I wonder if there are any major obstacles for upgrading.
Just tried an in-place upgrade on my home box :
make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so
res_config_mysql.so; do /usr/bin/install -c -m 755 $x
the attached log with verbose=6 and debug=6 refers.
we've got a sangoma A104 (no hwec) with PRI ports 1 3 loopbacked to each
other. we're trying to have txfax send out on one of those pri ports with
rxfax listening on the other side, hence having asterisk send a fax to
itself. we however have
Check out yntx
www.yntx.com
fear prices and recides in Asia and iss it sip on asteriks they will do !
try to buy one to trye it out before buying fore hole company..
/MVH Fille
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Salutacions!!!
Si has arribat fins aqu, s perqu he configurat el meu correu perqu
et retorni un resposta automtica ja que jo estar fora durent un
parell de mesos
Vaig de vacances a Sibria...
Jordi.
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Hello all,
I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected
to an asterisk 1.2.14 and I can't record any calls using the Recording
button on this phone. The extension I configured on this phone has the
values Recording on demand, an the voicemail enabled. I am using
On Dec 21, 2007 12:37 AM, d tbsky [EMAIL PROTECTED] wrote:
hi gnubie:
snom seems has some re-brand ip phones. do they use the same firmware?
if they are the same, i don't understand why snom do this..
If I'm not mistaken, they use the same firmware. I don't know about Aztech
and SNOM
On Fri, 2007-12-21 at 18:41 +0800, Dinesh Nair wrote:
the attached log with verbose=6 and debug=6 refers.
we've got a sangoma A104 (no hwec) with PRI ports 1 3 loopbacked to each
other. we're trying to have txfax send out on one of those pri ports with
rxfax listening on the other side,
Quick and dirty too hungover to think/search... ;) Sorry list
AsteriskBoxA - SBC/PSTN voodoo - world wide Interweb - AsteriskBoxB
Account on BoxA UsernameJohn12125551212
Account on BoxB UsernameJohn102
They're both the same users, had to do some funky trunking (managed
firewall provider is
hi Fredrik :
thanks for your information.
after checking yntx manuals, i found i have one phone in my hand, which has
the same firmware with yntx phones. although it is a different brand
and looks different.
the phone's basic function is ok, but we need some advanced functions
like xml
remco,
I just had the same problem/error on my CLI when I added a polycom
shoretel IP-100 phone to my network and enabled mgcp... couldn't
figure out how to get that working yet...
I don't think it is related to 1.4 as I have been running 1.4 has been
running for over a year now without that
hi mkezys:
ok. i will add linksys to our testing list. but cisco tend to lock things.
can we get firmware for linksys easily ? or we must pay like cisco
routers and switches?
2007/12/21, Mindaugas Kezys [EMAIL PROTECTED]:
Do not forget to evaluate Linksys SPA phones. Best I tried and not
I believe you can create a blank file to keep the phone from
complaining.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Rubenstein
Sent: Friday, December 21, 2007 10:16 AM
To: Asterisk -Users
Subject: [asterisk-users] 7970 CTLFile.tlv?
I've got a Cisco 7970 that's not completing its network registration to
Asterisk. The Registering message stays on the screen (with the moving
time wheel). After a few minutes, the onscreen message flashes Updating
CTL then Loading..., then the status messages update with:
No valid CAPF
Hi,
I have the following situation
I use asterisk as o gateway between networks.
What is the reason for such response?
What are the criteria for such evaluation?
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received=
192.168.129.74
Via:
Hi,
I have a Asterisk that connects to the PSTN via a PRI. After Asterisk
sends the setup message it immediately sends a 183 Session Progress. Is
there a way I can change it so that it sends a 100 Trying instead?
Because I am having some issues with a equipment where it does not play
a busy
You don't need the .tlv file. It's optional, and will be skipped if it cannot
be found. Your problem is elsewhere. I've found that the 7970s are very
finicky. I've never had luck with the SEPMAC.cnf.xml - only
XmlDefault.cnf.xml (case may vary - check your tftp logs)
Matthew Rubenstein wrote:
I contacted one of the list users and they sent me their configuration
files.
I used it as a template and it worked with my phone, so I'll be sure to
put it back up on the Wiki.
Chad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi
How about trying the Snom phones? They are good and feature rich with regular
firmware upgrades.
Thanks
Neel
d tbsky [EMAIL PROTECTED] wrote:
hi Fredrik :
thanks for your information.
after checking yntx manuals, i found i have one phone in my hand, which has
the same
Jim Duda wrote:
Thanks Russell, that's what I'm looking for.
You're welcome!
Any idea when this will become part an official asterisk release?
It will be a part of Asterisk 1.6. (and BE C.1 ...)
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
What is the reason for such response?
