I have an application where calls come into an *box from a DID
provider, and may be transferred to a meetme conference on another
*box (the call is released by the first *box after transfer).
These are ulaw IAX channel calls, and if the source is from a Verizon
or Nextel mobile phone to
Is anyone having issues with audio being passed inbound via Teliax?
Trying to isolate an issue here.
Thx,
-Rob.
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To UNSUBSCRIBE or
That is not going to work. Asterisk shouldn't be behind a NAT to get
registration of boxes behind NAT.
I've done it, and it works. It is not a great situation though because
of the provisioning problem. Specifically, an IAX device behind NAT
has no way of getting its provisioning out of the
hello austin
how to install perl module
i m following
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
and i did
sudo perl -MCPAN -e shell;
install Config::IniFiles
install Crypt::CBC
install Crypt::DES
install Authen::Radius
any other help full link i m new to perl
JD
I use Nagios to monitor lines. I use the check_asterisk script that you'll find floating around the place. I connect via the mgmt interface. Added to
nagios is nagiosgraph. This keeps historical RRD graphs of my line usage.
==
Rod Bacon
Empowered
Remco Barende ha scritto:
Does this version of chan_sccp replace the version at sourceforge or
is this Yet Another Fork(tm) :)
It's a fork.
Sergio
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On Wed, Jul 06, 2005 at 11:32:11PM -0400, Carlos Alperin wrote:
That sound like the Spanish TV Show. Is a similar of MRTG?
If that is the case, the problem is the SNMP module for Asterisk.
Why use snmp? you don't weant to minitor asterisk's snmp. You want to
monitor Asterisk. Either poll
Got a few and 8line one running good,got some compatibility problems
with some mother boards once but that was it
On Wed, 2005-07-06 at 16:08 -0300, Bartosz Jozwiak wrote:
Hello,
Is anybody there using quadBRI form Junghanns.net with Asterisk ?
I would like to order that card but first would
I had quite a lot of experience with it ... it works fine,
the only problem I got was that I couldn't transmit fax (data) calls
through it reliably ... although this was some time ago, so it
is possible that the kernel modules for them improved lately.
Ivan
Hello,
Is anybody there using quadBRI
In article [EMAIL PROTECTED],
Howard Ratzlaff [EMAIL PROTECTED] wrote:
I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core
3). Everythng was testing out and the configuration was working. After
running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel
cannot
How can I change:
Authorization: Digest username=70501956, realm=taraba.net,
algorithm=MD5, uri=sip:[EMAIL PROTECTED],
nonce=42ccd58240bd61c429ab1d2479d00209867a16a0,response=02fe9acd0bcb5f1866854b85439aebeb,
opaque=
to be:
Proxy-Authorization: Digest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
IM.Nobody
Sent: Wednesday, 6 July 2005 11:51 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] DECT VoIP Gateway
Hi all,
Just want to share with all of you a new hot DECT
or ci$co 7940 - features same as 7960, but only with two lines, instead
of six, but significantly cheaper than 7960...
PJ
Glenn Powers wrote:
Cisco 7960's work well and are highly recommended by many people,
including myself. They have the qualities you list.
cheers,
glenn
Hi,
Have anyone succesfully configured wifi roaming using Senao Wifi phone
model SI-680H?
If yes, please let me know your phone's firmware version and your configuration.
Thank you.
-eddie-
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If you are end user, there will be problem with direct communication
with ci$co, because ci$co standard way to solve problem is via ci$co
partner/reseller that sell the phone to you :-(
PJ
Andres Maduro wrote:
Hi,
I recently purchased from a friend 2 Cisco 7905G for testing them with
Ivan Meic (Vox Mundi) wrote:
I had quite a lot of experience with it ... it works fine,
the only problem I got was that I couldn't transmit fax (data) calls
through it reliably ... although this was some time ago, so it
is possible that the kernel modules for them improved lately.
I can
snacktime wrote:
On 7/6/05, Gundemarie Scholz [EMAIL PROTECTED] wrote:
Following the instructions on voip-ip.org I have implemented
Realtime with MySQL for my Asterisk server. The individual extension
configuration is managed in a table called extensions.
