29 aug 2011 kl. 15:05 skrev Kevin P. Fleming:
On 08/28/2011 01:56 AM, Tzafrir Cohen wrote:
On Thu, Aug 25, 2011 at 07:36:53PM +0100, Chris Hastie wrote:
Hi
I've just added direct support for AMI to a forthcoming version of
TBDialOut, a Thunderbird extension for dialling direct from
Hello,
We would like to share our webinar about Auto-Dialer Business Model
(Telemarketing).
It is educational video which we made for our clients and now we are sharing
it with you.
http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/
Enjoy.
NOTE: This is not attempt to sell you
Thanks Danny. Changing the ownership of the .call files seems to have
fixed the problem and I can now see that asterisk is adding a
StartRetry line to the end of the file after it makes the first call,
which it was unable to do before since the file was owned by root:
cp test5703.call
Hello,
I have been using wireshark to capture some traffic. I'm talking when the
PBX sends OK (200) connection accepted. 3Com PBX sends TZ=7200\n (an much
more things) in a SIP packet message body but Asterisk PBX sends packets
without message body, it only sends variables in the message header.
On 08/30/2011 07:36 AM, Jaime Lozano wrote:
I have been using wireshark to capture some traffic. I'm talking when
the PBX sends OK (200) connection accepted. 3Com PBX sends TZ=7200\n
(an much more things) in a SIP packet message body but Asterisk PBX
sends packets without message body, it only
On 11-08-30 05:16 AM, Kolmisoft Marketing wrote:
NOTE: This is not attempt to sell you anything. No product or service is
sold/marketed in the video.
That maybe the case, but this is still a non-commercial mailing list.
Please use asterisk-biz in the future.
--
Paul Belanger
Digium, Inc. |
On 27/08/11 10:14, Gordon Henderson wrote:
On Sat, 27 Aug 2011, Alan Lord (News) wrote:
On 26/08/11 19:02, linux guy wrote:
I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home
asterisk system.
We've been using the Siemens Gigaset 685IP range for over three years
and I'm
Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built
by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26
21:31:22 UTC]
The call flow is:
quintum gateway -- asterisk -- Dialogic IMG 1010
the call starts as a voice call, the remote fax
Hello
Up to version 1.6.0 we have been able to configure the same SIP device as a
user [inbound trunk] and as a peer [outbound trunk] w/o issues.
After we switched to version 1.8 this setup wont work, apparently one can not
have the same IP on 2 different trunks anymore. The trunk that is
On Tuesday 30 August 2011, Fabian Borot wrote:
Up to version 1.6.0 we have been able to configure the same SIP device as
a user [inbound trunk] and as a peer [outbound trunk] w/o issues. After we
switched to version 1.8 this setup wont work, apparently one can not have
the same IP on 2
yes, same thing
From: fbo...@hotmail.com
To: fbo...@hotmail.com
Subject: RE: same sip peer as user and provider
Date: Tue, 30 Aug 2011 10:35:01 -0400
yes my friend. same thing
From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: same sip peer as user and
Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue
line. The issue I am having is Avaya is sending the originating caller id
not the station id so Asterisk see that originating id so I can't route the
call correctly in Asterisk.
Thanks!
Dustin
--
Search the forum - I believe I remember a recent exchange on this subject
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dustin fails
Sent: Tuesday, August 30, 2011 10:44 AM
To: asterisk-users@lists.digium.com
Subject:
Hi,
Am Dienstag, den 30.08.2011, 09:44 -0400 schrieb Fabian Borot:
Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk
1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running
Linux on 2011-08-26 21:31:22 UTC]
The call flow is:
quintum gateway --
On Tue, 2011-08-30 at 01:31 +0200, Gilles wrote:
On Sat, 27 Aug 2011 09:31:12 -0600, linux guy linuxguy...@gmail.com
wrote:
I'm looking for an FXO device to connect to a POTS line that communicates
via USB or Ethernet.
For USB, AFAIK, there's only the one from Sangoma. All others are
both endpoints use public Ips, I just changed the real ones for the privates
ones to protect our ips but made a mistake and left the dest as a pub and the
orig as private, my bad.
but for the record, both are public IPs, there is no nat and iptables is off
also, I see that the quintum sends a
Hi people!
I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on
asterisk. On which DAHDI tells me also properly that both of my boards
are registered, one in TE and the other on in NT mode.
Calls do successfully come inside, but I want to connect my ISDN phone
at the board, but
will installing spandsp help with t.38 pass-through?
From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 11:42:41 -0400
both endpoints use public Ips, I just changed the real ones for the privates
ones to protect
On 08/31/2011 01:15 AM, Fabian Borot wrote:
will installing spandsp help with t.38 pass-through?
The only part of spandsp which is relevant to T.38 passthrough is its
modem tone detection module, and I don't think the standard Asterisk
distribution can make use of that. Some people do use it,
txs a lot for your explanation steve
so, it should work w/o spandsp fairly fine if we do not have a bad connection.
