On 2006-09-20 23:57:09 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said:
Hi Eric,
I'm confused on where I would put this?
I'm also confused on how this would help with external calls (which we
want to be g729) vs internal calls to
The problem is that most people aren't going to be able to answer this
question without trying it. Most voip providers (including Teliax)
advertise a rate to all New Zealand Mobile service providers, i.e. +64
2, not specifically +64 21xxx.
Note, I just tried a +6421 mobile number
Hi,
I am hating my ISP (comcast) and thinking about switching. One of my
options seems pretty good, but doesn't offer a static IP (maybe they
will for extra $).
Is anyone out there running an asterisk server via dynamic DNS and is
this a workable setup?
I know my remote ATA's are fine
Pan wrote:
I am trying to apply the Asterisk packetization patch found at
http://bugs.digium.com/view.php?id=5162
I have several versions of Asterisk, the most recent being 1.2.12.1,
but I can not successfully apply the patch.
Any suggestions on how to successfully apply this patch to a
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Perhaps you are tying to use wildcard destinations in your setup. This
does not scale.
Wildcard:
exten = 1234567,1,Dial(SIP/${EXTEN})
This does not scale.
Each extension should have it's own exten = line and Dial(... line.
On Thu, Sep 21, 2006 at 11:40:38PM -0500, Jordan Novak wrote:
I am in need of an additional x100p in one of my servers. It
already has a fully loaded tdm400p in it. I can't figure out
how to define the other one in zaptel.conf. Which one do I
define first, I am guessing it is dicated by the
Hi,
Is it possible to pick up a call that's in queue and pass it to an
agent directly. The use case is that some times some important calls
land up in queue which I need to pickup immediatly and pass it on to
an agent.
raj
___
--Bandwidth and
I'm trying to configure my asterisk server to detect fax on an outbound ZAP
call. The reason for this is that I have a bunch of interviewers in an
outbound callcentre who don't like listening to fax machines and I want to
be able to detect fax on the outbound leg before attempting to bridge the
Are you seeing any IRQ misses
Cat /proc/zap/1 and let us know.
You might be experiencing some interrupt conflict
.
M
-Original Message-
From: David Gagnon
[mailto:[EMAIL PROTECTED]
Sent: Friday, 22 September 2006
3:37 PM
To: 'Asterisk Users Mailing List -
On 09/20/06 15:06 Dinesh Nair said the following:
On 09/19/06 16:59 Steve Langstaff said the following:
I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124nbn=4
thanks for the link,
however,
Hi Robert,
I use mpg123 version 059r , Fedora Core 5, Shoutcast Server 1.9.7
The streaming bitrate is 56Kb/s mono
I have 3 Shoutcast Servers
2 servers over the Internet 128Kb/s / 24Kb/s for public listeners
1 special Shoutcast MOH server in the Asterisk Box
||
Hi list.
I have one quick question. Does Asterisk work with dual core processors in
version 1.2? Will it work with dual core processors in 1.4?
I'm planning to buy new machine for one installation and I have to decide will
I buy single or dual core processor.
--
Tomislav Parčina
Lama
I was afraid that may be the case - The issue I have with that approach is
how do you avoid manually mapping extensions to mac addresses in the
dialplan? Assuming I have a PRI with 100did and I want to use the last 4
digits of the DID as the internal extension, I want to use something like
Is there any (I prefer one port, but I could also buy two port) E1 PCI-Express
card?
As far as I can see, all Digim cards are PCI.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
How might you identify a mobile #? (assuming you refer to cellular phones)
Now that phone companies are allowing you to transfer your land line to a
mobile, it's no longer practical to use prefix blocking.
Where I worked, they just gave up and just restricted forwarding to long
distant
Hi all
I've read through the UPGRADE.txt file, but AFAIK it does not quite
discuss all the new stuff with 1.4. Neither the jitterbuffer nor the
packetization patch (#5162, if that ever made it into 1.4) are
mentioned. So, is there a document somewhere describing what's new in
asterisk?
Hi,
I'm going to install format_mp3 but I found other two choices, rawplayer
and madplay.
Anybody knows pros and cons?
