Is there a way to divert incoming calls on DAHDI T1 channels so telco gets
the diversion and send the call to new number and releasing the channel?
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New to
is there anyway to change Sip headers in local channels?
if a user sets forward on their handset, calls coming in to the handset get
diversion header added:
Diversion: "202" ;reason=deflection
Then asterisk sends the call to local channel:
- Now forwarding SIP/201-0483 to 'Local/33@tes
yes, thanks you!
On Sat, Mar 22, 2014 at 9:13 AM, Paul Belanger wrote:
> On Fri, Mar 21, 2014 at 11:58 PM, Al lists wrote:
> > looking more into this, looks like this is not a issue, its related to
> users
> > changing voicemail password from handset, asterisk rewrites the
looking more into this, looks like this is not a issue, its related to
users changing voicemail password from handset, asterisk rewrites the file.
On Fri, Mar 21, 2014 at 9:31 PM, Al lists wrote:
> passwordlocatio seems to be related to vmsecret
>
> from voicemail.co
; On Fri, Mar 21, 2014 at 3:22 PM, Al lists wrote:
> >
> > We noticed issues with voicemail and somehow looks like voicemail.conf
> has
> > been overwritten:
> >
> > ;!
> > ;! Automatically generated configuration file
> > ;! Filename: voicemail.conf (/e
We noticed issues with voicemail and somehow looks like voicemail.conf has
been overwritten:
;!
;! Automatically generated configuration file
;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
;! Generator: AppVoicemail
;! Creation Date: Thu Mar 20 06:48:16 2014
;!
i saw a bug for 1.4 an
i noticed in asterisk 10.12.3, i get messages like this:
[2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite:
Failed to authenticate device 305;tag=0d516e63
but not mentioning attacker ip (to be used for fail2ban)
is this expected?
--
_
I'll check this option and see if it helps next time,
just to clarify, there were no actual calls in place, just DOS register
attack.
On Wed, Jun 1, 2011 at 12:22 PM, Ira wrote:
> At 10:56 AM 6/1/2011, you wrote:
>
> Do you have:
>
> sip.conf
> [general]
> allowguest=no
>
>
> So because of thi
Hi List
Recently i have noticed this attack on couple of servers,
usually a foreign IP starts sending tons of register request without any
answer to authentication,
if you type sip show channels in cli you will see tons of these:
1.2.3.4 (None) 2389603298 00101/1 0x0 (nothing)N
caller id instead of overwritten caller id of
asterisk.
hope it makes sense
On Sat, Mar 12, 2011 at 5:47 PM, C F wrote:
> Reinvite its called
>
> On Sat, Mar 12, 2011 at 1:22 PM, Al lists wrote:
> > is there a way to have asterisk to flash transfer the call and not being
> in
> &
is there a way to have asterisk to flash transfer the call and not being in
sip path anymore?
for example, a sip trunk send a call to asterisk, asterisk rings a handset,
then sends/flash hooks the call to a cell phone through sip trunk, and not
being in path anymore?
--
I'm a long time user of Digium carts and stupid me i wanted to give Sangoma
a try.
We got Sangoma A400 with 6 FXO ports.
Asterisk version: 1.4.35
Zaptel version: 1.4.11
Wanpipe version: 3.5.11
we tried to use fxtune but looks like it wont work with Sangoma card, (
please correct me if i'm wrong)
On 7/30/09, Steve Totaro wrote:
> The first time is always free :)
>
> On Thu, Jul 30, 2009 at 1:50 PM, John Todd wrote:
>
>>
>> I know many of you have been waiting for this for a while, so I'll
>> keep this short: The Skype for Asterisk Public Beta is now available
>> on the Digium store.
>>
>
Foundry serverIron does support SIP and its ASIC not a linux box Load
balancer like F5,
Refer to Chapter 10 (page 677) of ServerIron manual.
It explains everything in detail.
Also you may need to play with source nat a little bit to make your specific
configuration work, but it should work, at leas
yes, make sure context line in general area has a dummy context, something
with one line to hangup.
On Fri, Nov 28, 2008 at 12:56 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:
> On Fri, Nov 28, 2008 at 11:00 AM, Mike wrote:
> > I was looking at my CLI the other day, and found a lot o
--
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Al lists
> *Sent:* Dienstag, 09. September 2008 23:40
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Asterisk REFER
>
>
>
>
Hi All,from what i'm understanding, Asterisk is back to back user agent.
Base on this my initial thought was even if we enable reinvite in sip.conf,
asterisk still will be in sip path after transfer.
