Hello all,
I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones,
on an amd_64 processor.
All goes well, the voice is clear on the remote side but in the Voip side,
where the Snom 320 is placed, I hear my voice, but don't in the line, the
echo is on the phone.
I just play with
On Wed, Nov 07, 2007 at 11:03:30PM -0600, Eric ManxPower Wieling wrote:
I'm not a fan of using the Wiki as a reference, but there really isn't
any info like this in the docs that come with Asterisk.
Agreed, except to note that the explanations given in Asterisk: The Future of
Telephony serve
I have a grandstream 488 using FXO port. With asterisk 1.2.23
When I have DTMF mode set to RFC2833 (asterisk and grandstream) and I
use a call file and AGI to
originate the call I dont get the DTMF tones on the device.
If I do it manually from a polycom 550 (set for RFC2833) it works.
When I
Hello,
When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors.
Failed dependencies:
libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
But i found the same files in
Hi all:
I don't know if exist any other mailing more apropiated for this question. If
exist, please let me know.
I need orientation for this situation:
1. 1.4.13-BRIstuffed with support for h323 with asterisk-h323 module
2. An analog Pbx with support por h323 make asterisk a call, that asnwer
Yes - we've been over this with Aastra support, and they acknowledge a bug
in their firmware but can't seem to find it. They said wait for the next
firmware release (and at least 2 releases have passed).
We had SOME success by creating a blank config file, changing the order of
entries in the
Hello,
I have some extensions that are using variables loaded by an AGI program.
Everything works fine and I am able to use NoOp to see the value of my
variables when using IAX, but the same variables don't work when using SIP.
I can provide further details, but right off of the bat does is
Is there any way to see the called number when a call gets redirected to
the 'a' extension from voicemail? Say x123 calls x456 and it rolls to
voicemail. x123 hits * and gets dumped into the 'a' extension in the
original context. I need some logic in 'a' to do a database lookup
based on the
I have a polycom 501 phone that is 1 hour off now.
Before last sunday (time change) the time was fine.
?xml version=1.0 standalone=yes?
PHONE_CONFIG
OVERRIDES _.0x20._log.level.change.sip=0
tcpIpApp.sntp.daylightSavings.stop.date=4
tcpIpApp.sntp.daylightSavings.stop.month=11
exten = whatever,1,Answer()
rest of your dialplan for the queue
Thanks,
Steve Totaro
Andre Quintaes wrote:
On our tests using asterisk, some calls have been terminated
abruptely with exact 185 seconds. This is happening with all our
incoming calls from a trunk from 1 of my DID providers (
Jerry Geis wrote:
I have a polycom 501 phone that is 1 hour off now.
Before last sunday (time change) the time was fine.
Google is your friend:
http://www.google.com/search?hl=enq=polycom+daylightSavingsbtnG=Google+Search
Top hit fixed it for us.
For the archives, in case the top hit is no
Hi Jerry,
Here's what's in my SNTP tag:
tcpIpApp.sntp.resyncPeriod=3600
tcpIpApp.sntp.address=192.168.15.50
tcpIpApp.sntp.gmtOffset=-18000
tcpIpApp.sntp.daylightSavings.enable=1
tcpIpApp.sntp.daylightSavings.fixedDayEnable=0
tcpIpApp.sntp.daylightSavings.start.month=3
On 11/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Is there any way to see the called number when a call gets redirected to
the 'a' extension from voicemail? Say x123 calls x456 and it rolls to
voicemail. x123 hits * and gets dumped into the 'a' extension in the
original context.
Off-hand, have you compared the output of agi debug (on the console)
between the working and non-working calls? I believe the variables all get
displayed.
-jr
On Nov 8, 2007 6:13 AM, Jason Wolfe [EMAIL PROTECTED] wrote:
I have some extensions that are using variables loaded by an AGI program.
7 nov 2007 kl. 14.26 skrev Tony Mountifield:
Quick question for those who know the innards of chan_sip:
Does chan_sip use the To: header of an incoming INVITE request,
for anything other than setting SIP_HEADER(TO) ?
No. Like e-mail software not using the To: header in the actual e-mail.
I think you can save/get the number in variable and then assign it to
callerid. I am doing similar and working for me.