TIA
Giorgio Incantalupo
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To UNSUBSCRIBE or
can please some one tell me where is what wrong.
iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del Lost % Drop OOO Kpkts
Jit Del Lost % Drop OOO Kpkts
IAX2/callaus-3
On Fri, Sep 22, 2006 at 05:17:03PM +1000, Mark Edwards wrote:
I'm trying to configure my asterisk server to detect fax on an outbound ZAP
call. The reason for this is that I have a bunch of interviewers in an
outbound callcentre who don't like listening to fax machines and I want to
be able to
Hi
Our VoIP provider complains we're sending INVITE retries too quickly. So I think I'm looking for an INVITEequivalent of registertimeout in sip.conf, but there doesn't seem to be one. Any suggestions?
David___
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Raphael Jacquot schrieb:
At some point in my dial plan, I need to find out the length of a sound
file in seconds (to weed out things that are way too short)
the record application doesn't seem to have any facilities to do that.
any ideas ?
i am wondering ... the voicemail app, does
I was thinking the same thing when reading the press release on sineapps
and writing a news article for asteriskguru.
I think this covers most of it:
- Generic Jitter Buffer
- t.38 passthrough
- Dial plan programming language (AEL v2)
- Asterisk can talk to googletalk and Jabber networks
-
Sorry, one other equipment query: does anyone know of an ATA with wireless
hardware which can act as a *client* to another wireless network?
The Linksys units have an integrated wireless access point, but I want
something which will work as a client onto an existing wireless network - so
you can
Dear list
which hardware solution would you suggest for connecting 60 analog
phones to asterisk ?
thank you very much
.mike
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi,
I'm try to setup a dial plan in freepbx to work properly with ENUM lookups.
However, the only example I can find that works in the UK is somewhat complex.
(http://www.voipuser.org/forum_topic_6651.html)
Basically, it has 3 outbound routes (local, national, internation) to strip
certain
quad port T1 card
3 channel banks.
Zoa
mike wrote:
Dear list
which hardware solution would you suggest for connecting 60 analog
phones to asterisk ?
thank you very much
.mike
___
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Hi,Where can you find error codes tables ?I googled to find that but could find anything.I guess there is something somewhere in source files showing for each error code, a text to display but is there also somewhere suggestions that programmers might leave for systems administators telling them
can please some one tell me where is what wrong.iax2 show netstats LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts
Jit Del Lost % Drop OOO KpktsIAX2/callaus-3 265 -1 0 -1 -1 0 -1 00 40 0 0
Zoa wrote:
quad port T1 card
3 channel banks.
If expandability isn't a big factor but cost is a dual port E1 card and
2 channel banks. This will get 60 exactly, not 64 tho.
[EMAIL PROTECTED] :o)
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2 of xorcom's astribank-32 (http://www.xorcom.com/astribank/features-32.html) On 9/23/06, mike
[EMAIL PROTECTED] wrote:Dear listwhich hardware solution would you suggest for connecting 60 analog
phones to asterisk ?thank you very much.mike
___
On Thu, 21 Sep 2006, Nick Couchman wrote:
When I try to set the port to 636 in the res_ldap.conf file, I get bind
errors (Can't contact server...). I imagine this is an issue with
certificates and trust, but I'm not exactly sure where I need to put my
CA certificate in order to make the ldap
WB == Wes Baehr [EMAIL PROTECTED] writes:
WB Use chanisavail to check if one or both phones is busy - if either
WB is busy, redirect to voicemail/busy/whatever.
Unfortunately chanisavail does not seem to actually ask the phone
whether it is busy. When I call it on SIP/somephone, AVAILSTATUS
On sep/22/2006, mike wrote:
Dear list
which hardware solution would you suggest for connecting 60 analog
phones to asterisk ?
Maybe a Top-Gate SIP gateway. It supports 16, 24 or 48 FXS ports.
You don't need T1 or E1 extra in the Asterisk machine, only one ethernet
card.
--
Paco
Hi all,
I was deploying Realtime Extensions when I realised that Realtime
Asterisk yet doesn't support ex-girlfriend logic, what made me abandon
that implementation!
Does Asterisk 1.4 going to support that feature?
Regards,
Ricardo.
___
--Bandwidth
On Fri, Sep 22, 2006 at 03:09:47PM +0530, Arun Kumar wrote:
can please some one tell me where is what wrong.
iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del Lost % Drop
Sure.
Agents are logged individually into queues and can therefore work offhook.
My application issues an 'originate' via AMI from the queue to the
destination number. When the call is answered it is bridged and connects the
Agent to the destination party.