But i read some information in asterisk using refer to transfer a
call completely to another sip or
last time i had this issue with teliax, they recommended to upgrade to 1.4
On Fri, Aug 29, 2008 at 3:44 AM, Chris Mason <[EMAIL PROTECTED]> wrote:
> I tried DTMFmode=auto and it did not help. Any further ideas?
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScan
While this is in place,
how about sip show channels and show channels ?
On Fri, Jul 25, 2008 at 4:56 AM, Atis Lezdins <[EMAIL PROTECTED]> wrote:
> On Fri, Jul 25, 2008 at 2:59 AM, Al lists <[EMAIL PROTECTED]> wrote:
> > I noticed that i' m not getting any manager ev
I agree, No manager gets fired even if a Cisco Call Manager goes south.
that's not the case with Asterisk.
With limited experience that i have with both, i hit more bugs using
Asterisk than a CCM, but this is not relevant to your final answer.
If you can afford CCM, and you can live with less flexi
I noticed that i' m not getting any manager event for hold and unhold of a
channel.
is this normal?
Also is there any easy way through either CLI or manager to find out which
one of the channels are on hold?
I checked "show channels" that did not show a channel being on hold or not,
also "sip show
If you are trying to reject an IP address to connect to asterisk, there is
no need to run iptables.
Each SIP definition in sip.conf can have:
deny=0.0.0.0/0.0.0.0
permit=192.168.135.1/255.255.255.0
just set these values and it wont accept anything from that IP.
On Mon, Jul 7, 2008 at 7:37 PM, Do
i used it on one server a little while ago.
my primary use was ability to show each user's status on spark.
i did not get consistence results, phone status was not accurate.
and did not try it after that, maybe its fixed in newer versions.
On Fri, Jun 20, 2008 at 2:44 PM, Julian Lyndon-Smith <[EM
anyone has used or bough one?
would appreciate comments.
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any of you guys have used FOP for drag and drop transfer on 30 40 phones
environment?
how stable is that?
I'm playing with it but so far drag and dropping phone icon to another phone
disconnectes the call.
On Wed, Apr 16, 2008 at 2:02 PM, Lee Jenkins <[EMAIL PROTECTED]> wrote:
>
Hi list,
Any good drag and drop transfer call application for windows based systems
you can advise ?
Something like HUD perhaps?
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I have seen this issue on both 1.2 and 1.4, was not able to reproduce to
find a cause or bug.
I have seen this after power failure boot up.
show sip peer command shows most of peers, except one or two (in my cases
trunk) .
if i issue a sip reload command, it will show all of them.
I can write a scr
Thu 4/3/2008 10:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Need some input for Quad T1 and channel
> banks.
>
>
>
> Just Google Quintum Tenor AX. Well worth the money.
>
> Thanks,
> Steve Totaro
>
> On Mon,
Atcom supports IAX:
http://www.voip-info.org/wiki/view/AT-530
On Sat, Apr 5, 2008 at 11:17 AM, Joseph <[EMAIL PROTECTED]> wrote:
> On 04/05/08 05:16, bilal ghayyad wrote:
> >Hi All;
> >
> >Till now I am not able to find a good IAX IP Phone or
> >Gateway that can be used with good quality.
> >
>
you can see users status in Jaber,
Install Open fire Jabber server with Asterisk pluging.
On Thu, Apr 3, 2008 at 1:55 PM, Earl Terwilliger <[EMAIL PROTECTED]> wrote:
> On Thursday 03 April 2008 02:59:07 pm faraz wrote:
> > FOP is quite clunky!
>
> one reason i wrote the event montor... which is
Re: [asterisk-users] Need some input for Quad T1 and channel
> banks.
>
>
>
> Just Google Quintum Tenor AX. Well worth the money.
>
> Thanks,
> Steve Totaro
>
> On Mon, Mar 31, 2008 at 10:03 PM, Al lists <[EMAIL PROTECTED]> wrote:
> > Im guessing T1cas
; On Mon, Mar 31, 2008 at 6:01 PM, Al lists <[EMAIL PROTECTED]> wrote:
> >> I'm looking to install a system with 80 FXS analog phones.
> >> At this time the only cost effective solution is using a 4 port T1 card
> and
> >> addit 600 channel bank.
> >&g
Its Nice, i agree, but we are looking at $4k to $5k with this.
