Chances are it's waiting to get the caller ID info (sent between the
first and the second ring)
On Wed, 2004-03-03 at 12:01, WipeOut wrote:
> Brian Mulligan wrote:
>
> >Sorry if this is a daft question but when a PSTN call comes in on my
> >X100P the console shows the following;
> >
> >NOTICE[121
t and T are for "#" transfers. Other types of transfer are done in
other ways. Zap FLASH transfers are set in /etc/asterisk/zapata.conf.
I don't know how you enable/disable SIP or other types of transfers.
On Thu, 2004-03-04 at 06:51, Zen Kato wrote:
> Hi,
>
> Thank you for the information. Th
Does anyone know of a way to reset the AbsoluteTimeout so it restarts
the timeout? i.e. regardless of how long the call is currently going
on, make sure it does not last more than 30 more seconds.
--Eric
--
Eric Wieling <[EMAIL PROTECTED]>
BTEL Cons
Maybe you are using inband DTMF with a compressed codec. DTMF on a call
with any codec other than ulaw or alaw MUST use OOB DTMF like RFC2833 or
INFO.
On Fri, 2004-03-05 at 20:39, Barton Hodges wrote:
> I'm having a problem with transferring a call that comes in a Zap
> channel and is connected w
What is your ACTUAL Dial line?
On Fri, 2004-03-05 at 21:19, Barton Hodges wrote:
> I'm using SIP INFO and ulaw. It seems that the same thing happens
> from SIP to SIP as well, regardless of what the DTMF setting is. The
> actual problem is that the calling user can transfer, but the called
> use
em in an *
> environment?
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First of all Asterisk does not support ${VARIABLES} as part of the
extension number. i.e. exten => ${BLAH},1,NoOp is not valid, but exten
=> 1234,1,NoOp(${BLAH}) is valid.
Also Asterisk NEEDS the sending fax machine to send standard fax machine
tones (CNG, I think) for it to be detected. When yo
On Wed, 2004-03-10 at 12:25, Michiel Betel wrote:
> Eric Wieling wrote:
>
> >First of all Asterisk does not support ${VARIABLES} as part of the
> >extension number. i.e. exten => ${BLAH},1,NoOp is not valid, but exten
> >=> 1234,1,NoOp(${BLAH}) is valid.
> >
> >Can you test this with an extension that goes into VoiceMailMain(). My
> >7960 and 7960G phones both get the first couple letters of "Commedian
> >Mail" cut off (usually "...median Mail").
For my first two or three months of using Asterisk I had this problem
(with a Cisco 1750 and Cisco FXO an
Bruno Haas wrote:
Wow, 1000 IRQs a second. I'm quite amazed. Does anybody know which
applications would require such a low latency ? It does seem to me that
this way of doing things is rather dangerous and prone to problems.
Anybody can comment ?
Processing voice from a card without a built in D
If you want reliable progress (busy, answer, etc) you must use a digital line
like a T-1 and/or PRI. It's as simple as that. There is no way for the X100P
to know when the call has been answered since analog lines do not signal that.
callprogress tries to fake it by listening to the audio and gu
One would assume that ZapRAS only works on Zap channels. ZapRAS only
works with ISDN DATA calls, not modem calls.
On Mon, 2004-03-15 at 15:13, Bisker, Scott (7805) wrote:
> Make that could not turn up in google.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECT
On Mon, 2004-03-15 at 18:25, Chris Lee wrote:
> First, is the lack of UK CLI on the x100P hardware or software related?
Check the extensive discussions in the mailing list archives.
> Secondly, My US Robotics Voice modem does get UK CLI, so could I get UK
> CLI and the same functionality as the
When Asterisk receives a 302 "Moved Temporarily" response Asterisk
correctly forwards the call to the new extension, but the ${CALLERID}
info is lost. Cisco 7905 sends this message when a call comes in and the
user presses "ToVM" button on their phone.
Also the ${DNID} and ${RDNIS} variables ar
One was posted to this mailing list TODAY.