The desired effect would be that when
Hi Users, I'm developing the Voicemail, By flat files I made it, But now I need to do in MySql Databases,In res_mysql.conf and cdr_mysql.conf I given the Database entitesWhile I'm reloading the asterisk server
I have arrrived below one message,Can any one tell what this messages
Martin Joseph wrote:
Hi,
I am hating my ISP (comcast) and thinking about switching. One of my
options seems pretty good, but doesn't offer a static IP (maybe they
will for extra $).
Is anyone out there running an asterisk server via dynamic DNS and is
this a workable setup?
I know my
Make sure you buy it with PCI slots.
I overlooked it and the default was PCI-Express.
This was for a file server and when I went to put in the SCSI controller, oh,
sh*@^$*@$.
--
--
Steven
http://www.glimasoutheast.org
Ryan Amos [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
It
Hi all,
Any polycom phone v1.6 IP301 references? I had purchase
three new phone and I cant connect them into Asterisk 1.2.11.
I do appreciate if some one can point me how and where ?
Thank you
___
--Bandwidth and Colocation
(AstATN) wrote:
Hi all,
Any polycom phone v1.6 IP301 references? I had purchase three new
phone and I cant connect them into Asterisk 1.2.11.
I do appreciate if some one can point me how and where ?
http://www.voip-info.org/wiki-Polycom+Phones
Doug
--
Ben Franklin quote:
Those
Hi Mike,
It's a while since I did this one myself, but I was doing the exact same
thing when using voipbuster (or whichever of it's sisters services I was
using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1}
HTH,
Mat
-Original Message-
From: [EMAIL
Craig Guy wrote:
I was afraid that may be the case - The issue I have with that approach
is how do you avoid manually mapping extensions to mac addresses in the
dialplan? Assuming I have a PRI with 100did and I want to use the last
4 digits of the DID as the internal extension, I want to use
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Perhaps you are tying to use wildcard destinations in your setup. This
does not scale.
Wildcard:
exten = 1234567,1,Dial(SIP/${EXTEN})
This does not scale.
Each extension should have it's own exten = line and
Group
Any known problems
with Asterisk SVN-trunk-r43322 and Asterisk Recording Interface or the vmail.cgi
script?
I'm unable to see
voicemailsvia the web even though the MWI is flashing and if I look in
/var/spool/asterisk/voicemail/default/100/INBOX
I do see msg files
in that folder.
I have Asterisk 1.2.12.1 with Realtime Static configuration.Ramdonly when I reload by the Cli command, It crash...I have queues.conf, agents.conf and extensions.conf in the ast_config table (Postgres database) and connect with Asterisk by unixODBC.
Any idea?
There are a couple more that I have run across.- Shared line Apperance support- Users.conf file for simple config of users and devices- follow me application and conf file- Asterisk Builtin mini-HTTP server
On 9/22/06, Zoa [EMAIL PROTECTED] wrote:
I was thinking the same thing when reading the
Here's the dialplan I am using at the moment.
[dialer-test-2]
exten = _X.,1,Set(TIMEOUT(resposnse)=10)
exten = _X.,n,dial(Zap/g1/${EXTEN},60,M(detect-fax^1^2))
exten = _X.,n,noop(back from dial in dialer-test-2)
exten = t,1,noop(timeout)
[macro-detect-fax]
exten = s,1,noop(detecting fax)
exten =
On 9/21/06, Denis Galvão [EMAIL PROTECTED] wrote:
bweschke, is there any news about using astdb to store the numbers to
be dialed?
This is related to this note on bug http://bugs.digium.com/
bug_view_advanced_page.php?bug_id=5574:
(0035684)
shmaltz - reporter
11-02-05 15:01
Also thinking
On Friday 22 September 2006 13:36, Mat Stace wrote:
It's a while since I did this one myself, but I was doing the exact same
thing when using voipbuster (or whichever of it's sisters services I was
using at the time).
I'm thinking that in the dial command you want
+44{EXTEN:1}
Thanks, but
I would like to tie outbound calls from specific extensions to specific
zap channels...I have multiple clients in an executive suite and would
like to be able to tie lets say extension 1234 to Zap Channels 1 and 2
and extension 5678 to channels 3 and 4 and so on...
This so that their caller ID
On 9/21/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Sep 21, 2006 at 08:41:37AM -0700, Elpidio Ramos wrote:
Ok, after requesting information to digium (no answer yet) and being informed
that asterisk-dev is *NOT* a support hot line, I am trying in this list to see if
someone has
Hi dudes
I read a lot
of callback tutorials but I failed to make it work, can any one tell me how to
do it in a brief attached with command line, and I will be thanks full .