On Wed, Apr 2, 2008 at 1:17 PM, Andrew Latham <[EMAIL PROTECTED]> wrote:
> Here I will say it http://xorcom.com
>
>
>
> On Mon, Mar 31, 2008 at 6:01 PM, Al lists <[EMAIL PROTECTED]> wrote:
> >
If you are asking about dial command on analog lines, here is what i do :
exten => _NXX,1,Dial(ZAP/g1/ww${EXTEN})
that should give you 2 seconds before actually start dialing, its good way
to wait for analog lines to stabilize first before dialing.
On Tue, Apr 1, 2008 at 9:49 PM, Pete Kay <
Im guessing T1cas not PRI,just because its giving 24 fxs per T1.
Steve, what are my options for SIP to fxs?
thank you!
On 3/31/08, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Don Pobanz wrote:
> > Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
> >
> >>
> >
> > This does not sound right. If it is 2
I'm looking to install a system with 80 FXS analog phones.
At this time the only cost effective solution is using a 4 port T1 card and
addit 600 channel bank.
Has anyone tried this solution? any good documents beside
http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check
Nope,
Coded is Ulaw on both sides and also this issue happens occasionally with no
change.
On Wed, Mar 26, 2008 at 6:17 PM, Adrià Vidal <[EMAIL PROTECTED]> wrote:
> Seems a codec problem, check the sip.conf from that spa942
>
> On Wed, Mar 26, 2008 at 11:59 PM, Al lists &l
I'm getting "Got SIP response 406 "Not Acceptable" back from 10.0.1.2"
occasionally when try to dial to SPA942 ,
anyone has any idea on this before i consider Firmware upgrade?
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Simple, add new interface in your system and put BOOTPROTO=dhcp in
ifcfg-eth1
if you have one gateway you can add that in the same file or in
/etc/sysconfig/network,
or if you have multiple gateways, you need to define a route to your voip
service through that interface.
On Tue, Mar 18, 2008 at 7
Or maybe you can show him some links ;)
Try this for send mail:
http://docs.snake.de/smtp-auth.html
this is very common these days and to make it more fun each mailserver
(provider) has their own criteria to decide if your email is spam or not.
to give you and example:
make sure you are using stat
i was reading posts on wiki and noticed lots of posts about Avaya 4610
handset having issue with MWI,
Anyone has any more updates?
Is this still the case?
Any good tutorial for configuring these phones and Asterisk?
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Has anyone checked asterisk with check_udp plug in?
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Always rely on free -m to see how much free memory you have not top.
in terms of memory leak, i have asterisk running on servers with uptime of
400 days (CentOs), if there was any leak, i'm guessing i would have crashed
server long time ago.
On Thu, Feb 14, 2008 at 4:23 PM, Doug Bailey <[EMAIL PRO
Just wondering how your experience is with HPEC,
Is it just for analog interfaces or we can use it on TE122 as well?
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check here:
http://www.voip-info.org/wiki/view/Asterisk+cmd+ParkAndAnnounce
On Feb 6, 2008 4:22 PM, Tim Nelson <[EMAIL PROTECTED]> wrote:
> Could you possibly post what steps you took to make this work so others
> (including myself :-) ) may benefit? Thank you!
>
> Tim Nelson
> Systems/Network S
wrote:
> On Sun, 2008-02-03 at 22:42 -0700, Al lists wrote:
> > Theoretically, setting TOS value ( these days called DSCP) wont change
> > anything in switch behavior, unless you are using Layer 3 switches.
> > What makes a difference in a switch is COS bits, and i'm n
Theoretically, setting TOS value ( these days called DSCP) wont change
anything in switch behavior, unless you are using Layer 3 switches.
What makes a difference in a switch is COS bits, and i'm not sure how
asterisk sets that.
I guess to be safe, you would need to create 2 VLANS and in the switch
Is there any way to have Asterisk call an extension in dial plan instead of
original extension after timeout?
Like extension A puts the caller in parking lot, he leaves the phone and
forgets about it, instead of having that phone rings after timeout, have a
group of phones rings.
__
I have been using Dell servers and have no issues with linux, in fact when i
implemented my last install with their top of the line server (dual xeon
quad core and SAS drives on Perc 6i) i was amazed how smoothly it went
trough.
Beside that i like their open manage, it runs nice on linux and its a
Thank you Paul!
Its impressive!
On Jan 23, 2008 4:55 PM, Paul Hales <[EMAIL PROTECTED]> wrote:
>
> http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm
>
> It was the bottom news item on voip-info.org - I was worried I would have
> to really search for it!
