A password prompt is avoidable with a ",s" in the VoicemailMain appdata
Robert Berlin
Manager of Operations & Systems Development
Florida High Speed Internet
(321) 205-1100 x109
From: "D'Arcy J.M. Cain&qu
this
helps!
VoicemailMain(${SIPPEER(${CHANNEL(peername)}:mailbox)},s)
Robert Berlin
Manager of Operations & Systems Development
Florida High Speed Internet
(321) 205-1100
From: "D'Arcy J.M. Cain" <da...@vex.net>
Se
"timing test" does similar, it just doesn't do the automatic calculation.
Confbridge normally operates at a mixing interval of 20ms, which is 50 ticks
per second. That would be what you would want to test.
If you don't get 50 per second then that means ConfBridge will not provide a
steady
g the timerfd module.
Regards
Robert McGilvray
SS GlobeOp
Associate Director, IT Network Security
GlobeOp Financial Services | 1565 Front Street | Yorktown Hts NY 10598
t: +1 (914)-293-3584 | f: +1 (914)-293-3510
rmcgi...@globeop.com | www.ssctech.com<http://www.ssctech.com/> |
> Are you selectively loading modules? If so you need the new res_pjproject.so
> loaded.
Yes. That did it, thanks.
Bob
This email with all information contained herein or attached hereto may contain
confidential and/or privileged information intended for the addressee(s) only.
If you have
ad.so.0 (0x7fe0138ca000)
libc.so.6 => /lib64/libc.so.6 (0x7fe013509000)
libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x7fe0132f2000)
/lib64/ld-linux-x86-64.so.2 (0x7fe0169ef000)
Regards
Robert McGilvray
o: 914 293 3584
From: asterisk-users-boun...@lists.digium.com
Hello,
We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe
and chan_sip for conferences. I have been testing the new versions of Asterisk
with PJSIP and ConfBridge but have run into an issue which is preventing us
from moving forward. Everything works fine until a
MUST be marked as
sendonly or inactive in the answer.
"
Is this a bug or am I wrong in my interpretation of the dialog?
Thanks!
Robert McGilvray
o: 914 293 3584
From: Robert McGilvray
Sent: Thursday, March 17, 2016 12:55 PM
To: 'asterisk-users@lists.digium.com'
Subject: Hold/Resum
.
So the ability to use DPMA with Asterisk RT is very important for our
large deployments.
Anyone willing to contribute towards a bounty for this feature?
--
Robert Broyles
On 5/7/15 7:14 AM, Matthew Jordan wrote:
On Fri, May 1, 2015 at 10:43 AM, Robert Broyles
rob...@webservicesaz.com
We love our Digium phones and DPMA - but we really need it to work on
our Realtime Platform. Otherwise we lose all the cool features and they
are just standard SIP phones.
Anyone working on a solution for this? Or anyone from Digium see this on
the roadmap?
--
Hi all,
on my atserisk box call recording and cdr doesn't work. In the log files I
have a strange entry - does this have something to do with that?
Version: Asterisk 13.1.0
Host: debian wheezy 7.7
Thanks a lot for a brief hint .
Walter.
***
[2015-Jan-26 11:34:04]
To you as well.
On Dec 25, 2013 8:56 AM, Nick Cameo sym...@gmail.com wrote:
God Bless and Merry Christmas to All!
Nick.
--
_
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New to Asterisk? Join us
https://bbs.archlinux.org/viewtopic.php?pid=920549
On Sat, Jun 8, 2013 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote:
When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
I get a motorboating sound or warble - or - just not clear audio.
When I switch that to ALSA direct
Pulse Audio 4.0 just came out and has gotten good reviews as it improves
audio quality...I installed it on the devel and support mediaports and will
test tomorrow.
http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/4.0/
On Mon, Jun 10, 2013 at 7:59 PM, Robert Krakora
rob.krak
I believe there are options for rtp / audio..
Look at pcap play and rtp echo...
