What version are you running, and is your [Cisco]
definition the last one in the file? I have the same problem with 1.0.7,
and the ugly fix I came up with was to add a dummy entry as the last sip
entry.
B. J.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Sent: Friday, November
I have a script that doesn't quite fit your needs, but does send out email
reminders for on a regular basis, and runs as a daemon. If you are
interested, please let me know and I will send it to you. A little warning,
this was one of my first major perl scripts, so it may be a little ugly
and
The only way is if you are using DHCP to get an IP address
to the phone. If you are, then you can have it point the phone to a TFTP
server with config files with a new password. If you are using a static
IP, then you are out of luck. I opened up a TAC case about a year ago, and
that is what
I've tried it in 1.2, and maybe I'm just not smart enough to get it to work.
Do you have a working example? What I am looking for is
[EMAIL PROTECTED] to be different that [EMAIL PROTECTED] As
far as I can tell, currently for registrations asterisk only looks at
everything to the left of the @
Jerry, here are the relevant parts of my 7206 config. Some things have been
changed to protect the innocent. ;)
dspint DSPfarm1/0
codec med
!
isdn switch-type primary-ni
!
!
voice call send-alert
!
voice service pots
fax protocol pass-through g711ulaw
!
voice service voip
fax protocol
Try putting the command timers buffer-invite 5000 in the sip-ua config.
This works on both our 3640 and 7206. I'm not sure if this command is
available in the 12.3 series as I have 12.3T on my equipment.
B. J.
-Original Message-
From: Max Braz [mailto:[EMAIL PROTECTED]
Sent:
1.2.0-beta1 tarball re-released
B. J. Bomar wrote:
I am also having the same issue from the ftp tarball.
I've tested the tarball on a bunch of different systems and it worked
properly.
Please post the contents of the include/asterisk/version.h file from
your source tree after the build
-released
B. J. Bomar wrote:
I could not find a .version file at the top level of the tarball. Below
is
what my include/asterisk/version.h file contains.
Please re-download the tarball, making a note of the IP address of the
server you get it from. If it still doesn't contain a .version file
Upon further investigation, even if you untar a fresh copy and just issue
make, it will do a make clean. It looks like the first thing it does is
make cleantest which in turn issues a make clean because the file .lastclean
does not exist.
Another question, why does make clean even remove the
I am also having the same issue from the ftp tarball.
B. J.
-Original Message-
From: Martin Morey [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 30, 2005 8:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.0-beta1 tarball
Here is my entry in sip.conf that works for 7200's, 3600's,
and 2600's.
[gateway]type=friendhost=192.168.1.61canreinvite=yescontext=gw-inboundqualify=nodtmfmode=rfc2833insecure=yesdisallow=allallow=ulawallow=alaw
Hope that helps.
B. J.
From: Scott Miller [mailto:[EMAIL PROTECTED]
Sent:
Here is a crude hack, but it requires the user to press # at the end.
exten = s,1,Playtones(dial)
exten = s,2,Read(1stnumber,,1)
exten = s,3,StopPlaytones
exten = s,4,Read(restofnumber)
exten = s,5,SetVar(totalnumber=${1stnumber}${restofnumber})
Hope that helps.
B. J.
-Original
I think the file you want to edit is the dialplan.xml. I don't remember the
syntax off the top of my head, but I'm sure it is documented on the Cisco
web site.
B. J.
-Original Message-
From: Roland Zagler [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 12, 2005 11:22
To:
Here is an example of my crude hack to emulate DISA. It requires the user
to hit the # key when they are finished dialing.
[DISA_hack]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Authenticate(${DISA_password})
exten = s,4,Playtones(dial)
exten = s,5,Read(1stnumber,,1)
exten =
I believe it is vm-login.gsm. Hope that
helps.
B. J.
From: Henry Devito [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 29, 2005 12:49To: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users]
Voicemail sounds
Which sound file is the one you hear when you
Try using the Cut application. For your example channel you can use the
following.
exten = whatever,n,Cut(my_variable=CHANNEL,@,1)
That should give you your IAX2/white_phone. For more info, take a look at
either the wiki or CLI help.
B. J.
-Original Message-
From: Thomas Andrews
, 2005 14:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Channel name (and substring)
On Thu, Mar 17, 2005 at 02:06:24PM -0600, B. J. Bomar wrote:
Try using the Cut application. For your example channel you can use the
following.
exten = whatever
Take a look at
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a
few small modifications it should work like a champ on the Polycom phones.
B. J.
-Original Message-
From: Eric Rees [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 01, 2005 10:38
To: Asterisk
Discussion
Subject: Re: [Asterisk-Users] Polycom Auto-Answer
B. J. Bomar wrote:
Take a look at
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a
few small modifications it should work like a champ on the Polycom phones.
