I was about to post on this thread that I have contacted the makers of
iSip and they got back to me, we're working on a fix. We because I did
For info, the fix seems to solve the problem, VNET is waiting for
Apple approval on the new version of the app. I really like the
multiple accounts of
On Mon, 2009-08-10 at 21:37 -0500, Tom Poe wrote:
I want to set sflphone as extension on asterisk. I have a sip
account/DID with vitelity.net. Not sure what to put in the wizard:
alias ???
hostname ??? is this the asterisk server hostname, or the hostname
where my sflphone is sitting
Hey here are the sample configuration. Create a trunk in sip.conf file, add a
registry string.
Registry String.
register= user1:passw...@anysipprovider.com:5060
[user1]
type=peer
host=anysipprovider.com
port=5060
context=default
country=us
dtmfmode=rfc2833
restrictcid=no
canreinvite=yes
Say,
I need to replicate what happens on a wired extension when a call is transfered
and transfered back.
Asterisk has to detect and pass through the flash hook to the Quintum when its
pressed on the Eyebeam.
My setup is:PBX--Quintum FXS port -- Asterisk 1.4 Server--Eyebeam 1.5
softphone
The
Thanks for your response cause I really need the answer of my problem,
yes I made a mysql database that cdr data is being saved there. but I
checked config files, there is no cdr_mysql.conf file cause we use cdr_odbc
file. in cdr_odbc there is not any setting for user field. we also set in
While I read on some other mailing list that the human ear is a poor
testing device, it is still a widely available testing device and I
often don't have anything better.
In order to help that device better detect sound quality issues, I tend
to prefer to use lengthy music files. Once I'm
On 11 Aug 2009, at 12:03, Tzafrir Cohen wrote:
So I'm looking for a music file that:
1. Sounds well (enough) even at the standard PSTN quality (8kHz, mono,
16 bits per sample).
2. Is long enough. E.g. ~10 minutes.
3. No usage limitation. Freely usable. So I can point it out to
someone
Anybody tried one with Asterisk yet ? Views ?
Apparently not available until the end of August.
We've certainly used the Snom 820 with Asterisk without any issue in the past,
and since both are based on (largely) the same software, I doubt there'll be
any major problems with asterisk
- Chris Bagnall li...@minotaur.cc wrote:
Anybody tried one with Asterisk yet ? Views ?
Apparently not available until the end of August.
We've certainly used the Snom 820 with Asterisk without any issue in
the past, and since both are based on (largely) the same software, I
doubt
Hi,
I am getting the following messages for a few days.
WARNING[10148]: chan_iax2.c:1219 __send_lagrq: I was supposed to send a
LAGRQ with callno 15691, but no such call exists (and I cannot remove
lagid, either).
I got 2 iax trunks which used by test purposes only.
And no iax clients or no
Hi. Using asterisk 1.4 svn 21112, when I try to use the mixmonitor
feature I get the following in the log file.
[Aug 11 09:22:54] WARNING[32057] file.c: Tried to write non-voice frame
[Aug 11 09:22:54] WARNING[32057] channel.c: Failed to write data to
channel monitor write stream
After several
Asterisk Project Security Advisory - AST-2009-005
++
| Product | Asterisk |
The Asterisk Development Team is pleased to announce the releases of 1.2.34,
1.4.26.1, 1.6.0.12, and 1.6.1.4. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of 1.6.1.4 fixes a remote crash security vulnerability in the SIP
Hi,
I'm currently having problems detecting FSK BT (UK) caller id in our API
(Pika boards).
I have a recording to test on but it is giving me checksum errors.
I'm wondering if someone from UK using BT lines could send me a recording
with the FSK signal so I can have more data to work on? If you
Hello,
I think there is a problem with chars in the extension name. I have a
similar issue if i try to use my my que management macro with a
extension with characters.
On Mon, Aug 10, 2009 at 3:16 PM, Tarek Sawahtareksa...@hotmail.com wrote:
i faced the same problem with callcentric.. when i
Can't find an answer to this, but maybe I've not looked hard enough ...