On Tue, 2004-03-16 at 12:32, DanJr wrote:
> Anyone know of any software Fax program that will work with asterisk
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BTEL Consulting
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"show application dial"
On Tue, 2004-03-16 at 19:49, Jim Sneeringer wrote:
> Thanks so much. This sounds encouraging.
>
> I can't find a description of the "d" option. What does it do? Do I just put
> a "d" in the extension? Right now I have this:
>
> exten => fax,1,Dial(Zap/8)
>
> and I'm e
This really does seem too good to be true.
On Tue, 2004-03-16 at 19:36, Matthew Marlowe wrote:
> The FT201 is currently being manufactured and will be available shortly!
> The retail price will be $129.95 USD...
>
> http://www.virbiage.com/products/lanphones.php
___
> The FT201 is currently being manufactured and will be available shortly!
> The retail price will be $129.95 USD...
>
> http://www.virbiage.com/products/lanphones.php
The web page does not say:
1) how many call appearances does the phone has
2) does firmware costs extra
3) does it come with a p
For the archives
BellSouth Tariffs and Price Lists:
http://cpr.bellsouth.com/
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On Thu, 2004-03-18 at 07:47, Carey Jung wrote:
> Anybody have a list of area codes and prefixes for which Nufone can provide
> DIDs? I can't find any such list on their site.
Michigan only, but I believe they have decent coverage within Michigan.
I seem to recall they were planning on Chicago DI
Check the extensive thread regarding this EXACT ISSUE in the mailing
list archives.
On Thu, 2004-03-18 at 04:36, tim mickelson wrote:
> Hi.
>
> I'm not being able to make my Voicetronix Openswitch 12 work with
> Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is
> r
If the phones are behind NAT you can try lowering the registration
interval to something LESS than 4 hours, like 60 seconds or 120
seconds. Usually this problem is because the NAT firewall sees no
activity on the registration port and so deletes the NAT translation for
that port.
On Fri, 2004-03-
Useful Asterisk Docs (BOOKMARK THEM!):
http://www.digium.com/index.php?menu=documentation (look at the
"Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and
http://www.fnords.org/~eric/asterisk/ (my site) and
http://asteriskdocs.org/
___
A
ou
daled the number directly. This way AGI scripts, .call files and stuff
like that don't need to know what interfaces you have or be told what
interface to route calls matching specific pattern should go to. You
just Dial(Local/blah) and your dial plan will route the call.
--
Er
s" extension.
--Eric
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t; otherwise it will stop playing and return either 1 2 *
That's not running the Playback application.
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My Cisco 7910 works fine with chan_skinny.
I'm now trying to use the 7910 with chan_sccp. The phone hangs with a
message "Requesting Server List".
Has anyone seen this problem. Happens with both chan_sccp CVS and with
0.02.
--Eric
--
Eric Wieling * BTEL Consulting
Oh, it also seems to crash my Asterisk. (0.7.2).
On Sun, 2004-03-21 at 16:27, Eric Wieling wrote:
> My Cisco 7910 works fine with chan_skinny.
>
> I'm now trying to use the 7910 with chan_sccp. The phone hangs with a
> message "Requesting Server List".
>
The 7905G (but not the non-G) supports SIP. It does NOT support XML.
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To
905 only
> > supports the Cisco proprietary firmware (not sip) while the 7960
> > supports either Cisco or sip. That's probably why you're seeing v1
> > verses v6.3 or whatever.
>
> The 7905 and the 7905G both run the 1.01 SIP firmware.
--
On Mon, 2004-03-22 at 14:35, John Brown (CV) wrote:
> I really don't want to burn a DID per WATTS line
> so that I can route on the DID number.
Look at asterisk/doc/README.variables. Assuming you are using ZAP
interfaces look at ${DNID}.
--
Eric Wieling * BTEL Consultin
On Mon, 2004-03-22 at 15:22, Bill Hamlin wrote:
> I didn't find anything like "ldassume" using google. Can you tell me more
> about that?