Thanks,
Viv
On 11/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Is there any way to see the called number when a call gets redirected to
the 'a' extension from
Peder @ NetworkOblivion wrote:
Is there any way to see the called number when a call gets redirected to
the 'a' extension from voicemail? Say x123 calls x456 and it rolls to
voicemail. x123 hits * and gets dumped into the 'a' extension in the
original context. I need some logic in 'a' to
Dialing Exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) is valid. If you can post
your dial plan and we can take a look (though I have never seen this error
before).
- Original Message -
From: Rizwan Hisham [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Paul wrote:
I have six cisco 7911g connected on asterisk over
chan_skinny. Four of them are working OK. two of
them even the screen on the phone is indicating that
is registered and has number loose connection to
asterisk . On asterisk the message is Skinny Client
was lost,
Setup a 2nd registration on the phone that only allows 1 call at a
time. Ideal setup it up as a shared appearance so call forwarding,
etc dont work on that registration. This way your phone has 2
registrations 1 for any direct call and another for shared calls,
queues, etc.
* hint: The 'hint' priority associates an extension with an
Asterisk channel for the purpose of mapping the state of the channel
to a state of the extension.
In asterisk, a channel (technology/device) can have several states
(unavailable, in-use, busy, ringing, etc) but an extension is just
On Tue, 2007-11-06 at 22:16 -0800, Josh Richards wrote:
This may be what you need:
http://www.misdn.org/index.php/FAQ#Why_are_my_dtmf_tones_not_detected_everytime.3F
Also, something here may be helpful:
http://www.voip-info.org/wiki/view/Asterisk+DTMF#Troubleshooting
-jr
On Nov 6,
On Nov 6, 2007 2:25 PM, Jarga Jallow [EMAIL PROTECTED] wrote:
Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx
D N 5066 UNREACHABLE
11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE
10/10 76.xxx.xxx.xxx
All,
If someone calls into my asterisk box and has a private number I would like to
set the callers id to a specific telephone number, only when the ANI is
missing, otherwise if present just pass it along. Any ideas?
TIA,
Jon___
--Bandwidth and
Openfire has a SIP soft phone plugin (and an AsteriskIM plugin -
different functionality though)
http://www.igniterealtime.org/
Eric Chamberlain wrote:
Hello,
Im looking for a SIP to
XMPP Jingle voice gateway.
I see that Asterisk has
Jabber and Jingle support, but it
Have a look at the smartCID script on www.generationt.com
It allows you to have a database of numbers and override the name (and
number), flag numbers for screening, etc.
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Thursday, November 08,
Dan,
What happen w/ this? Did you figure it out? I've setup w/ XO using ESF/B8ZS,
and 5ESS for the switchtype, worked great and got the ANI as well. I dont
think you can get ANI on EM Wink trunks, how about feature group d?
-Jon
- Original Message -
From: Dan Casey [EMAIL PROTECTED]
Hi there!
Hein Maxi!
Don't hate me for this hâ?!?!?
Well, I think that the subject is just right for asterisk...
I need access a base on MS Access, to retrive some status, like just Yes or
No.
Just it!!!
I think to work with ODBC in this case, and use the app_odbc...
I try found some trip on
Hi all,
We're running AsteriskNOW Beta 6, and port 5060 is closed. We've checked it
with a port scanner and the port is definitely closed. How do we open that
port, either through the GUI or CLI. Is there any way to access the linux
command line through AsteriskNOW?
Thanks in advance!
Paul wrote:
Thank you for your answer. I am using asterisk
1.4.13 and keepalive has a value of 120 in
skinny.conf.
You can try reducing the keepAlive. The phone
will still loose registration, but will re-register
faster. Other than that, I would look at the
health of your network,
I'm not sure if anyone has done this before or not but, I was able get
the Cisco IP Communicator soft phone to work with Asterisk using SIP.
Thought I would share my experiences. The key is on the installation. To
have the software use the SIP protocol type the following command:
msiexec /i
For fastest handover disable any sort of encryption and use the same
SSID for all AP... infact I don't know how you would setup roaming
otherwise. Channels don't have to be the same, but optimize for the
best RF performance/least channel overlap.
___
I get the same response with or w/o ANI... :(
- Original Message -
From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 08, 2007 8:14 PM
Subject: Re: [asterisk-users] If caller id is null
Openser has SIP-to-XMPP Gateway and JABBER IM and PRESENCE
interconnection modules. I used the XMPP module with Google Talk and
Asterisk a while back. It resulted in a segmentation fault, but again
that was a long time ago.