Does MixMonitor work without transcoding?
ie. if I have a g729 stream passing through and I'm recording it with
e.g. MixMonitor(/dump/filename.g729,b)
and specify g729 in the filename, does MixMonitor transcode both legs
I tried patching my 1.6.0.10 source to 1.6.0.12 and during compilation
(make) I get a string of errors in chan_sip.c that start:
chan_sip.c: In function \u2018handle_incoming\u2019:
chan_sip.c:18669: error: expected expression before \u2018\u2019 token
chan_sip.c:18674: error: \u2018ret\u2019
I have a cisco 1760 phone running sip and I need to configure for our
receptionist so that she can answer calls on more then one extension.
What is the easiest way to configure this so that incomming calls go to
the next availble extension?
Does each extension on the phone need to be set
Sorry I mean to say cisco 7960 phone.
From: Jimmy Ezell
Sent: Tuesday, August 11, 2009 9:15 AM
To: 'asterisk-users@lists.digium.com'
Subject: Cisco 1760 Multiline phone
I have a cisco 1760 phone
I'm trying to use func_odbc to write to a MS SQL db.
Here's my func_odbc conf:
[OPTIN]
dsn=MSSQL-Optin
write=INSERT into OptIn (orgID) values (${VAL1})
Dial Plan
exten = +18665551212,n,Set(ODBC_OPTIN()=dave)
When I do an odbc show, it shows that I am connected to the db. If I use isql,
I
Yes each extension needs to be configured separately in the cisco CNF file.
I use a distinct extension on each phone (2 phones can't register to one
'extension' afaik) and ring them in order:
1,1,Dial(SIP/xx)
1,n,Dial(SIP/xx1)
1,n,Dial(SIP/xx2)
Or ring them at the same time:
On Tuesday 11 August 2009 11:33:30 David Budny wrote:
I'm trying to use func_odbc to write to a MS SQL db.
Here's my func_odbc conf:
[OPTIN]
dsn=MSSQL-Optin
write=INSERT into OptIn (orgID) values (${VAL1})
Dial Plan
exten = +18665551212,n,Set(ODBC_OPTIN()=dave)
When I do an odbc show,
On Tue, 11 Aug 2009, Gordon Henderson wrote:
Can't find an answer to this, but maybe I've not looked hard enough ...
Does MixMonitor work without transcoding?
ie. if I have a g729 stream passing through and I'm recording it with
e.g. MixMonitor(/dump/filename.g729,b)
and specify g729 in
David wrote:
When I do an odbc show, it shows that I am connected to the db. If I use
isql, I can write to the db, however, when I use func_odbc, a record will not
write. I'm using asterisk 1.4.9. Any idea what might be wrong?
Sounds sneakingly like a permissions problem in your database
Barry L. Kline wrote:
I tried patching my 1.6.0.10 source to 1.6.0.12 and during compilation
(make) I get a string of errors in chan_sip.c that start:
chan_sip.c: In function \u2018handle_incoming\u2019:
chan_sip.c:18669: error: expected expression before \u2018\u2019 token
On Tuesday 11 August 2009 12:29:48 Myles Wakeham wrote:
Personally I don't use ODBC for this sort of thing. Its not really a
'Linux' type thing, and relies on a lot of middle-ware layers between
the Asterisk server and the database. Not that you have too many
choices, but what we do that
Thanks for the help, I really appreciate the feedback.
I tried ringing them all at the same time as you suggested:
exten =
workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomm
ing4SIP/incomming5)
but it does very strange stuff:
- I have to push the extension button twice
Jimmy,
To clarify, you want to configure the phones like this where p means phone and
l means logical line:
Phone 1:
P1l1
P1l2
P1l3
Phone 2:
P2l1
P2l2
P2l3
Phone 3:
P3l1
P3l2
P3l3
It sounds like (and looks like) you're dialing all of the extensions on one
phone at the same time, which is
Yes this is a well-known change. If you use the latest Snapshot of
Asterisk-Java 1.0.0 (http://asterisk-java.org) the issue should be resolved.
=Stefan
Alex Balashov wrote:
Appearances suggest that some part(s) of the AGI protocol changed
between 1.4 and 1.6.
hh174 wrote:
Hello,
I
Date: Tue, 11 Aug 2009 12:19:15 -0500
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
Subject: Re: [asterisk-users] func_odbc insert with mssql
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
Tilghman Lesher wrote:
You've just stated the primary reason for folks to use ODBC. Perhaps
you've
written off the technology too soon and for the wrong reasons.
