 1000+ lines of:
...
...
Jul 16 20:21:57 ERROR[23794] utils.c: warning negative
timestamp -257340.-252000
Jul 16 20:21:57 ERROR[23794] utils.c: warning negative
On 7/16/05, Asterisk Supporter [EMAIL PROTECTED] wrote:
1) Trying to create a browser based Click-to-Call feature for * that
appears to the user as a hands free call on Cisco 7960 phones (SIP). If I
use the Action: Originate function, the phone does not auto answer, but
rather rings and if
Just run somthing like this:
rm -R /var/spool/asterisk/vm/default/1xx/* (I think this should do,
otherwise something similiar will).
On 7/16/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
For our hospitality system, voicemail management is an issue. I looked
at vmail.cgi and it works for the
This together with the other post doesn't make sense. Anyhow, such a
macro will just do what the macro (err app) voicemail does. So why
invent the airplane when it was done already.
On 7/16/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
For our hotel application, we don't want to have to write
C F wrote:
Just run somthing like this:
rm -R /var/spool/asterisk/vm/default/1xx/* (I think this should do,
otherwise something similiar will).
Yeah, I'm sittng around waiting for guests to check out! No, this is a
job for php and an authenticated web page.
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent: zaterdag 16 juli 2005 12:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk
usingBristuff zaphfc in NT
Try terminating using 50 ohm resistors as suggested by this guide:
http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html
in chapter 2.2 (Connect ISDN telephones to your ISDN card.)
Best regards,
Jan Snelders
I did something along the lines of that, and it works great now.
But instead
On Sun, Jul 17, 2005 at 06:15:44AM +0200, Carl Andersson wrote:
My problem turned out to be a termination problem. When using zaphfc
together with other zap cards, it seems to be of importance in which
order the drivers are loaded as well - At least in my case it would only
work right if
Known issue. This was reverted later.
Check the thread on the mailing list
http://lists.digium.com/pipermail/asterisk-users/2005-July/116246.html
Thanks,
Erik
On 7/16/05, Walter Klomp [EMAIL PROTECTED] wrote:
Hi,
My Asterisk CVS is apparently not doing much (other than keeping SIP
IAX2
Hi, all
I have couple of SIP phones and they are in [from-sip] context.
I also have an IAX2 phone. I have put this one in [iax-user] context.
I want to make calls between SIP and IAX2 phones. If I put them all in same
context all is fine, however when they are in different contexts they will
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