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP
192.168.129.74
:5160;branch=z9hG4bK17c3.17db29e7.0;received=192.168.129.74
Via: SIP/2.0/UDP 192.168.129.74 ;branch=z9hG4bK17c3.23083974.0
Via: SIP/2.0/UDP
Hi,
I have a Polycom 330 that emits a beep every 30s or so when there is a
message waiting. Is there a way to disable that? It is pretty annoying.
Regards,
Ugo
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I'm working on a 500 seats Asterisk project.
I'm wondering whether or not I should consider using Asterisk
Realtime and a database to manage phones registrations.
As far as stability goes, I've had no problem with realtime. In fact
I've run a nationwide VoIP provider with asterisk using
Hi!
d tbsky wrote:
ok. i will add linksys to our testing list. but cisco tend to lock things.
can we get firmware for linksys easily ? or we must pay like cisco
routers and switches?
You can download latest firmware from linksys.com, also here is firmware
release notes with full changes
I did (/tftpboot/CTLFile.tlv), but the phone keeps complaining (no
change).
On Fri, 2007-12-21 at 10:25 -0500, Anciso, Roy wrote:
I believe you can create a blank file to keep the phone from
complaining.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
It may be a year old.. but until Digium is drinking their own dog food.. I
won't be using it.
On Dec 21, 2007 9:26 AM, dave cantera [EMAIL PROTECTED] wrote:
remco,
I just had the same problem/error on my CLI when I added a polycom
shoretel IP-100 phone to my network and enabled mgcp...
Try to change your verbose setting of tftboot server and look what file
is asked for exactly
Matthew Rubenstein schreef:
I did (/tftpboot/CTLFile.tlv), but the phone keeps complaining (no
change).
On Fri, 2007-12-21 at 10:25 -0500, Anciso, Roy wrote:
I believe you can create a
I've got in.atftpd running out of inetd:
- /etc/inetd.conf
tftpdgram udp waitnobody /usr/sbin/tcpd /usr/sbin/in.tftpd
--logfile /tmp/atftpd.log --pidfile /tmp/atftpd.pid --tftpd-timeout 300
--retry-timeout 5 --mcast-port 1758
This is pretty easy to suppress using the configuration files. Check:
http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
On Dec 21, 2007 11:55 AM, Ugo Bellavance [EMAIL PROTECTED] wrote:
Hi,
I have a Polycom 330 that emits a beep every 30s or so when there is a
Asterisk 1.2.13
I am trying to figure out the best way for a night bell at work.
Note: I have no spare buttons available on the phones. But I do have two lines
and two park positions as buttons.
Option 1 (easiest and the one I just implemented)
When asterisk is in night mode,
Terry Wilson wrote:
And a free hint: if you are going to have to do anything that
resembles number porting, swapping extensions, etc.--don't use
extensions/phone numbers as SIP usernames. You have to regenerate
config files, etc. Make your SIP usernames meaningless and use
Yes did all that. I've configured sendmail before so know the basics. I
tried modifying sendmail.mc and creating the sendmail.cf file and also tried
modifying sendmail.cf directly. I always restart sendmail after changes.
Would I need to create a noreply mailbox in sendmail perhaps?
What
Steve Johnson wrote:
This is pretty easy to suppress using the configuration files. Check:
http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
Looks easy once you have the config file provisioning in place, but it
looks overkill and a lot of work to set this up for the only
make sure the firmware in asterisk and the firmware on your phone matchup. it
seems like your phone is trying to update it's firmware because the firmware on
asterisk and your phone is different.
Matthew Rubenstein [EMAIL PROTECTED] wrote:
I've got in.atftpd running out of inetd:
On Friday 21 December 2007 13:16:17 Matt wrote:
It may be a year old.. but until Digium is drinking their own dog food.. I
won't be using it.
I beg your pardon. The Digium IVR has been on 1.4 since about April or so.
--
Tilghman
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Has anyone experienced the situation where you receive a
PRI_EVENT_PROGRESS message from a PRI that is then sent to a SIP channel
where the SIP client (tried 2 different phones/manufactures) never
acknowledges, Asterisk resends the message two more time and then begins
hanging the call up?
BerkHolz, Steven wrote:
Option 3 (I believe this is best, but am not sure where to start)
When asterisk is in night mode,
I'm doing option 3, menu item on the IVR to ring the night bell. Plays
an awfully loud horn noise on the PA while ringing a phone out an the
plant floor
I have mine set up to ring a group of designated phones. Each one of
those phones has a dedicated line button that subscribes to their
particular account in the group. This way when the phone rings the user
KNOWS that it is the main building number that is ringing.