Transcoding would be another beast - if you are allowing it
Sent from my iPhone 5
On May 22, 2013, at 10:02 AM, Tommy Cooper tomcoope...@yahoo.com wrote:
From the little experience I have I do not think that that
I am having the same problem with Asterisk 11.2.0 and Linphone and it is
exactly 15 minutes and occurring with SIP running on our LAN.
On Thu, Mar 21, 2013 at 3:31 AM, Florian Wolters flor...@florian-wolters.de
wrote:
Hi @ll,
I just moved my Asterisk Box and changed the Provider and Internet
Hi,
If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have
tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC
video from one machine to another machine running Linphone. Contact me at
this e-mail address robkrak...@messagenetsystems.com for source
Might also want to check the google hasnt detected an unusual login and is
asking for the ip to be accepted.
Log in to gmail with that account and check
Sent from my iPhone 5
On Feb 5, 2013, at 4:31 PM, Joshua Colp jc...@digium.com wrote:
Josue Freitas wrote:
Thank you!
What about the
On 18 Jan 2013 15:22, Klaus Darilion klaus.mailingli...@pernau.at wrote:
Hi!
I want to forward a call to another destination if the outgoing call leg
has an rtptimeout. But as far as I see there is no way to find out if the
hangup was due to a rtp timeout or any other reason. I thought that
Sometimes just the act of collecting performance data degrades the quality
Sent from my iPhone 5
On Jan 6, 2013, at 6:00 AM, XBrian bobo...@yahoo.co.uk wrote:
Thanks
What would you use to measure jitter / packetloss in real time?
--
Good luck! Finding the right person at VZ has always been a beef of mine
Sent from my iPhone 5
On Jan 5, 2013, at 11:12 AM, Logan Bibby lo...@keobi.com wrote:
Does anyone have a good contact for their sales? I've attempted calling their
Enterprise sales a few times and was just spun around
Asterisk sip show peers lists the qualify value in ms (milliseconds).
Please read up on this and the setting for it in sip.conf config file
Sent from my iPhone 5
On Jan 5, 2013, at 5:30 AM, XBrian bobo...@yahoo.co.uk wrote:
Joachim, thanks for the reply
- delay you can somewhat estimate
Wow! Thanks so much for all the information. I now have a lot to look over.
Bob R
On 01/02/2013 10:03 AM, Tzafrir Cohen wrote:
On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote:
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
apt
Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
Bob R
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New to Asterisk? Join us for a live introductory
I again would recommend a more thorough explanation of the configs
I've been using Asterisk for years - but the configs for this need some
explanation in the wiki
The samples contradict what the wiki has.. And as I indicated I could
not get audio working...
On 10/15/12 10:11 AM, Joshua Colp
.
I configured the config files as per Digium wiki.
Maybe there isn't many people who have tried out Motif yet.
Here's hoping that the old school way on Ast10 will be reliable
On 10/10/12 3:48 PM, Joshua Colp jc...@digium.com wrote:
Robert wrote:
My apologiesŠ I will clarify the situation.
We
Just installed 11 and trying to get MOTIF / XMPP working to E164/PSTN
number.
We can get ring and a connected call but no audio
SIP = ASTERISK = MOTIF
Is there any specific configurations for getting audio to work?
--
_
--
PM, Joshua Colp jc...@digium.com wrote:
Robert wrote:
Hola,
Please in the future don't cross post as you have done to both the
developer list and users list. If it's not related to development of
Asterisk the users list is where it should stay.
Just installed 11 and trying to get MOTIF / XMPP
Edgewater 4350 or cheaper vigor 2910 dreytech
On Tue, Oct 9, 2012 at 3:24 PM, Mike Diehl mdiehlena...@gmail.com wrote:
I hope no one considers this off topic...
I have a phone customer who wants 2 Internet connections so that if one
goes down, he can use the other for phone service.