>
> later,
>
> Pau
Yes, this prompt will shows up on SIP 2.2.2 as well.
I never had any issues with this though, it will clear up after next
registration of phone.
I just downloaded SIP 3.0 and have not got a chance to check and see if it
happens with this firmware as well.
On Jan 22, 2008 2:53 PM, Steve Johnson <[
Hello all,
is there any way to tell asterisk what port to use for source of any
registration request?
for example the simple register command,
register => user:[EMAIL PROTECTED]:port
will send the register packet from asterisk_IP:5060 to proxy:port .
Is there anyway to have asterisk to use differen
Cool!
I didnt know Fedora has Asterisk in their repository. Nice!
On Jan 5, 2008 4:26 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, Jan 05, 2008 at 03:15:13PM +0530, Bhrugu Mehta wrote:
> > hi, all
> > i want to create cd-rom with asterisk. how it possible.
> > when i put disk in cdrom
Guys!
what i was looking here was a simple hint/recommendation for installing
IaxModem and Hylafax.
Let me try it myself and see how feasible this solutions is.
On Jan 1, 2008 5:02 PM, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Jonn R Taylor wrote:
> > I have always said that if some one said
I'm not looking at T.38 , at this time its terminating a SIP trunk with
multiple DID's for fax.
I'm using this configuration with linksys PAP ATA and satisfied with
results.
I'm looking at removing these ATA 's and using Asterisk ( or giving it a try
) for terminating fax.
> >>
> >>> Last time I
at this time is terminating a SIP trunk,
each DID will get its own fax box.
I guess at this time i'm looking to find a tutorial for installing iaxmodem
and hylafax as it seems to be the answer.
On Dec 31, 2007 9:11 PM, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
> On Dec 28, 20
thank you all, still i'm seeking answer to original question, which one is
more preferred in fax servers with 100 usres?
On Dec 29, 2007 12:10 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Sat, Dec 29, 2007 at 08:43:30AM -0700, Al lists wrote:
> > Any recommended how
Any recommended how to for 1.4 iaxmodem and hylafax+ ?
On Dec 29, 2007 6:49 AM, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Al lists wrote:
> > So HylaFax and IaxModem is more preferred than using rxfax/txfax ?
> > any reason?
>
> HylaFAX+ has built-in support for handli
So HylaFax and IaxModem is more preferred than using rxfax/txfax ?
any reason?
On Dec 28, 2007 6:40 PM, Lee Howard <[EMAIL PROTECTED]> wrote:
> Al lists wrote:
> > what method is preferred:
> > haylafax and Iaxmodem or spnadsp for faxing.
>
> I think that you mean t
what method is preferred:
haylafax and Iaxmodem or spnadsp for faxing.
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Agreed!
Polycom and Polycom and Polycom!!
On Aug 20, 2007 3:26 PM, Michael Graves <[EMAIL PROTECTED]> wrote:
> Sorry to top-post..but I haft agree here. Polycom is the KING of this
> sort of thing.
>
> Also, there really is a difference beteen a desk phone and a
> conference/borard room phone. Ha
this message is basically tells you asterisk is not running.
can you check and see if asterisk is running and present in memory?
something like
ps -ef | grep asterisk
On 10/20/07, Dominic Son <[EMAIL PROTECTED]> wrote:
>
> I was previous using Asterisk 1.2.9.1 and decided to get some real
> serv
I Just wanted to add something here,
Having separate VLAN does nothing in terms of QOS.
In fact having a computer feeding from phone make more sense because phone
will untag packets coming from PC.
and after that its all about your switch how to prioritize packets.
Unless there is a way in your swi
i'm using Polycom 601 in an office of 30 handsets.
I have not heard my customer complaining about phones being rebooted after
page.
On 10/9/07, Bill Andersen <[EMAIL PROTECTED]> wrote:
>
> > I could not tell you in asterisknow but I use this feature with Polycom
> > phones on all of my installs.
Not sure about 723 bu you can buy 729 from digium
just got to their website and its really easy to install, it comes with all
instructions you need.
On 10/7/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:
>
> Hi List;
>
> From where I can buy the G.729 and G.723 licenses, and
> how I can install it
check tz option in your voicemail.conf
On 10/5/07, Chuck Bunn <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I have a really oddball time problem. When I check the server time using
> 'date' it is correct. When I review the time in Freepbx (under time
> conditions) it is correct. When I look at the time s
frequency?