Regards
*
No employee or agent is authorized to conclude
Can I get the hint status from the dialplan?
I am intending to add lit buttons for the parking slots.
currently I am using 1.2.11 with 1 parking button and several pickup
buttons (speed dials to the parking slots)
since 1.4 allows park() to specify a parking slot, I figure that I can
remove
BJ, I believe that asteiskdb is before realtime. It does not give the
same functionality, since asterisk apps can only update asteriskdb
thru the DP, and built in commands.
There was some discussion around this feature in app_followme in the
IRC chat rooms and it was decided that for at least
John Marvin wrote:
Earle Clubb wrote:
Hello,
I'm trying to play an audio file to a phone an arbitrary number of
times. The audio is a five-second segment of a sine wave. I need
this to be played repeatedly without gaps between playbacks. I've
tried doing this in the dial plan, e.g.:
I can have a go at explaining.
I've had a quick dig through my extensions.conf, and I've got it in an
outgoing sipgate dial command.
exten = _0.,1,Dial(SIP/+44${EXTEN:[EMAIL PROTECTED],30,t)
What it does is in the dial command, it sends +44, then the extension which
you dialled, minus the first
Khaled Chehab wrote:
Hi dudes
I read a lot of callback tutorials but I failed to make it work, can
any one tell me how to do it in a brief attached with command line,
and I will be thanks full ..
You will need to give us an example of what you want it to do before
that can be done.
I am using the latest asterisk 1.2.12 etc...
I have uniden UIP-200 phones, Cisco 7960 phones, Cisco 7912 phones,
Cisco 7940 phones.
It seems like once in a great while (perhaps every other month)
All of these phones lock up and have to be rebooted.
Are others experiencing this?
The UIP-200
On Wed, 2006-09-13 at 10:57 +1000, Paul Hales wrote:
From memory, it canalmost
I used quite a few Grandstreams on a job a while ago, and my memory says
that they will do alpha if you are lucky. If not, you get rubbish. My
memory also tells me that UPPER CASE worked better than mixed
Hello,
On 9/21/06, Lee Howard [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
Everything is fine when caller use ECM but when ECM isn't in use I
often got unusable incoming faxes (much often that it should be). But
when I switch back to fax machine that receive faxes perfectly (at
least no
Where are yours ?
Mark Phillips wrote:
Yet another set?
I get about 50 downloads a week for mine.
Mark
On Tue, 2006-06-06 at 22:27 +0100, Steve Kennedy wrote:
I'd like to announce that the UK Male English Voices are now up on
http://www.tel.net/
There's a complete set of base sounds
Steve Kennedy wrote:
I'd like to announce that the UK Male English Voices are now up on
http://www.tel.net/
There's a complete set of base sounds and additional sounds (it should
be complete compared to current Asterisk and Asterisk-Sounds-1.2.1).
There's also a set with the word 'pound'
On Mon, 2006-09-11 at 21:14 +0200, Remco Barendse wrote:
Hi list!
I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the
TEI check request message were I was getting errors.
Concerned about that I switched to plain vanilla bristuff.
Now everything *seems* to be
On 9/22/06, C F [EMAIL PROTECTED] wrote:
BJ, I believe that asteiskdb is before realtime. It does not give the
same functionality, since asterisk apps can only update asteriskdb
thru the DP, and built in commands.
There was some discussion around this feature in app_followme in the
IRC chat
I have done looping playback and never experienced significant gaps.
Earle Clubb wrote:
John Marvin wrote:
Earle Clubb wrote:
Hello,
I'm trying to play an audio file to a phone an arbitrary number of
times. The audio is a five-second segment of a sine wave. I need
this to be played
Just upgraded my * box to 1.2 and don't seem to be able to get MWI working.
Worked with my previous installation. My conf files are the same ( except
for a few 1.2 changes ). I've tried:
In sip.conf
fromuser=Anyname
fromdomain=my * ip
vmexten=7000
in extensions.conf
exten =
21 Sep 2006 12:15:01 +0200, Benny Amorsen [EMAIL PROTECTED]:
I have considered various ways to solve this. One is to make a queue,and only allow one caller in the queue. As far as I can see this won'twork, at least not when I am busy because I did an outgoing call.Another way is to use GROUP() to
See:
http://www.voip-info.org/wiki/view/Asterisk+SS7
Jorge Mendoza
Jay R. Ashworth wrote:
On Thu, Sep 21, 2006 at 05:15:36PM -0600, MF wrote:
Hi I need to connect at least 2 (and 2 more in the future) links to a
switch via SS7,
does anyone knows if this can be done with Digium
Trying again
Has anyone an explanation why this error happens?