It's in the RedHat 9 RELEASE NOTES.
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story,
t a T-1 for various reasons.
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[EMA
On Mon, 2004-03-22 at 20:42, Simon Brown wrote:
> I have my Asterisk server behind a Cisco firewall. I am trying to set up IAX
> but I cannot work out which ports I need to open up on my firewall. I have
> opened 4569, 5036, and 5060 but IAX calls will not proceed unless I turn off
> all access l
=> 3300,8,Dial(Local/[EMAIL PROTECTED]&Zap/25,,r)
exten => 3300,9,Hangup
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
_
th of these patches are in CVS versions after March 18 2004.
http://www.fnords.org/~eric/asterisk/
--Eric
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gam
On Tue, 2004-03-23 at 09:49, Eric Wieling wrote:
> I'm having a similar problem with 0.7.2 but ONLY if I dial multiple
> destinations at the same time. Here is a copy of my extension section
> that does NOT provide ringback no matter what I do. In this example the
> caller he
isco IP Phone 7960G Global
> 7960 IP Phone With One CallManager Express Station
> 7960 IP Phone with one Station User License
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the c
s and SIP software licenses the
same cost?
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"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
Asterisk-Users
ble passed to the hangup.agi script.
> I have tried
> $var = $ARGV[0];
> $var = $ARGV[1];
> but still can not get the passing variable value.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
hem.
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http://lis
xt step is to get an Ethereal log from both ends and investigate what
> > is going on with the SIP and RTP packets.
>
> Yes, that's the next thing I have to try. Hopefully this evening.
> Interested to see you are just up the road: I'm in Winchester.
reinvite= is a myth. It d
vmbe+true,1,Dial(Zap/1,30)
exten => 2111+vmbe+true,2,VoiceMail(u2196)
exten => 2111+vmbe+true,102,VoiceMail(b2196)
# Mailbox 2111 in voicemail context [default] does NOT exist
# Ring forever
exten => 2111+vmbe+false,1,Dial(Zap/1)
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Eric Wieling * BTEL Consulting * 504-
lbox of the corresponding
> number?
>
> Thanks
>
> Deepak
> ___
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GhostScript can output:
faxg3 faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4
tifflzw tiffpack
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Michael Welter wrote:
I have a few 7940G phones on a LAN hub, and I'm looking at the 3COM
3CNJPSE power injector. Can I put one of these behind my LAN hub and
power all the phones, or do I need one for each phone?
From the spec, it looks like PoE tries to discover whether a device is
powered
Jeb Campbell wrote:
Anyway, the only stuff off list was trying to debug the connection.
1. With a crossover there is no sync (YELLOW and RED alarms)
2. With standard cable I get a pri error that they think they are the
NET, but we are the NET.
(This is asterisk 1.0 stable and the directions from
anyone offer their experience ?
>
>
>
> Cheers
>
> Peter
>
>
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
__
available" (not the 102 busy as expected).
>
> Do I need to do something different to hit the 102 busy priority?
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that
ou updated your Asterisk. I believe that the CVS stable as of a few
days ago fixes the problem. If not, put a "r" option at the end of the
dial like to work around the problem. e.g. Dial(Zap/1,20,r) If you
don't use a timeout then use something like Dial(Zap/1,,r)
--
Eric W
On Wed, 2004-03-31 at 20:57, Gene Kochanowsky wrote:
> Thanks Eric. Is this the CVS branch you are referring to? -
>
> # cvs checkout -r v1-0_stable asterisk
Yes.
--
Useful Asterisk Docs:
http://www.digium.com/index.php?menu=documentation (look at the
"Unofficial Links") and http://www.voip
> Asterisk-Users mailing list
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are on my Cisco routers to get them to grok rfc2833.)
>
> -T.i.A., Jim
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would monitor multiple
Asterisk servers from the same screen. Not a major issue, but would be
nice.