> >
> > Regards
> > Bilal
> >
> >
> > No, ignorepat is for FXS ports (FXS ports use FXO
> > signaling). Also,
> > ignorepat does not apply to SIP phones, because SIP
> > phones provide
> > their
> > own
Send me an email off the list, i have em somewhere in my HDD.
On 10/2/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
>
> Kenneth Padgett wrote:
> >> Dear Atacomm Customers,
> >> We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm
> >> and its parent company Ataractic Corporat
er a leading digit has been dialed on
> FXS ports. How does ignorepat help this guy?
>
> Al lists wrote:
> > ignorpat is your friend
> >
> > On 9/30/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> >> On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wr
ignorpat is your friend
On 9/30/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote:
> > Dear List;
> >
> > How can I place a call via Zap/g1 (group) but need to
> > determine the line (FXO port)
> > that will go via it?
>
> Simply don't u
yea thats what i did i put SIP 1.6 and its working like a champ, there
should be a way to get it working with 2.2, i'll wait for my next 601 and
play with it.
On 9/26/07, Doug <[EMAIL PROTECTED]> wrote:
>
> At 00:18 9/26/2007, Al lists wrote:
> >One more thing i noticed
One more thing i noticed today,
with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to use with
hints.
I'll spend more time on it later to see what is up with that.
On 9/25/07, Mike <[EMAIL PROTECTED]> wrote:
>
> I am having a similar issue with 4.0.0. Mine is that it doesn't get any
any firewall in between?
On 9/18/07, Richard <[EMAIL PROTECTED]> wrote:
>
> Sorry if this comes thru twice, I had the wrong account selected to send
> the
> first time...
>
>
> Callers to the number get ringing, I get stuff in my asterisk console, and
> it calls my softphone and ata, but answerin
of their
> TDM offerings), I would consider it. It's all IP in the core now
> anyways, no real reason to use TDM for the last mile.
>
> Maybe it has something to do with the number of simultaneous calls you
> can stuff down a data T1 using G729....
>
> Thanks,
> Steve
>
In VOIP, your quality of your voice is as good as your network.
if you want clear call quality, QOS is a must.
Well, when the call leaves your network and enters internet, QOS is not
enforced.
As a general rule choose the closest to your network.
for me its Teliax, i get to their proxy after 7 hops
i did have same issue with DISA in 1.4 and TDM400 FXO,
I switched back to Authenticate and waitexten.
On 9/14/07, Benjamin M. <[EMAIL PROTECTED]> wrote:
>
>
>
> Originally posted at http://forums.digium.co
Looks good!
i need to find a distributer to buy one.
On 9/13/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
>
> Anthony Francis wrote:
> > Aastra now makes a full SIP DECT system with cell style seamless hand
> > off from access point to access point.
> >
> > Caveat: This does not use standard wire
I'm using Linksys Wip300 and i'm not happy with it.
On 9/13/07, Dave Walker <[EMAIL PROTECTED]> wrote:
>
> On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote:
> > Hi folks:
> >
> > I know it's come up a few times before, but I need some more detail.
> >
> > I'm looking for a SIP DECT (cordles
Although you can find a router with QOS or dedicated bandwidth feature,
I would suggest a QOS enabled Switch.
Any IEEE802.1p enables switch,(these days less than $100 for 16 port) can do
the job.
you cant do alot when your traffic reaches internet, thats why most you can
do is up to your modem.
cos
I'm trying to get some more information on this myself as its a new product
from Cisco.
What i know, Cisco attendant console works with skinny,Cisco page and SLA
also works wiht skinny and not SIP.
So its either having these or SIP.
On 9/10/07, Drew Gibson <[EMAIL PROTECTED]> wrote:
>
> Jeremy M
I liked the queue game concept!
although it could be cruel!
On 9/11/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
>
> http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up
>
> Seems the Adtran relationship goes way back...
>
> Thanks,
> Steve Totaro
>
>
Maximum retries exceeded on transmission usually comes from NAT issues.
you can try this system without NAT and see if problem has resolved.
On 9/7/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
>
> Hi All,
>
>
>
> I'm working from home today (DSL -> Internet -> 2MB leased line -> A*K
> server behi
Also your Disk subsystem speed.
having disk RAM , makes sense in your case.
On 9/10/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
>
> Barton Fisher wrote:
> > Thanks, OK, a bit confused The cards are TE410P. I really don't
> > see how the set a codec for this, other than it might default to
>
Hi list,
I'm trying to get some ideas on this subject.
Normally astersik sends emails with voicemail attached trough local MTA.