I see that this version has a lot of fixes related to t.38
but is the implementation already mature enough to guarantee a decent success
rate with fax calls?
From:
On 08/30/2011 06:32 PM, Tamer Higazi wrote:
Hi people!
I have managed to set up asterisk 1.8.5. with my 2 ISDN HFC boards on
asterisk. On which DAHDI tells me also properly that both of my boards
are registered, one in TE and the other on in NT mode.
Calls do successfully come inside, but I
Hi Patrick!
Now i got it.
I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1.
The patches are automatically integrated at Gentoo. I didn't have to
patch anything. That did the community.
Another question, I really don't like to buy a new ISDN phone with
external power connector, can I
On 08/30/2011 12:53 PM, Fabian Borot wrote:
txs a lot for your explanation steve
so, it should work w/o spandsp fairly fine if we do not have a bad
connection. I see that this version has a lot of fixes related to t.38
but is the implementation already mature enough to guarantee a decent
success
Greetings,
I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones.
With one of my sites, we're having an issue where when a call is transferred,
the MOH is not playing and all the caller is hearing is silence. The caller of
course thinks they have been hung up on, but the
It seems a reasonable likelihood that your moh at the offending site does
not match the codec of the call (IE your MOH is wav and your call codec is
SLIN). Set your verbosity and debug up to 15 and try a call to verify this.
From: asterisk-users-boun...@lists.digium.com
On 08/30/2011 08:37 PM, Tamer Higazi wrote:
Hi Patrick!
Now i got it.
I am using Gentoo Linux, Asterisk 1.8.5 and Dahdi 2.4.1.
The patches are automatically integrated at Gentoo. I didn't have to
patch anything. That did the community.
Thanks for the info.
Another question, I really
I noticed the CLI shows that the music on hold actually stops for some reason?
Here's the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28
-- Executing [s@ivr-boi-ntc-day:3] Answer(SIP/gw1-05d6, ) in new
stack
--
Your Dial command stops the MOH - if the command were Dial(SIP/1021,20,m)
the music would continue until connected or timed-out.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 2:37 PM
To:
On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
cur...@telecomabmex.com wrote:
Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.
Thanks for the tip. It looks like the smallest option is 8 FXO ports:
www.xorcom.com/telephony-interfaces/astribank-models.html
--
Well, we've taken the time to check out the wiring. It's only 3 years old and
looks like the people who did it knew what they were doing. Nice work.
Rebooting the cable modem, router, and switch didn't fix the problem.
Also, we had an instance today where ALL of the phones went down within
- Original Message -
Well, we've taken the time to check out the wiring. It's only 3 years
old and
looks like the people who did it knew what they were doing. Nice work.
Rebooting the cable modem, router, and switch didn't fix the problem.
Also, we had an instance today where ALL
On Tuesday 30 August 2011 3:14:50 pm Tim Nelson wrote:
- Original Message -
Well, we've taken the time to check out the wiring. It's only 3 years
old and
looks like the people who did it knew what they were doing. Nice work.
Rebooting the cable modem, router, and switch
Thanks Danny. I tried that but all that did is make it so when I call the site,
I get hold music instead of ringing. Still has no affect on the call transfer
MOH. :/
Interestingly, the music is playing for about 3-5 seconds before stopping
during the transfer.
I've built all of my phone
Hello List!
I have small but strange problem with ekiga 3.2.7 (on Debian Squeeze as
well as on Win XP). For every contact I have listed in ekiga that has a
corresponding hint in the dialplan ekiga tries to subscribe to the hint
but fails. In asterisk the a message like this is shown for every
Hello,
I'm using Asterisk 1.6 with Polycom SoundPoint 650, everything is running
fine except that I can't program a button on Polycom to transfer inbound
call to Voicemail directly.
I have the following in my extension.conf
exten = _547xx,1,Voicemail(${EXTEN:1}@default,u)
Reception can
Take a look at the network traffic, things like arp storms etc. A lot of
noise on the net can cause reboots. Even if you don't find anything try
turning on the storm filter (if it is not on already), its in the Settings
- Advanced- Administration - Network settings - Ethernet I think.
g
On Tue,
I'm just throwing in my 2c (I don't have polycom)
Are your phones auto provisioned then maybe the provisioning server is sending
a reboot for some reason or maybe something on the server is sending a sip
notify of reboot
-Original Message-
From: Gord Urquhart gord...@gmail.com
Sender:
Txs a lot Kevin.
I had just created and account on https://issues.asterisk.org/jira
Let me know if this is the right place to post both the pcap capture and the
sip logs. If not please help me out creating the account in the right place so
that I can provide all the information you need.
The sip
- Original Message -
From: Mike Diehl mdi...@diehlnet.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 30, 2011 5:13:22 PM
Subject: Re: [asterisk-users] Polycoms rebooting themselves
Well, we've taken the time to check out the wiring. It's only 3
years old and
looks
39 matches
Mail list logo