Still I have to keep some data
hi,
we are currently planning are large site which will migrate from an old
siemens hicom pbx to asterisk.
the customer is currently using a paging system (small receivers which
display a callback number and a base station (transmitter) with several
antennas at the site)
the problem is,
hi,
we are currently planning are large site which will migrate from an old
siemens hicom pbx to asterisk.
it will be a slow migration, the asterisk server will be inserted
between the telco E1 and the hicom. new phones will be sip ones.
the customer has several fax machines and analog
http://www.broad-tel.com/products/wireless.php
On 7/7/05, Ola Lidholm [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
IM.Nobody
Sent: Wednesday, 6 July 2005 11:51 PM
To: Asterisk-Users@lists.digium.com
howdy,
the problems with data and fax calls were mainly caused by asterisk,
e.g. echo cancelation always on, failed native bridging, gains,
Since bristuff 0.2.0-RC8e those issues have been solved. We have quite
a few customers running loads of ISDN data calls between their
locations without
At 15:21 06/07/2005 +0200, Tobias Wolf wrote:
Hi,
i was successful in compiling app_conference and setting up an conference
was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from inside
the conference. So, if i dialed into an conference i want to be able to
Vahan Yerkanian wrote:
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
So we're are waiting the free g729 codec for Europe now ...
___
Asterisk-Users mailing
Hi!
- could the T1 channelbanks be connected to a TE405P with two channels
in E1 mode (telco and hicom pbx) and two channels to the channel banks
(i think yes, but just to be shure)?
Yes - no problem.
- will the faxmachines work (56kpbs-64kbps)? is asterisk translating
this (btw. how do
Klaus,
Can the data transmission work reliably now between
an incoming PRI line (Digium TE405P) and outgoing BRI line (QuadBRI) ?
Ivan
the problems with data and fax calls were mainly caused by asterisk,
e.g. echo cancelation always on, failed native bridging, gains,
Since bristuff
Ivan,
as long as you use BRIstuff it will work fine with any zaptel hardware,
even with Digium or Sangoma.
best regards
Klaus
--
Klaus-Peter Junghanns
Am Donnerstag, den 07.07.2005, 12:25 +0200 schrieb Ivan Meic (Vox
Mundi):
Klaus,
Can the data transmission work reliably now between
an
On Thu, 7 Jul 2005, Anton Tinchev wrote:
Vahan Yerkanian wrote:
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
So we're are waiting the free g729 codec for Europe now ...
No need for
On Thu, 2005-07-07 at 11:21 +0200, Frank Sautter wrote:
[snip]
the problem is, that the currently operative base station uses 4 ISDN
BRI interfaces. But the protocol is old germany 1TR6 (and not EuroISDN).
Did you try contacting the vendor of the base stations to see if they
have a EuroISDN
Hello list,
I'm pretty new to Asterisk, but so far I managed to setup the server, added a
couple of mISDN channels (one TE, one NT), connected to a VOIP provider and
called out to the World :)
Now I started to play with codecs because I wanted to try the sound quality
of each of them, and
Is anyone having issues with audio being passed inbound via Teliax?
Trying to isolate an issue here.
Nope, works fine here with cvs-head from about a week ago.
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Hi,
I've oh323 chan installed and working to make calls from SIP to H323
devices. The problem is can no hear sound with the H323 device. I think
this is some related with codecs o nat, because the H323 have one public
IP from a different subnet from the asterisk box.
If I use netmeeting in
I am new to asterisk and have a few architecture questions.
I currently have a 3MB bonded T1 running our network and was wondering if
Asterisk be connected to the existing network and bonded T1 (which also
includes normal day to day network traffic), or do I have to dedicate a new
T1 to
Hi
I have a problem with the queues on Asterisk. The setup is [EMAIL PROTECTED]
v1.0 with Asterisk 1.0.7.
I have 1 queue (4500) set up, with leastrecent strategy. There are no
agents configured in this queue.
Agents log in by dialing 4500* on their phones. All incoming calls are sent
to the
You can try to open up port for SIP 5060udp and RTP 10-2udp...