-Original Message-
Doug Lytle wrote:
BerkHolz, Steven wrote:
Option 3 (I believe this is best, but am not sure where to start)
When asterisk is in night mode,
I'm doing option 3, menu item on the IVR to ring the night bell. Plays
an awfully loud horn noise on the PA while ringing a
You are probably running into the problem described below. Below that
is a link to the original document with the code patch. I put it on a
PRI box we use inhouse and it took care of the 183 before a busy for
me. However, this is a box we use inhouse. I've never put it on
anything in
On Dec 21, 2007, at 1:29 PM, Brian Capouch wrote:
Terry Wilson wrote:
config files, etc. Make your SIP usernames meaningless and use
func_odbc to look up what extension is tied to which device.
I wouldn't say to make the names meaningless, though; there are
different ways to use those
And in case that link doesn't work so well in text email clients here
is the real address.
lists.digium.com/pipermail/asterisk-dev/2006-May.txt.gz
Richard
On Dec 21, 2007, at 4:24 PM, Richard Revels wrote:
You are probably running into the problem described below. Below
that is a link to
shadowym wrote:
What creates the asterisk mailbox? Does that happen when I make
asterisk?
The return-path is set to [EMAIL PROTECTED] Your asterisk process is running as
the user 'asterisk', no? If so, that's why you're getting the return
path set to 'asterisk'.
I haven't used sendmail in
MatsK wrote:
Doug,
I see that you use cp to copy the call file to spool directory, that
is not recommended, use mv instead since it is a atomic command
whitch cp isnt.
Thanks!
I'll have to update that.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
And a free hint: if you are going to have to do anything that
resembles number porting, swapping extensions, etc.--don't use
extensions/phone numbers as SIP usernames. You have to regenerate
config files, etc. Make your SIP usernames meaningless and use
func_odbc to look up what
Running on branch/1.4
I have been watching some the queries from Asterisk and I think I have a place
where some efficiency can come, but I am at a lost as to what is calling it...
It appears that every 10-15 seconds, every mailbox INBOX and Old gets queried
for the number of voice mail files.
Hi All
I have been asked if it is possible for an external application to be
aware of the position of the playbcak of a file with control playback
ie a file is playing and the user hits the fast forward button , is
there a manager event that show how far into the file it has been played?
So I would like to change this, just not sure where to look. It
seems to me we only need to query this when we call the VM app or
a user enters the VM system.
Maybe the same code that calls the externnotify command (a custom
post-exec script) could trigger the SIP notification beforehand. A
Tony Plack wrote:
Running on branch/1.4
I have been watching some the queries from Asterisk and I think I have a
place where some efficiency can come, but I am at a lost as to what is
calling it...
It appears that every 10-15 seconds, every mailbox INBOX and Old gets queried
for the
Ugo Bellavance wrote:
Steve Johnson wrote:
This is pretty easy to suppress using the configuration files. Check:
http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio
Looks easy once you have the config file provisioning in place, but it
looks overkill and a lot of
Dec 21
13:43:48 DEBUG[15840] pbx.c: Function result is '20071221-134348'
Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Executing AGI(Zap/2-1,
recordingcheck|20071221-134348|1198205023.1657) in new stack
Dec 21 13:43:48 VERBOSE[15840] logger.c: -- Launched AGI Script
/var/lib/asterisk/agi-bin
Hi there,
I have a Polycom phone that has two extensions registered to it, let's say 200
201.
Is there anyway to code in the Asterisk dialplan to show BOTH lines are busy
when either of 200 or 201 are in use?
Reason is that my Polycom phones will show the presence info via BLF red light,
hi, all
proble:
I have add CALL-LIMIT field in my sip table in mysql.
but when i call using sip same error occurred when use simple sip.conf file.
is this possible to add CALL-LIMIT field in sip realtime table in mysql.
if yes than how
Bhrugu Mehta
I will be out of the office until Wednesday, January 2, 2008. If this is an
emergency, please call Customer Service at (877) 791-7700. Thank you have a
great holiday season!
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21 dec 2007 kl. 10.12 skrev Remco Barendse:
I wonder if there are any major obstacles for upgrading.
Just tried an in-place upgrade on my home box :
make[1]: Leaving directory `/usr/src/asterisk-addons-1.4.5'
for x in app_addon_sql_mysql.so app_saycountpl.so cdr_addon_mysql.so
21 dec 2007 kl. 22.24 skrev Richard Revels:
You are probably running into the problem described below. Below
that is a link to the original document with the code patch. I put
it on a PRI box we use inhouse and it took care of the 183 before a
busy for me. However, this is a box we
22 dec 2007 kl. 06.40 skrev Bhrugu Mehta:
hi, all
proble:
I have add CALL-LIMIT field in my sip table in mysql.
but when i call using sip same error occurred when use simple
sip.conf file.
You can check if it works with sip show peer. The call limit you set
in the
database should be
Friends,
Thanks for all the feedback. If you have additional success stories or
important
issues, feel free to continue the discussion.
I've learned a lot from your input. As a developer, I spend too much
time in
the bug tracker, working with particular bugs, so I often wonder how
on earth
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