So,
Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom
PSTN gateway - pstn line to telcoi'm using xlite for windows
when I make a phone call (sip - outgoing channel),I can hear my own voice
so clear. it's very annoying mewhen talking a little loud... any solution?
Sorry for my last post,
Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom
PSTN gateway - pstn line to telcoi'm using xlite for windows
when I make a phone call (sip - outgoing channel),I can hear my own voice
so clear. it's very annoying mewhen talking a
Hi all,
Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom
PSTN gateway - pstn line to telcoi'm using xlite for windows
when I make a phone call (sip - outgoing channel),I can hear my own voice so
clear. it's very annoying mewhen talking a little loud... any solution?
Smokeping with sip probe is quite nice
Sent from BETA iOS6
On Jul 13, 2012, at 4:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
On 12-07-13 08:37 AM, Mike wrote:
On 12-07-13 06:00 AM, Elliot Murdock wrote:
Hello,
Which tools are recommendable for monitoring VOIP, bandwidth,
Thanks whoever is running an auto response ticket system!
Look forward to getting more spam from you!
Sent from BETA iOS6
On Jul 13, 2012, at 4:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
On 12-07-13 08:37 AM, Mike wrote:
On 12-07-13 06:00 AM, Elliot Murdock wrote:
Hello,
Linksys firmware?
I've had issues with older firmwares and VoIP
Sent from my iPhone 4S
On Feb 11, 2012, at 1:05 PM, David Woodfall d...@dawoodfall.net wrote:
On (17:38 11/02/12), David Woodfall d...@dawoodfall.net put forth the
proposition:
On (16:48 11/02/12), David Woodfall
I run two off virtuozo vps boxes - but capacity will always be the defining
value
Sent from my iPhone 4S
On Feb 10, 2012, at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote:
Hello everybody
someone in this list, has installed asterisk, in a virtual server like
proxmox? I'm thinking
Here's what I do... Changed some variables for obscurity. 911 is the inbound
#... exten 6000 rings to SIP/TEST
exten = 911,1,GotoIf(${BLACKLIST()}?blacklisted)
exten = 911,n,Macro(stdexten,6000,SIP/test)
exten = 911,n,Playback(transfer,skip)
exten = 911,n(blacklisted),Goto(blacklisted,s,1)
Take a look at Blacklist
I love that command and love to send nice intercept messages to the other
side J
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Thursday, December 29, 2011 8:40 PM
To:
Right check out Cordia.LT
Sent from my iPhone 4S
On Dec 19, 2011, at 9:58 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org
wrote:
On Tuesday 20 Dec 2011, Steve Edwards wrote:
On Mon, 19 Dec 2011, Nick Khamis wrote:
SIP in India is illegal.
What about IAX, Skype, VPN, etc?
The only
Are you using FreePBX or another packaged Asterisk?
Sent from my iPhone 4S
On Dec 12, 2011, at 9:23 AM, silent sayz silent.s...@gmail.com wrote:
hello ,
I have been working hard to solve the issue of custom CDR in the Asterik with
Mysql but in vain.
I searched google for complete 2
Agreed. And facilities based CLEC even scarier.
Regulatory / billing / PUC legals etc ugh
Sent from my iPhone 4S
On Nov 14, 2011, at 8:33 PM, Alex Balashov abalas...@evaristesys.com wrote:
Worst reason to become a CLEC: improved cost structure. Or, to be precise,
it is a counterfactual
/2011 08:36 PM, Robert-IPhone wrote:
Agreed. And facilities based CLEC even scarier.
I'm curious what sort of thing would be considered a non-facilities based
CLEC, since UNE-P was cancelled in 2003.
There are some non-interconnected CLECs out there that exist for the sole
purpose
Also used for calling card platforms :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andreas
Sikkema
Sent: Thursday, October 06, 2011 5:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
I am adding dickish to my dictionary - thats a hot one!