Only hear my echo and not the other side anymore...
and the other side can't hear me...
Version asterisk 1.2.9
-- Executing Macro(SIP/1001-9c43, stdexten|1010|SIP/1010) in
new stack
-- Executing Dial(SIP/1001-9c43,
On 9/21/06, Denis Galvão [EMAIL PROTECTED] wrote:
bweschke, is there any news about using astdb to store the numbers to
be dialed?
This is related to this note on bug http://bugs.digium.com/
bug_view_advanced_page.php?bug_id=5574:
(0035684)
shmaltz - reporter
11-02-05 15:01
Also thinking
Giorgio Incantalupo wrote:
Hi,
I'm going to install format_mp3 but I found other two choices,
rawplayer and madplay.
Anybody knows pros and cons?
TIA
Giorgio Incantalupo
___
I used madplay for a month. It crashed once a week, taking asterisk
down
mike wrote:
Dear list
which hardware solution would you suggest for connecting 60 analog
phones to asterisk ?
thank you very much
.mike
Depends on current and future needs. I like the Quintum Tenor AX.
___
--Bandwidth and Colocation provided
http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+channelsOn 22/09/06,
Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, Sep 21, 2006 at 05:15:36PM -0600, MF wrote: Hi I need to connect at least 2(and 2 more in the future) links to a switch via SS7, does anyone knows if this can be done
Hi Tzafrir,
I prefer to use safe_asterisk even if it is not robustI have never
had crashes problems until today.
What I want is a little script that sends me a mail when something
happens and safe_asterisk seems to do it (I hope).
The only problem as I told you is having two safe_asterisk
I seen something in the bug tracker and svn about SMDI. Not sure if it was included it 1.4 though. Would be interested if anyone knows if this will work with nortel system (option 11 in particular).
On 9/22/06, Bruce Reeves [EMAIL PROTECTED] wrote:
There are a couple more that I have run across.-
Hi everybody,
Is there any significant difference between using
Macro(dialout-trunk,1,${EXTEN}) and Dial(Zap/g1/${EXTEN})? If so, what
are the differences?
I am not using freePBX, or any variant of it, but want the
functionallity of dialout-trunk. If I define the trunk in zapata.conf,
will
Eric ManxPower Wieling wrote:
I have done looping playback and never experienced significant gaps.
Can you give me an example of what worked for you? Did the files
contain tones or voice?
Earle
___
--Bandwidth and Colocation provided by
Hi list,
Does anyone knows whether Asterisk is able to talk to MSN peers or not,
and if yes to what extend? text-only, audio, video?
Thanks
Yoann
___
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asterisk-users mailing list
To UNSUBSCRIBE or
I have an incoming call from pastn number ,the system with deliver it from
e1 .
So I want to close the line an call him .(callback)
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Friday, September 22, 2006 4:49 PM
To: Asterisk
[EMAIL PROTECTED] wrote:
Just upgraded my * box to 1.2 and don't seem to be able to get MWI working.
Worked with my previous installation. My conf files are the same ( except
for a few 1.2 changes ). I've tried:
In sip.conf
fromuser=Anyname
fromdomain=my * ip
vmexten=7000
Are you missing
Hi all,
Can anyone help me... i need to display the cost of a call during a
conversation on a sip or iax phone.
I see on voip-info that some snom phone support sendtext application,
but i don't know how to update the message with the cost on the phone
during the conversation.
Every suggestion
Can you get an Ethereal trace that captures the RTP streams going
to/from Asterisk? If so, you might look for SSRCs changing mid-stream.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Richard Klingler
Sent: 22 September 2006 15:22
To: Asterisk
b'coz I have same setup at other client is working fine no problem.On 9/22/06, Tzafrir Cohen [EMAIL PROTECTED]
wrote:On Fri, Sep 22, 2006 at 03:09:47PM +0530, Arun Kumar wrote: can please some one tell me where is what wrong.
iax2 show netstats LOCAL -
Maybe a Top-Gate SIP gateway. It supports 16, 24 or 48 FXS ports.