On Fri, 2004-04-02 at 08:25, Nicolas Gudino wrote:
> Hi Eric,
>
> - Original Message -
> From: "Eric Wieling" <[EMAIL PROTECTED]>
> Sent: Friday, April
applications' command to list all available
> apps. If you hit tab, it acts just like BASH's auto
> complete. Wonderful feature!
>
> Mitch Sharp
> Innovative Solutions
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"
On Tue, 2004-04-06 at 12:29, Brian Rathman wrote:
> Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco
> AS5300 with * in the media stream. Unfortunately, the only way I can get the
> calls to connect is with t or T at the end of the Dial() statement and then
> that picks off th
ny major issues, critical
> > security updates, etc, that his system might need to be updated for.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gamblin
There are several ways to know what changes in Asterisk's CVS.
This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly
up to date CVS changelog summary information.
You can also sign up for the Asterisk-CVS mailing list at
http://lists.digium.com/mailman/listinfo/asterisk-cvs
Arc
terisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling
ude => default
> ;
> ;[submenu]
> ;exten => s,1,Ringing ; Make them comfortable with 2
> seconds of ringback
> ;exten => s,2,Wait,2
> ;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales department.
> Press 1 f
On Thu, 2004-04-08 at 10:30, Jeremy Koski wrote:
> Does anybody know if the 3com NBX 100 phones will work with
> Asterisk? The phones require a boot image to be sent either through
> layer2 or layer3 before they will function properly after being powered
> on each time.
You didn't see the informat
P
> nothing:-(
Cisco does not document the SCCP protocol. There are no public docs
available for this protocol. The Asterisk SCCP/Skinny support was done
by using a packet sniffer.
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recentl
ound to search the archives was month by month.
>
> There's an option to download all of the mailing list archives, but its
> 129MB...
>
> On Thu, 8 Apr 2004, Eric Wieling wrote:
>
> > On Thu, 2004-04-08 at 10:30, Jeremy Koski wrote:
> > > Does anybody know
No that will not work. You would want three exten => lines, one for
each area code.
Michael Munger wrote:
> Will this work to match any number from the 770,404, or 678 area codes?
>
>
>
> _[404|770|678]NXX
>
>
>
> If this won't work, is there a pattern that will do this?
>
>
>
>
That will match the following as well
770
700
740
400
470
670
600
604
608
etc.
You example says:
The first digit can be 7 or 4 or 6. The 2nd digit can be 7 or 0 or 4.
The 3rd digit can be 0 or 4 or 8.
Mike Trest - Personal wrote:
> [746][704][048]
>
> [At 01:21 PM 2/21/2008, you wrote:
>
This problem would happen if you did not have /etc/asterisk/indications.conf
Fons van der Beek wrote:
> I tried that, its gives me the same problem.
>
> Kevin P. Fleming schreef:
>> Fons van der Beek wrote:
>>
>>> Because i want a ringing signal while people are in a waiting queue i've
>>> cr
> on the other hand, internaly: it's ok
>
> exten => 205,1,queue(receptie|r)
> exten => 205,2,busy
>
> 205 gives ringing
>
>
> Eric Wieling schreef:
>> This problem would happen if you did not have
>> /etc/asterisk/indications.conf
>>
&g
Replying to my own post. Asterisk uses indications.conf when it has to
provide tones AFTER the line is answered. You might get a message on
the console like "Unable to handle indication 15" or something like that.
Eric Wieling wrote:
> Don't answer the line. Also try using
NOT answering did the trick!
> Tnx a lot! now it works like it should work!
>
>
>
> Eric Wieling schreef:
>> Replying to my own post. Asterisk uses indications.conf when it has
>> to provide tones AFTER the line is answered. You might get a message
>> on the conso
I must have started reading this thread after you reported that you
actually had an AUDIO problem rather than a RINGBACK problem.
The issue you experienced is a common one. Someday I hope Digium fixes
that bug/design flaw.
Fons van der Beek wrote:
> Jared YES
> That seems to be the problem
My guess is a mismatch between Asterisk, Zaptel, and libPRI. Make sure
you are running the latest of each.