Sent from my iPhone
On Sep 25, 2011, at 4:41 PM, Alex Balashov abalas...@evaristesys.com wrote:
On 09/25/2011 02:23 PM, Bruce B wrote:
Stop wishing for that. I like Asterisk and I will raise a voice
when I feel uncomfortable with
Sounds like a great idea.. Hopefully the page/account never gets hacked and
bad IP's published.. I could see a great hack of
127.0.0.1
192.168.0.0/16
10.0.0.0/8
getting up there somehow and next thing you know - BAM!
But I haven't RTFM - I'm guessing there is probably a white list
://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Robert
--
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I'm using them for inbound and outbound on Asterisk and FreeSwitch
Sent from my iPhone
On Sep 13, 2011, at 5:14 PM, Danny Nicholas da...@debsinc.com wrote:
That’s what this part of extensions.conf should do:
; inbound context example for your DID numbers, do not add the number 1 in
front
I personally would never install a GUI o/s. By doing so you always open
yourself up to more security concerns.. Packages / ports / etc.
Course one might argue - it's behind a firewall..
In my professional experience with running numerous ISP and VoITSPs the rule
has always been install the
Of linux guy
Sent: Monday, September 12, 2011 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init
level 3 or init level 5 ?
On Mon, Sep 12, 2011 at 12:24 PM, Robert Huddleston rhuddles...@gmail.com
wrote:
I
Asterisk is a company? This is news to me
Sent from my iPhone
On Sep 12, 2011, at 5:35 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote:
On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro stot...@asteriskhelpdesk.com
wrote:
See comments inline.
On Mon, Sep 12, 2011 at 2:21 PM, linux
www.buildityourself.org
:)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Friday, September 09, 2011 2:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Reporting for
what do you mean? Like speed dial or directory?
Sent from my iPhone
On Sep 4, 2011, at 6:47 PM, neo haux neo.h...@gmx.com wrote:
Hi,
It is possible to save all the phones numbers on asterisk servers instead of
doing so manually in each VoIP device ?
Does SIP take care of such
Hi
I'm using asterisk 1.6.2.18.1
I'm having a problem where only the first four digits are collected in the
cdr when the call is dialed overlap but if the call is dialed en-block the
whole dialed digits are recorded
chan_dahdi.conf
[trunkgroups]
[channels]
language=uk
switchtype=euroisdn
Have you looked at pin sets in freepbx / trixbox / elastix? I haven't tested
it myself - but I know the feature is present there
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps
Sent: Friday,
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:
Discussion
Subject: Re: [asterisk-users] 1.8.5 Voicemail duration incorrect
Hi,
Am Mittwoch, den 24.08.2011, 13:18 -0400 schrieb Robert Huddleston:
Anyone else seen this?
I saw a jira but was in feedback status..
I just checked with a voicemail of 60 seconds. It was reported
in .txt-file
Anyone else seen this?
I saw a jira but was in feedback status..
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New to Asterisk? Join us for a live introductory webinar every Thurs:
This is off topic...
Asterisk will not provide you with the ability to SMS random cell phones.
Being able to transport the SMS yourself is a grewling process.. Look at
software like Kamel...
Basically you have three options:
( a ) cheat and use the email method - i.e. determine everyone's
Seriously Again?
This is off topic...
Asterisk will not provide you with the ability to SMS random cell phones.
Being able to transport the SMS yourself is a grewling process.. Look at
software like Kamel...
Basically you have three options:
( a ) cheat and use the email method - i.e.
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones
Robert.
Thanks
Anyone have any testing experience with T38 and HT-502 Grandstream?
I just want to confirm that t.38 is working on this device.
Thanks
--
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New
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T38 Fax with Grandstream HT-502
On 08/01/2011 12:02 PM, Robert Huddleston wrote:
Anyone have any testing experience with T38 and HT-502 Grandstream?
I just want to confirm that t.38 is working on this device.
You'd be more likely
- Non-Commercial Discussion
Subject: Re: [asterisk-users] T38 Fax
On 2/08/2011 1:02 AM, Robert Huddleston wrote:
Anyone have any testing experience with T38 and HT-502 Grandstream?