You maybe right as I've been a SQL developer for about 25 years
(actually before ODBC was ever around) and I might be tainted by
Anyone have a chance to test any of the various iPhone SIP apps?
I see there are a few out there, but most of the iTunes reviews aren't
sufficiently technical to be useful.
Thanks.
___
-- Bandwidth and Colocation Provided by
Philip A. Prindeville wrote:
Anyone have a chance to test any of the various iPhone SIP apps?
I see there are a few out there, but most of the iTunes reviews aren't
sufficiently technical to be useful.
Thanks.
I got iPico and it is working pretty well for me so far.
Gordon Henderson wrote:
Transcoding is something that's not an option here. Hm. Maybe old
fashioned 'monitor' and offline mixing although I'm open to suggestions
here..
In general, it is not possible to mix compressed audio; it must be
uncompressed first.
--
Kevin P. Fleming
Digium, Inc.
On Tue, Aug 11, 2009 at 1:57 PM, Philip A.
Prindevillephilipp_s...@redfish-solutions.com wrote:
Anyone have a chance to test any of the various iPhone SIP apps?
Here's a discussion of a few we've tried: http://VUC.me
I like iSip but the other two are good, too. iSip has multiple
accounts which
For the price WeePhone is decent.
Sent from my iPhone
On Aug 11, 2009, at 4:57 PM, Philip A. Prindeville
philipp_s...@redfish-solutions.com
wrote:
Anyone have a chance to test any of the various iPhone SIP apps?
I see there are a few out there, but most of the iTunes reviews aren't
I have a provider that in order to set outbound CID they want me to
make sure that the From Header in the sip invite matches the caller ID
while the contact header matches the registration info.
For example.
My phone number with my provider is 2125551212 which is also my
username. I want caller ID
Sorry for not being real clear.
What I have is 1 front desk phone only with 6 lines
Front Desk Phone line 1 - incoming extension 1
Front Desk Phone line 2 - incoming extension 2
Front Desk Phone line 3 - incoming extension 3
Front Desk Phone line 4 - incoming extension 4
Front Desk Phone line 5
Myles Wakeham schrieb:
My biggest negative to ODBC is that the calls that the client
application has to make to the database are hard to visualize and debug,
particularly if you are crafting a SQL QUERY statement that goes on for
pages. Its for this reason that I'd want to have something
What I have is 1 front desk phone only with 6 lines
Front Desk Phone line 1 - incoming extension 1
Front Desk Phone line 2 - incoming extension 2
Front Desk Phone line 3 - incoming extension 3
Front Desk Phone line 4 - incoming extension 4
Front Desk Phone line 5 - incoming extension 5
With that phone what you really probably want to do is just configure them
all with the same details...
i.e.
# Line 1 appearance
line1_name: incoming
line1_shortname: Incoming (Line1)
line1_authname: incoming
line1_password: password
# Line 2 appearance
line2_name: incoming
line2_shortname:
On Tue, Aug 11, 2009 at 5:12 PM, Jimmy Ezell jez...@hmhca.com wrote:
Sorry for not being real clear.
What I have is 1 front desk phone only with 6 lines
Front Desk Phone line 1 - incoming extension 1
Front Desk Phone line 2 - incoming extension 2
Front Desk Phone line 3 - incoming
Hello,
I recently completed a fresh install of Asterisk
SVN-group-srtp-r183146M-/trunk , and I'm running into an issue getting the
voicemail application module to load. Output from debug shows:
---
[Aug 11 22:00:01] NOTICE[20173]: loader.c:875 load_modules: 1 modules
Hi,
Is anyone successfully using SIP-enabled Cisco 79XX phones with Asterisk ?
Could you then configure this phone to display non-english menus (in french,
spanish, german, ...) ?
Mine is using a rather old SIP firmware (8.3 ?) with which I could get
non-english menus.
Regards
This isn't asterisk related but I figure several developers on this list
have built apps for Twitter (or other 3rd party API's).
Just found out a few hours ago I'm being sued by Twitter
Feel free to tweet this link ( www.MyTwitterButler.com/I'm_Being_Sued
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