(default setting) to your asterisk box. You will also have to specify
that your extensions are nat=yes your externip=xxx.xxx.xxx.xxx (in
SIP.conf) so that the SDP protocol will write the public IP and port
translations for RTP
Our BIGGEST problem is that every single one of the 80 phones are on a
direct connect T1. All are on qualify=1 yet sometimes we get 'TOO
LAGGED'. HTF can you get that kind of lag on a dedicated, direct
conected T1?
Sounds more like a lost packet rather than lag...
Try a ping -c 1000
Hello,
Is it possible for an mISDN channel to transfer a call to a new phone,
instead of opening a new channel to connect it? I have a couple of isdn
phones connected to Asterisk; transferring a call will open a *second* isdn
channel instead of connecting the two ISDN phones directly.
Best
Hi all and specially Steve...
Im using CVS HEAD with spandsp FAX solution.
Im getting this erros when starting Asterisk:
Jul 7 10:29:45 VERBOSE[31091] logger.c: [app_txfax.so]Jul 7 10:29:45
WARNING[31091] loader.c: /usr/lib/asterisk/modules/app_txfax.so: undefined
symbol:
Is it possible to use G729 on asterisk without the license?
It is to connect devices which use the codec to termination providers in a phone
card application.
Will decoding the DTMF tones from the caller require G729 processing?
Blake Krone wrote:
Hello all, I HAD video working before I upgraded to 1.08 (latest
stable with Gentoo) and now it won't work. I just see noise bars and
not the video. I know the camera works as I can use it in other
programs such as AIM Yahoo.
Which codec are you using for video in the
I would like to check the status of my PRI's (I believe that should include
ZAP), IAX (between Asterisk boxes) and SIP channels.
The reason for choose MRTG was because they track historic using RRDB in a
very good way.
The reason for snmp, was that there was a snmp module developed for the
anybody recommend a supplier in the UK for a pri/isdn30 line (other than BT)
thanx very much
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
Hi
I am currently using gnugk (www.gnugk.org) as a H.323
gatekeeper/proxy, mostly for our video conferencing devices.
What I would now like to do is add ISDN functionality to this and make
our H.323 gatekeeper box also function as an ISDN --- H.323 gateway
so that H.323 endpoints can call
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Howard Ratzlaff [EMAIL PROTECTED] wrote:
I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core
3). Everythng was testing out and the configuration was working. After
running YUM update, kernel 2.6.11-1.35_FC3smp was
Richard Malcolm-Smith wrote:
Is it just me that sees the post above as spam?
If we (tinw) even consider buying stuph from spammers, then we are
encouraging them in their sociopathic behavior, and as a consequence they
will do more spamming.
What is the consensus here?
It is a product
Hi,
If you are terminating the call from/to a T1/E1 card or modifying the
call in anyway e.g. playing IVR prompts not just voice in - voice out,
you will require the codec.
Regards,
Sahil Gupta
VoiceValley
On Thu, 7 Jul 2005, Obelix wrote:
Is it possible to use G729 on asterisk without
I am trying to get * to use proxy-auth when dialing out,
to mimc what x-lite does when force proxy is set to
yes. Is there any options that can be set to do this?
This particular sip provider does not support
the username:[EMAIL PROTECTED]/number for
Dialing, so I connect as a peer. But it seems
Title: Asterisk/Grandstream Budgetone disconnect issue
I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the
Yes, I've read that. Ztmonitor is simply a very _basic_ tool that provides
you with a little bit of feedback to adjust the rxgain and txgain
settings to something relatively close to what the human ear considers
reasonable audio levels.
The tool cannot detect or determine what settings
I do have a problem see my post few yesterday with subject:
Incoming 800-number over IAX
--
#Joseph
On Wed, 2005-07-06 at 23:05 -0700, Robert Goodyear wrote:
Is anyone having issues with audio being passed inbound via Teliax?
Trying to isolate an issue here.
Thx,
-Rob.
Obelix wrote:
Is it possible to use G729 on asterisk without the license?
Yes, as is clearly documented on the wiki :-)
Will decoding the DTMF tones from the caller require G729 processing?
No, because you cannot use inband DTMF with G.729 anyway. Since you will
need to be using
Hi,
I'm trying to send a fax from a Zap channel to another Zap channel and I
can't - below the logs. How could I solve this? Thanks!