I just want to confirm that t.38 is working on this device.
Thanks
Yes, it works.
I currently
hard to equate sip attack to ping performance.. Run mtr for a bit.
Also try tcpdump or wireshark or tethereal.
If you are really paranoid recycle all your passwords
Sent from my iPhone
On Jul 31, 2011, at 7:04 PM, Dave George dgeo...@teletoneinc.com wrote:
My asterisk server is getting bogged
Personally I like to just hook up an old ghetto blaster / boombox to the
line in port on my sound card :)
Kidding aside - I think audio quality for MoH is not always going to sound
as nice as you might want.
I mostly stream online radio over my MoH and the quality is not the
greatest.
Maybe
I like Xen. It's free and rock solid. VMWare is great but their money
greedy.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Wednesday, July 27, 2011 9:30 AM
To: asterisk-users@lists.digium.com
Subject: Re:
Maybe write a cludgy init.d script / bash. I.e. make a file somewhere called
rebootflag. set it to 1 after you issue a shutdown -r -n now. then check it
in init.d script.
Pseudo code
In init.d / startup scripts
If /etc/manualreboot = 0 or file not found
echo 1 /etc/manualreboot
...@lists.digium.com] On Behalf Of Robert
Huddleston
Sent: Wednesday, July 27, 2011 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Lightning and thunder
Maybe write a cludgy init.d script / bash. I.e. make a file somewhere called
rebootflag. set it to 1
gerbals
Sent from my iPhone
On Jul 27, 2011, at 5:32 PM, Hans Witvliet h...@a-domani.nl wrote:
On Wed, 2011-07-27 at 09:44 -0400, Claude Hayn wrote:
We are frequently losing power during lightning storms. (Yes we have
UPS, but often by the time power comes back up the UPS has run out of
Also consider the setting localnet in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Tuesday, July 26, 2011 9:24 AM
To: asterisk-users@lists.digium.com
Subject: Re:
Such a pointless argument. The same problem can happen on any voip platform
including freeswitch.
Again it's a knowledge thing.
BTW if you were paying attention to your logs or practiced good admin skills
you would have seen the attacks and stopped them.
I swear by fail2ban and other hardening
When I get hacked I typically run a rootkit checker
http://www.chkrootkit.org/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace
Sent: Thursday, July 21, 2011 2:18 PM
To:
I prefer
How do we do that? Isn't Asterisk a SIP Proxy ;)?
That's a good question...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, July 19, 2011 2:18 PM
To:
Boy if only it was Enron :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, July 18, 2011 8:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Requires
First
Subject: Re: [asterisk-users] Requires
On 07/18/2011 09:00 AM, Robert Huddleston wrote:
Boy if only it was Enron :)
Baby steps. Success is not built overnight; you have to work your way
up the totem pole of fleecing people. Start small: persistently ask
basic, RTFM-grade newbie questions while
wrong address - but I can come Monday if you like ;)
Sent from my iPhone
On Jul 16, 2011, at 8:58 AM, mahesh katta maheshka...@flexydial.com wrote:
Dear Ashirwad,
Please make ready below things for demo in pune .MONDAY needs to be ready for
test in our office.
1. PRI card single span
2.