As far as i know there is no way for asterisk to authenticate to an external
mailserver to relay these emails.
Well, these days every provider has some sort of spam blocki
Nice to know, luv to have this practical numbers.
On 8/28/07, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I thought I'd give a follow up to this discussion for the archives...
>
> Currently I'm trunking 30 channels of g.729 traffic (no transcoding going
> on, the call comes in and goe
Actually i'm using Polycom 501's behind nat and i have no issues.
what i usually do is putting static routeable IP for asterisk and using nat
and qualify in sip.conf.
no issues for me so far.
i'm a big fan of Polycom phones, quality of voice, working great with
asterisk and low failure rate.
On 8/
Stephen
> Bosch
> Sent: Monday, 27 August 2007 4:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Polycom firmware download
>
> Hi:
>
> Doug wrote:
> > At 13:29 8/25/2007, Al lists wrote:
> >> Thats just sad
What Digium is using is rpath, RHEL /Centos
On 8/25/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
>
> Matt Riddell wrote:
>
> > Steve Totaro wrote:
> >> I am bringing up several Fedora Core 7 boxen into production now.
> >>
> >> Besides a knee jerk reaction that "Fedora Sucks", can someone give a
Thats just sad,
I got SIP 2.2 from trixbox now, but still we need to have some sort of place
at least for ourselves to download this stuff.
Looking for boot loader now.
On 8/25/07, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
>
> On 8/25/07, Al lists <[EMAIL PROTECTED]> wrot
Hi,
I'm trying to use Polycom 330 and apparently it needs latest firmware (SIP
2.2.0).
I dont have access to polycom site to download and was wondering if any of
you guys have it.
Thank you!
___
--Bandwidth and Colocation Provided by http://www.api-digita
Is iaxtel still around?
I was not able to go to www.iaxtel.com .
did the address changed?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
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so you are not talking about vanilla asterisk, there are some other
applications involved.
Paging by nature is resource intensive, but still not sure what else is
going on in your system.
On 8/14/07, William McCloskey <[EMAIL PROTECTED]> wrote:
>
> The stability problems we have seem to be related
Cant help you with storm issue but second problem you have is coming from
bad FXO module.
Replacing that module should fix it.
On 8/8/07, Michael J. Liberatore <[EMAIL PROTECTED]> wrote:
>
> Hi, I am having some major problems with 2 digium cards in two seperate
> servers they are both TDM400P ca
I'm using Page application with Polycom 501 and 601 and have not seen these
issue,
i would check firmware on 601 and play with couple different firmware.
are you checking if the chanavail before sending the Page?
On 8/8/07, Bill Andersen <[EMAIL PROTECTED]> wrote:
>
> Asterisk 1.2.13 - Evolution
Clarify this, what you are trying to achieve?
To see if handsets are being used or not?
Or to see if any trunk is being used or not and share it?
These are 2 different concepts, first is BLF you can have your asterisk to
provide that information with hint priority, and the second one is SLA.
On 8
SLA is not BLF.
The only thing you need to configure to have BLF is adding hint priority to
your dial plan.
On 8/8/07, James Collier <[EMAIL PROTECTED]> wrote:
>
> Flash Operator Panel would do it.
>
> Also the Aastra 55i phones with the expansion module, which has 36 lines
> on
> it should work,
Nat?
On 8/6/07, Jason Walker <[EMAIL PROTECTED]> wrote:
>
> I am getting this error
> [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum
> retries exceeded on transmission [EMAIL PROTECTED] for seqno
> 102 (Critical Response)
> [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944
what you are reading on Cisco manual "DN" is a completely different concept
that what we are dealing in asterisk.
In CME you refer to each number as a DN, that concept does not exist on
Asterisk.
Although Asterisk support SCCP (Skinny) and H323, but its always easier and
better to use SIP or IAX.
i
easiest way of connecting multiple Asterisk boxes are trough IP network.
I know Digium cards supports HDLC encapsulation but i'm not sure about
framerelay.
On 8/4/07, Michael Munger <[EMAIL PROTECTED]> wrote:
>
> What modules do you want on it?
>
>
>
> Yours,
>
> Michael Munger, dCAP
>
> 404-438
Iax channel can be encrypted.
Not just the authentication, even rtp data, see:
http://www.voip-info.org/wiki/view/IAX+encryption
On 8/4/07, Michael Munger <[EMAIL PROTECTED]> wrote:
>
> IAX is not encrypted. What you're seeing in wireshark is likely the
> authentication method you've chosen. (RSA
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