In RECEIVER:
-- Accepting call from '' to '1980' on channel 0/2, span 1
Urgent handler
-- Executing Answer(Zap/2-1, ) in new stack
Urgent handler
Urgent
You can try to open up port for SIP 5060udp and RTP 10-2udp...
(default setting) to your asterisk box. You will also have to specify
that your extensions are nat=yes your externip=xxx.xxx.xxx.xxx (in
SIP.conf) so that the SDP protocol will write the public IP and port
translations
I would like to know if it is possible to slow down the dialing process in
asterisk.
I have 4 of my 8 phone lines that are VoDSL. When we try and dial out these
4 VoDSL Lines, the number is often miss dialed, or incomplete. I added a
wait before Asterisk tries to dial the whole number, but that
Just an FYI guys we have some of the leading open source developers
and projects going to speak/showcase at Cluecon.
These include:
Mark Spencer - Asterisk
Bob Andreasen - SIPFoundry
Craig Southeren - OpenH323
David Sugar - Bayonne
This should be an exciting event for all. Register Today!
Is there a way to log SIP response codes without enabling verbose
logging? Reason being is that from time to time I see a call fail on
our primary provider and roll-over to our backup providers. If I
happen to catch it on the console I can see the code 484 or similar.
It would really help in
are you sharing IRQ on yuor zap device?what version of libtiff you have?be sure you not are sharing IRQ whit your zap and other devices andbe sure you have the more recent version of libtiff.
On 7/6/05, Bohuslav Coufal [EMAIL PROTECTED] wrote:
Hi all,
I try to use app_rxfax.
hi patrick,
Patrick schrieb:
Did you try contacting the vendor of the base stations to see if they
have a EuroISDN firmware update? My Eicon Diva Server BRI card supports
the 1TR6 protocol. The firmware can be found here:
ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/
Perhaps AVM
On Wed, Jul 06, 2005 at 05:24:06PM -0500, Andy Brezinsky wrote:
[Span 3 D-Channel 0] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI
Spare: 0, Exclusive Dchan: 0
[Span 3 D-Channel 0]ChanSel: Reserved
[Span 3 D-Channel 0] Ext: 1 DS1 Identifier:
Hi!
I would like to use the realtime extension of Asterisk and got the
latest asterisk-addons from CVS. Upon compiling things, I got a couple
of error messages from app_addon_mysql... is it me, or are the files in
the CVS broken?
Thanks,
Christoph
___
Oops.
I forgot to add the recommendation to use the most current stable
release of all firmware/boot loader/OS for the SNOM's as that can make a
significant difference. Also, it may be better to see if you can
purchase them from a vendor that will also support you, if possible.
Sorry about
Hello,
I am getting problem for delay call hang-up with the below
scenario:
PSTN User (calling Party)---PSTN Line
FXO with Asterisk Box-SIP IP Phone
(called party)
I am using X100P card with my Asterisk-1.0.7 box. I am also
using Zaptel-1.0.7 version.
In article [EMAIL PROTECTED],
Jean-Hugues ROBERT [EMAIL PROTECTED] wrote:
But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact
that unfortunately it does not work for SIP channels due to the mixing
not being done in the zaptel driver but app_meetme itself, sort of, AFAIK).
It's
Ok can you tell me if you get any errors on a short free call? :P
You forgot to tell us what version of asterisk on both ends... wen
can only guess at this point what the problem might be.
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an
Christoph wrote:
Hi!
I would like to use the realtime extension of Asterisk and got the
latest asterisk-addons from CVS. Upon compiling things, I got a couple
of error messages from app_addon_mysql... is it me, or are the files in
the CVS broken?
Thanks,
Christoph
Please explain why your
Hi,
I spent quite a few days with this and in the end I find that the 1.07
release is by far the most stable.
I had a lot of trouble with the CVS release.
Ofcourse, thats just in my case, what do the others feel on this?
Regards,
Sahil Gupta
VoiceValley
On Thu, 7 Jul 2005, Christoph
HI all, thanks Carlos, now its all working, but i
have other cuestion, how y transfer call to other peer, when i try sip y do it
pressing the # key but with iax it is not working.