You don't need T1 or E1 extra in the Asterisk machine, only one ethernet card.
thanks very much to everyone for the comments and the suggestions !
___
--Bandwidth and Colocation
On Thu, 2006-09-21 at 23:17 -0700, Martin Joseph wrote:
Hi,
I am hating my ISP (comcast) and thinking about switching. One of my
options seems pretty good, but doesn't offer a static IP (maybe they
will for extra $).
Is anyone out there running an asterisk server via dynamic DNS and is
Sorry then, I didn't know that, since I don't use realtime. I don't
see any reason to introduce another point of failure for a setup that
doesn't absolutely need realtime (like a cluster setup). I think my
point is still valid, that asteriskdb comes before realtime. Please
anybody outthere give
On Fri, 2006-09-22 at 12:35 +0300, Zoa wrote:
quad port T1 card
3 channel banks.
Zoa
Or 2 Astribank-32 () units that connect to the USB port on your
server.
--
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161
Eric ManxPower Wieling wrote:
Earle Clubb wrote:
Hello,
I'm trying to play an audio file to a phone an arbitrary number of
times. The audio is a five-second segment of a sine wave. I need
this to be played repeatedly without gaps between playbacks. I've
tried doing this in the dial plan,
Hi,
I am using Asterisk 1.2 with internal isdn phones connected via a hfcpci
card in nt-mode with misdn. Bridging calls from the internal hfcpci via
a avmfritz card (also chan_misdn) to the PSTN works flawlessly. However
when I use a sip channel to route the outgoing call via voipstunt, it
On Fri, 22 Sep 2006, Conrad Wood wrote:
On Mon, 2006-09-11 at 21:14 +0200, Remco Barendse wrote:
Hi list!
I was using bristuff-0.3.0-PRE-1s with florz patch but where normally the
TEI check request message were I was getting errors.
Concerned about that I switched to plain
On Thu 21 Sep 2006 Nick Couchman wrote:
When I try to set the port to 636 in the res_ldap.conf file I get bind
errors Cant contact server I imagine this is an issue with
certificates and trust but Im not exactly sure where I
Just spitballing:
1. Execute a macro in the dial command to spawn an AGI that would return
it's PID to Asterisk and accept the IP address or SIP address of the phone
as an argument. Call the variable, say, ${INCREMENTCOSTPID}
2. The AGI would store call cost variable plus the increment. It would
I am trying to get a SNOM 320 working with Asierisk.
It does register and I can make outbound calls. But it
would not take inbound calls. This is what I get;
-- Executing Dial(Zap/2-1, SIP/102|20|Tt) in
new stack
-- Called 102
-- Got SIP response 404 Not Found back from
192.168.1.105
On Sep 21, 2006, at 6:15 PM, MF wrote:
Hi I need to connect at least 2 (and 2 more in the future) links to
a switch via SS7,
does anyone knows if this can be done with Digium cards?
if not, which box could I use to convert from SS7 to isdn,
(could anyone please recomend one of these
All my current Asterisk 1.2.12.1 are running on UnixODBC realtime. I just
downloaded Asterisk 1.4 beta release this morning and but having problem to
compile asterisk with res_odbc on a new server. Have anyone experience this
yet and/or hint for me?
UnixODBC, UnixODBC-devel and postgresql-odbc
Dear All, thanks for the help on the TDM2400P. I have resolved the issue. I
isolated the problem and ended up finding out it was the Polycom phone that
had a problem. Those phones have spectacular quality but they are way too
complicated to setup. Also, it's absurd Polycom only supplies you with
Hello list!
Before I tried the new Asterisk 1.4-beta2 I thought I'd try the 1.4 Spanish
sounds on 1.2. When I go to voicemail to get messages it immediately hangs
up. Debug shows a missing vm-youhaveno sound file.
I took a look at the Asterisk 1.4-beta2 app_voicemail.c and it is still
looking
Khaled Chehab wrote:
I have an incoming call from pastn number ,the system with deliver it from
e1 .
So I want to close the line an call him .(callback)
This can be done several ways. The receiving operator can pass the call
to a special extension that would either ask for a callback
On 18:22, Fri 22 Sep 06, Remco Barendse wrote:
It seems that development on bristuff is stalling a bit, maybe because
Asterisk is working on native support of MISDN.
Hmm,
Will the quad/octobri and gsm cards be supported by MISDN ?
I think not.
I worked with the cheap HFC-pci bri cards but
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