Eric Chamberlain wrote:
Hello,
I’m looking for a SIP to
2007/11/8, Gilberto Nunes [EMAIL PROTECTED]:
Hi friends
I have an application that storage data in MS Access.
I need some library that can read the status of data in this base...
Some one can help me?!?!?!
Thanks
Gilberto,
what does your post has to do with Asterisk?
Anyway, probably
Asterisk Project Security Advisory - AST-2007-024
++
| Product | Zaptel|
Jon Weisman wrote:
All,
If someone calls into my asterisk box and has a private number I would
like to set the callers id to a specific telephone number, only when
the ANI is missing, otherwise if present just pass it along. Any ideas?
[incoming]
exten =
Hi friends
I have an application that storage data in MS Access.
I need some library that can read the status of data in this base...
Some one can help me?!?!?!
Thanks
--
Gilberto Nunes
Itajaí - SC
___
--Bandwidth and Colocation Provided by
Hi Dan,
Thank you for your answer. I am using asterisk 1.4.13 and keepalive has a
value of 120 in skinny.conf.
2007/11/8, Dan Austin [EMAIL PROTECTED]:
Paul wrote:
I have six cisco 7911g connected on asterisk over
chan_skinny. Four of them are working OK. two of
them even the screen on
Ah, but is EXTREMELY useful for people like me who need to know where
the call was originally destined for when it bounces off a Centrex
DMS-100 due to call forward no answer or call busy condition. I can
treat it as the DID for the original number and then send the caller
into VM or other
On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy wrote:
I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones,
on an amd_64 processor.
All goes well, the voice is clear on the remote side but in the Voip side,
where the Snom 320 is placed, I hear my voice, but don't
Can anybody give me a rough idea how much disk space is requiered for a
typical install? I want to install it in a system with solid state
storage and I don't want to buy more than I need. Would 1GB be enough?
Thanks,
--
Andres
___
--Bandwidth
Hello,
I'm looking for a SIP to XMPP Jingle voice gateway.
I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk
acts as a Jabber client.
Are there any Jabber server solutions, where Jabber users can call SIP users by
using the SIP URI and vice versa?
--
Eric
For someone that's network-aware, but hasn't sat down and plowed through
umpteen SIP-related RFC's and memorized the standards, is there a good
primer on troubleshooting SIP issues?
I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk
and my Sipura 942's, for instance...
Not
Problem solved.
You need to remove the config (local.cfg) that resides on the phone,
before restarting it.
I used the Web UI for awhile to tinker with the XML softkeys and SIP
config. What happens is that the input was set into the local.cfg. The
admin manual says that there is a precedence rule
Jason White wrote:
On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy wrote:
I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones,
on an amd_64 processor.
All goes well, the voice is clear on the remote side but in the Voip side,
where the Snom 320 is placed, I hear my
Doug Lytle wrote:
Jon Weisman wrote:
All,
If someone calls into my asterisk box and has a private number I would
like to set the callers id to a specific telephone number, only when
the ANI is missing, otherwise if present just pass it along. Any ideas?
[incoming]
exten =
I have found the new 7.x.x series firmware to be pretty much unusable in
speakerphone mode, which is slightly disappointing as I like the Snom
phones.
PaulH
On Fri, 2007-11-09 at 03:34 +0100, Philipp Kempgen wrote:
Jason White wrote:
On Thu, Nov 08, 2007 at 10:22:41AM +0100, voip crazy
I have had similar problems. My solution was to upgrade to a sangoma
a200d that has echo canellation built in. I will NEVER buy an fxo card
that doesn't have onboard echo cancellation ever again. There is just
no other way to get good sound and no echo.
-Original Message-
From:
Hi, I am getting a new sangoma t1 card soon and that will max out my
slots, which means i need to take out a card. I am going to take out my
pci network interface card (10/100)
I have an open pci-e slot i have never used in the machine so i am going
to buy a pci-e 10/100 or gigabit network
Hi,
Which is your favorite desktop wifi sip hardphone ?
I'm looking for something like
http://www.mitel.com/DocController?documentId=19401 which could be easily
moved from one meeting room to another.
(In this specific case, finding an electrical plug to power a large desktop
phone is seen more
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