- Original Message -
From:
Carlos
Alperin
To: 'Asterisk Users Mailing List -
I'm having a strange problem with transfers on IAX phones. I have two
IAX phones behind my firewall that are extensions from my office phone
system. Both phones can receive calls, but only one of the extensions
can do blind transfers by pressing the # key. I have a similar problem
at the
Hi,
I'm trying to set up two ACT SIP/IAX capable phones to communicate with
each other on the same internal network, using asterisk 1.0.9 on SuSE
9.3 (because I intend to grow the situation after this basic setup is
functioning)
The phone IPs are set to 192.168.0.201 and 202 respectively.
I have this problem
zaphfc: empty HDLC frame or bad CRC received
My configurations are
cat /proc/zaptel/1
Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [NT] layer 1 ACTIVATED (G3)
AMI/CCS
1 ZTHFC1/0/1 Clear
2 ZTHFC1/0/2 Clear
3 ZTHFC1/0/3 HDLCFCS
cat
On Thu, 2005-07-07 at 17:04 +0200, Frank Sautter wrote:
hi patrick,
Patrick schrieb:
Did you try contacting the vendor of the base stations to see if they
have a EuroISDN firmware update? My Eicon Diva Server BRI card supports
the 1TR6 protocol. The firmware can be found here:
At 15:31 07/07/2005 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Jean-Hugues ROBERT [EMAIL PROTECTED] wrote:
But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact
that unfortunately it does not work for SIP channels due to the mixing
not being done in the zaptel
Ok... You will need to give us more information...
What type of SIP Phones are you using?? (Make and Model)
What model of WRT54G are you using? What firmware do you have on the WRT54G?
Regards,
Derek
- Original Message -
From: Blake Krone [EMAIL PROTECTED]
To: Asterisk Users Mailing
Randy MacKay wrote:
I would like to know if it is possible to slow down the dialing process in
asterisk.
I have 4 of my 8 phone lines that are VoDSL. When we try and dial out these 4
VoDSL Lines, the number is often miss dialed, or incomplete. I added a wait
before Asterisk tries to dial
Sahil Gupta wrote:
Hi,
I spent quite a few days with this and in the end I find that the 1.07
release is by far the most stable.
I had a lot of trouble with the CVS release.
Ofcourse, thats just in my case, what do the others feel on this?
Regards,
Sahil Gupta
VoiceValley
Been using
What about define those phones on the SIP.conf and use sip, instead of IAX.
That protocol use be more used to communicate Asterisk servers more than
phones.
Regards,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent:
Do you have different dialplan for IAX
SIP?, that shoudnt depend on the protocol used.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID
Sent: Thursday, July 07, 2005
12:27 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jason Frisch
Sent: Wednesday, July 06, 2005 4:22 PM
To: Jimmy Smith; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] URGENT: hardware spesifications needed
Hi carlos, the dialplan its the same, i have only
change the line dial[sip/peer] by dial[aix2/peer].
- Original Message -
From:
Carlos
Alperin
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Thursday, July 07, 2005 6:51
PM
Subject: RE:
Hi, I am sorta a newbie to the asterisk community at least in the realm of
hardware types. I was wondering, what type of card is used to allow asterisk,
on a slackware installation to talk to a standard phone line so that asterisk
can call out?
Dan
Carlos Alperin wrote:
What about define those phones on the SIP.conf and use sip, instead of IAX.
That protocol use be more used to communicate Asterisk servers more than
phones.
Regards,
Carlos Alperin
Ah - ok - I understood from the docs that IAX was better and, as the
phone was
On Thu, 7 Jul 2005 10:49:32 -0700
Dan Adams [EMAIL PROTECTED] wrote:
Hi, I am sorta a newbie to the asterisk community at
least in the realm of
hardware types. I was wondering, what type of card is
used to allow asterisk,
on a slackware installation to talk to a standard phone
line so that
Take a look here:
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400P
MARK.