I stand amused that people want to experiment with VoIP and Asterisk - but
aren't willing to:
( a ) Read wiki / manuals / faqs
( b ) demand packages for their o/s
This ain't windows folks :)
./configure
make
make install
Is really simple :)
-Original Message-
From:
On 07/12/2011 08:26 AM, Kevin P. Fleming wrote:
It is unknown whether it will continue to be usable after that period;
Skype has the ability to disable SFA from accessing the Skype network
if they feel that is what they want to do. Since it won't get any
updates between now and then, it is
Read the wiki / manuals
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup
Sent: Tuesday, July 12, 2011 11:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CDRs
Hi
Like we
+1 for Xen
-1 for VB
Sent from my iPhone
On Jul 8, 2011, at 10:00 PM, Doug Lytle supp...@drdos.info wrote:
Warren Selby wrote:
Not trying to start a war here,
That may be, but I have experience with VB.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty
/0/1/31 (master=TE4/0/1/31)
Im using TE410P
dahdi-linux-2.4.1.2.tar.gz
dahdi-tools-2.4.1.tar.gz
--
Robert
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
On Mon, Jun 20, 2011 at 6:10 PM, Robert-iPhone rhuddles...@gmail.com
wrote:
You are supposed to go via cisco and support contract BUT Google is your
friend
Hahahah Baltimore and SE DC. How about Philly too J
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Tuesday, June 21, 2011 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
wow I think someone needs to just spend some time reading and playing. Getting
these phones working is not rocket science and there are similarities with how
to do firmware / config pushes.
Not to sound mean but RTFM
Sent from my iPhone
On Jun 21, 2011, at 7:45 PM, Warren Selby
I'm using the sip firmware.. It's alright.. I feel like I'm not receiving
all the features I should.. But MWI works and multiple call appearance..
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, June 20,
You are supposed to go via cisco and support contract BUT Google is your
friend (JFGI)
Sent from my iPhone
On Jun 20, 2011, at 6:44 PM, bilal ghayyad bilmar...@yahoo.com wrote:
If I need to use SIP, from where to get the suitable firmware for these Cisco
IP Phones 7942G?
Where do u
both show transfercapability DIGITAL
Regards
Robb
On 16 June 2011 23:40, Richard Mudgett rmudg...@digium.com wrote:
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
any reason why this would happen, should I report a bug on the issue
tracker?
Robb
On 17 June 2011 19:55, Richard Mudgett rmudg...@digium.com wrote:
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten =
I'm using ISDN30 for a bridged application
in all the old versions of asterisk the time slot number is shown in the
channels and dstchannel fields of the cdr
I understand this has chaned in 1.8,is there a way of getting the time slot
information stored somewhere at the end of the call so this
Anyone have recommendations for a gateway / ATA for business that can do
GroundStart? Preferably with an rj-21 - but okay if not..
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, Robert Huddleston rhuddles...@gmail.com
wrote:
Anyone have recommendations for a gateway / ATA for business that can do
GroundStart? Preferably with an rj-21 - but okay if not..
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- but if it only allows T1/E1 for WAN - I'm shot.
From: John Novack [mailto:jnov...@stromberg-carlson.org]
Sent: Tuesday, June 14, 2011 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Robert Huddleston
Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway
Robert
I'll have to look at that then - as I thought the card actually said Ground
Start on it.. I may have missed or it was scratched off the word loop start
From: John Novack [mailto:jnov...@stromberg-carlson.org]
Sent: Tuesday, June 14, 2011 5:20 PM
To: Robert Huddleston
Cc: 'Asterisk Users
not be
available to you or outside your budget window
How does this relate to Asterisk, or does it?
John Novack
Robert Huddleston wrote:
I’ll have to look at that then – as I thought the card actually said “Ground
Start” on it.. I may have missed or it was scratched off the word loop
sip trunking can easily turn into a
PITA.
John Novack
Robert-iPhone wrote:
considering providing the sip trunking nyself via asterisk.
the sip trunking looks expensive - card and licenses from nec.
Sent from my iPhone
On Jun 14, 2011, at 6:06 PM, John Novack jnov...@stromberg
I also had trouble w/ these phones at first. There was a DHCP option (?81?)
you'll have to google it.
The phones would not talk to tftp until I set dhcp option.
The console aux cable is easy to build and VERY useful
Sent from my iPhone
On Jun 13, 2011, at 8:31 PM, Mark Engelhardt
the From and to tag.
Asterisk 1.6.2.9 and 1.8.5rc1
Thanks,
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Robert
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by asterisk.
Thanks,
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Robert
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asterisk-users
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http://lists.digium.com/mailman/listinfo/asterisk-users
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Robert
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