B. J.
Polycom has a much better way to do auto
Hello all, I am
looking at replacing our current Cisco PRI gateway with a new server with a
TE405P card.My primary concern is receiving CallerID Name info on
the D-Channel. Does anyone have any experience terminating a local Qwest
PRI from a 5ES switch into the TE405P or similar? We are
Indirectly. You can use the ex-girlfriend feature to route the calls
differently. The following is a simple example.
exten = s/5551234,1,Goto(context1,s,1)
exten = s/678,1,Goto(context2,s,1)
exten = s,1,Goto(default,s,1)
I hope that helps you out.
B. J.
-Original Message-
I was doing some general browsing around, and found this press release
about some new SIP based Polycom offerings.
http://www.convergedigest.com/PacketSystems/packetsysarticle.asp?ID=13530
The SoundStation IP 4000 looks interesting, and appears to run on the same
firmware as the rest of their
While your solution may work now, it might not work on the next release as
both incominglimit and outgoinglimit are deprecated. Here is an idea on how
to use SetGroup and CheckGroup using your template as an example.
exten = 1051,1,SetGroup(${EXTEN})
exten = 1051,2,CheckGroup(1)
exten =
waiting again..
B. J. Bomar wrote:
While your solution may work now, it might not work on the next release as
both incominglimit and outgoinglimit are deprecated. Here is an idea on
how
to use SetGroup and CheckGroup using your template as an example.
exten = 1051,1,SetGroup(${EXTEN})
exten
Have you tried adding SetGroup(), and CheckGroup() functions to the dialplan
that rings the phone? It maybe something to try.
B. J.
-Original Message-
From: Sean A. Newton [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 25, 2005 17:18
To: asterisk-users@lists.digium.com
Yes, the Shoreline IP100 is just a rebranded Polycom Soundpoint IP500, and
uses the same software. By default it uses a MGCP image, but it can be
changed to run SIP. See
http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones for more info.
B. J.
-Original Message-
From:
One thing that absolutely bothers me about DISA is the time it waits to
dial. In order to get around it I hacked together a context that does a
similar function. In case anyone wants it, here it is.
[remote_access]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Authenticate()
exten =
I have also purchased Shoreline IP100's, and they are functionally the same
as the Polycom SoundPoint IP500's. In regards to your questions, I can
answer three of the four.
1. I believe you can get MGCP to work with asterisk, but I just went ahead
and updated to the SIP image.
2. Not sure on
If you really want to just listen to the dial tone, you could use the
Playtone(dial) app. Do a show application Playtone for more info.
B. J.
-Original Message-
From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 13, 2005 22:07
To: [EMAIL PROTECTED];
seem clear whether this patch was included or not in the latest
stable. I'm about to upgrade but need to know if I should apply this patch.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of B. J. Bomar
Sent: Tuesday, January 11, 2005 9:05 PM
To: 'Asterisk
Hello all. I
just found out that I am no longer able to exit out of voicemail properly by
hitting the 0 key, but the * key works. Asterisk comes back and says "I'm
sorry, I did not understand that response" and goes on in the context. Is
this a new "feature" or bug? Is anyone else having
: Re: [Asterisk-Users] o extension broken?
you need to define the o extension to do whatever you want it to do.
-Matthew
- Original Message -
From: B. J. Bomar [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday
operator=yes defined for each voicemail user.
see patch: http://bugs.digium.com/bug_view_page.php?bug_id=0003080
-matthew
- Original Message -
From: B. J. Bomar [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday
What does the console debug look like when you try
it? What happens if you remove the "p" option on the MeetMe app?
Other than that, it looks close to what I have working for our Cisco
7960's.
B. J.
From: Christopher Tuska (HOME)
[mailto:[EMAIL PROTECTED] Sent: Tuesday, December 28,
Here is an idea to try. Maybe someone else has a cleaner solution.
exten =
9043442342,1,Dial(local/[EMAIL PROTECTED]local/[EMAIL PROTECTED]local/[EMAIL
PROTECTED]l
ocal/[EMAIL PROTECTED],,20)
exten = 9043442342,2,Voicemail(u102)
[rollover]
exten = _10X,1,Dial(SIP/10${EXTEN:2},,21)
exten =
I have a quick question for the list. For what reason would you have
multiple instances of asterisk running on a single box? I can maybe see it
if you have multiple IP addresses, but other than that I am drawing a blank.
Thanks,
B. J.
___
Does anybody know if the CS version of the Polycom handset will take the SIP
image. If I have read correctly, the CS version is for Cisco Call Manager,
and is Cisco certified. Thanks in advance.