Dan Adams wrote:
Hi, I am sorta a newbie to the asterisk community at least in the realm of
hardware types. I was wondering, what type of card is used to allow asterisk,
on a
Hello,
We've just started using TDMoE(local T1s connecting between Asterisk servers
in the same building over the LAN) to connect a few of our high-availability
servers instead of using crossover T1 cables. The 3 servers we have
connected to each other over TDMoE are running just fine and we have
Darren:
Thanks for your interest
I would like that once you have been verified you can use aah dial plan so
you can get all the reports for the astcc calls
Thanks for your help
Erick Weber
- Original Message -
From: Darren Wiebe [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Since May 05 I have been unable to call any non-geographic number in
the UK via Broadvoice. Thse are numbers such as the 0800 range (free
to call) 087xx (local / national rate calls). Broadvoice support have
been unhelpful, and can't say if there's any intention to fix this. A
case has been upen
Russell Horn wrote:
Since May 05 I have been unable to call any non-geographic number in
the UK via Broadvoice. Thse are numbers such as the 0800 range (free
to call) 087xx (local / national rate calls). Broadvoice support have
been unhelpful, and can't say if there's any intention to fix this.
Here is what I use:
http://www.digitnetworks.com/store/product_info.php?cPath=22products_id=28
I have used it with Slack, but now I am running it with FC4.
-Original Message-
From: Dan Adams [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 07, 2005 12:50 PM
To:
what does asterisk says in the console when you try to transfer from
the buggy phones??
asterisk -vvr
On 7/7/05, Brent Davidson [EMAIL PROTECTED] wrote:
I'm having a strange problem with transfers on IAX phones. I have two
IAX phones behind my firewall
Broadvoice could connect to non geographic numbers without difficulty
until the fourth week of May 2005.
I can call non-geographic numbers from my land line in the US, my
mobile phone and from any calling card I have tried. This isn't an
issue with BT but with broadvoice and those they contract
i have similar problem, but the sip phone just rings 1 or 2 more
times, not until the timeout expires. what is your config in
zapata.conf
specifically callprogress an busydetect parameters can help
best regards
On 7/7/05, Nahid Hossain [EMAIL PROTECTED] wrote:
Hello,
I am
its not you, its their false advertising that makes you think you can
dial these (after all their rates page *still* claims they provide
service and that its unlimited based on plan).
There are threads on voxilla.com in the broadvoice forums, which have
chat logs between me and the CTO nathan
Who are you to decide what Information can and cannot be legitimately be
sought here:?
Just curious.
--Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty
Shackleford
Sent: Thursday, July 07, 2005 12:03 PM
To: 'Asterisk Users Mailing List -
Does anyone have comment on this?
I am getting:
NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on
Primary D-channel of span 1
on my asterisk box and it seems to be causing a poping sound in the
phones, I am wondering if anyone can shed some light on this. I have
scanned the
how to edit the time 3 ms for ringing to 4
ms, i ve tried but i dindt know how,so please help me please.
__
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On Thursday 07 July 2005 19:55, Russell Horn wrote:
I can call non-geographic numbers from my land line in the US, my
mobile phone and from any calling card I have tried. This isn't an
issue with BT but with broadvoice and those they contract to supply
connections to the UK PSTN.
nod If
Hello Everyone,
Pardon me if im sounding like a total idiot, but im new to this and have to ask.
Numerous people have been telling me that I will be able to somehow
do long distance calling for free when I roll out Asterisk.. and yet none
of them can explain to me how exactly that will be.
My take on this is that they are protecting themselves against fraud.
Discounting the freefone numbers for a while, the national rate
numbers are charged at variying rates and so how is a company to know
just what they are gonna get charged.
Gavin Hamill wrote:
On Thursday 07 July 2005
After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from
CVS, Asterisk crashes on startup with an apparent MySQL
(res_config_register) error:
# asterisk -vvvgc asterisk_startup_error1.log
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so: un
I don't have all the answers. You should be able to save money on LD
because you can (in my experience) pick up substantially better rates
for voip termination than typical pstn LD. You can get plans from some
providers that allow unlimited long distance but it is all a balancing
act.
Hope
Title: Parial Hang with cvs-HEAD and queues/agentcallbacklogin
Hi
Last night I upgraded an asterisk install from cvs of early this year to current cvs head and all seemed to be working OK, but now Im having several problems which seem to be related to queues. First off queues dont work,
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