B. J.
___
Asterisk-Users mailing list
[EMAIL
I use a Plantronics Supra H51 plugged straight into the headset port, and it
works great.
B. J.
-Original Message-
From: Nate Carlson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 15:05
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] OT: Headset for Cisco 7960?
Sorry, I
Title: Message
Hello all. We
have an asterisk system set up, and we are seeing a lot of multiple DTMF digits
being read by asterisk. In digging through the archives the only answer I
have seen is to put in the statement relaxdtmf=yes in the zapata.conf
file. Since we are not using any
Title: Message
Do you
need to have a zap interface for it to work?
B.
J.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker,
Scott (7805)Sent: Wednesday, February 18, 2004 15:34To:
[EMAIL PROTECTED]Subject: RE:
Title: Message
Has anyone got the
call pick to work on the Cisco 7960's? I have tried to get it to work a
couple of time, but all I get is the following error.
NOTICE[1142135600]:
chan_sip.c:5355 handle_request: Nothing to pick up
Thanks,
B.
J.
Title: Message
Here
is some config that I cooked up. It may be a little rough around the
edges, and it incorporates multiple users.
exten
= *801,1,Answerexten =
*801,2,SetVar(temp=${loggedin${CALLERIDNUM}})exten =
*801,3,GotoIf($[${temp} = 1]?50:)exten = *801,4,GotoIf($[${CALLERIDNUM}
=
I too have seen a couple of system freezes for no apparent reason. I am *
on a RH9 box with kernel 2.4.20-28.9.
B. J.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Biggs
Sent: Monday, February 09, 2004 12:05
To: [EMAIL PROTECTED]
Subject:
Title: Message
Hello all, I am
trying to figure out how to have * release a phone call. We are noticing
some call quality issues on people who have a "find-me" feature, and answer the
call through a cell phone. Here is the call path we are seeing, and all
VoIP connections are using SIP.
d any real information about it. Is this simply call forward
or is their more to it.
thanks
- Original Message -
From:
B. J. Bomar
To: [EMAIL PROTECTED]
Sent: Thursday, February 05, 2004 1:01
PM
Subject: [Asterisk-Users] Release phone
Do not define any members in the queues.conf. Instead have them login to
the queue using the AddQueueMember application. If there is no one logged
into the queue when a call comes in, it will go to the priority in the
context. Hope this helps.
B. J.
-Original Message-
From: [EMAIL
the phone.
Rgds,
Adam
-Original Message-
From: B. J. Bomar [mailto:[EMAIL PROTECTED]
Sent: Monday, January 19, 2004 6:42 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Remote reload Cisco 7960
I've tried to use that script, but the phones seem to ignore it. I am in
the process
I've tried to use that script, but the phones seem to ignore it. I am in
the process of upgrading to 6.1 on the phones, maybe they will behave like
they're supposed to.
B. J.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Friday,
Title: Message
Does anyone have a
working way of having a Cisco 7960 reload its config remotely. I have
tried some of the scripts that I have found on the web, but to no avail.
Thanks for the help.
B.
J.
reload Cisco 7960
On Fri, Jan 16, 2004 at 12:13:19PM -0600, B. J. Bomar wrote:
Does anyone have a working way of having a Cisco 7960 reload its config
remotely. I have tried some of the scripts that I have found on the web,
but to no avail. Thanks for the help.
I just telnet
PROTECTED]
Subject: Re: [Asterisk-Users] Remote reload Cisco 7960
Quoting B. J. Bomar [EMAIL PROTECTED]:
Does anyone have a working way of having a Cisco 7960 reload its config
remotely. I have tried some of the scripts that I have found on the web,
but to no avail. Thanks for the help.
Try
ACL's, the mask is the inverse of the standard IP mask.
-Original Message-
From: B. J. Bomar [mailto:[EMAIL PROTECTED]
Sent: Monday, January 12, 2004 1:56 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Securing Cisco SIP gateway
I too am attempting to lock down a Cisco gateway. I
I too am attempting to lock down a Cisco gateway. I have been trying to use
the voice source-group command. This is what I currently have.
voice source-group test
access-list 61
disconnect-cause call-reject
!
access-list 61 permit 10.1.1.2
access-list 61 permit 10.1.1.3
access-list 61 deny
Title: Message
Hello all. Has
anyone had any success using ChanIsAvail with only SIP channels? Is there
another, better way to check if an extension is busy without dialing
it?
Thanks,
B.
J.
Title: Message
Hello
all. I know * doesn't directly support recording mp3 files, but I was
wondering if anyone has created an AGI to do it indirectly. Thanks in
advance.
B. J.
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