Tzafrir Cohen wrote:
> On Mon, Feb 25, 2008 at 03:27:07PM -0500, arkda wrote:
>> Hi,
>>
>> I've been playing with a TE120P on Asterisk 1.4.18 with zaptel 1.4.9 and
>> I've ran into an issue
Could this be ECFO?
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as "screeching",
"feedback", "static", or other useless terms. If users report "static"
on a system where there cannot be static (all digital, PRI, SI
phone1.cfg:
call.callsPerLineKey="1"
Raúl Gómez C. wrote:
> Michael,
>
> I haven't used nor configured a Polycom phone, but you should check in
> /etc/asterisk/sip.conf the "call-limit" param of the phone's config.
>
> On Thu, Feb 28, 2008 at 12:31 PM, Michael Munger <
> [EMAIL PROTECTED]> w
Andres Jimenez wrote:
> On Sat, Mar 1, 2008 at 6:00 PM, Lacy Moore <[EMAIL PROTECTED]> wrote:
>> Not sure on #1, but #2 is not possible on SIP.
>
> Busy Lamp Field IS available on SIP.
>
> If it is not, I cannot imagine how my GXP-2000 does it.
I think the poster was specifically referring to
Use most anything except FXO signaled ports. PRI, BRI, SIP, IAX2, etc.
aymen warfalli wrote:
> How to calculate the PSTN call durations through zaptel ,where in the CDR it
> gives the time durations started when the zaptel answerd + PSTN dialing
> time + ringing time even thoug the destinatio
I would strongly recommend ESF/B8ZS. If you have a RED alarm that means
the device does not see a line connected to it -- check cabling.
Mark Best wrote:
> Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system
> to work with *?
>
> I'm not getting inbound or outbound calls to w
Yes, they are listed in the Admin Manual for the Polycoms
Peter Hessler wrote:
> Is there a way to crank the volume on Polycom speaker phones? The 430
> and 4000 that I have are quieter than expected. The volume on the
> device is turned all the way up, but are there firmware options that
> c
gt; automatically adapt?
> TELCO---T1---MITEL
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Eric
> Wieling
> Sent: Tuesday, March 04, 2008 12:58 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subjec
They get UTC/GMT from the NTP server. It is up to the firmware on the
phone to convert that date/time into the local time. No, it is not up
to Asterisk, it is up to the phone firmware.
Chris Carey wrote:
> They get the time from their NTP server
>
> On Mon, Mar 10, 2008 at 11:59 AM, Don Smith
Lower the rxgain and txgain on your Zap channels.
bilal ghayyad wrote:
> Hi Brent;
>
> I have been suffering from this problem since about 2
> monthes and until now still did not resolved 100%.
>
> First of all, I need to tell u that mostly u have a
> problem that the first digit is duplicated,
lines. My rxgain
> is already -2.8 and if I drop my txgain below +4 callers complain that
> they can't hear the users on the Sip phones inside the offices.
>
> Thanks,
> Brent
>
> Eric Wieling wrote:
>> Lower the rxgain and txgain on your Zap channels.
>>
&g
Mian M Asif wrote:
> Hi All,
> i want to configure voice mail on Asterisk 1.4 for multiple users. let
> me explain you the scenario.
>
> i have 10 users with the name of
> 1000,2000,3000,4000,5000,6000,...and these user can call to each
> other. Now i want to configure separate voice mail bo
The only messages I have EVER seen Digium remove from the mailing list
archives are discussions about this unlicensed codec.
Martin wrote:
> Download an appropriate binary from
> [url removed]
> and just drop into /usr/lib/asterisk/modules/
> add allow=g723 to your sip.conf as necessary and resta
exten => i is for IVRs.
You would need a wildcard catchall extension like the one below.
Unfortunately you are doing the classic newbie mistake of thinking you
can have a simple dialplan by making the SIP user/account ID be the same
as the extension. Eventually you will realize this is a bad i
As this is the mailing list, not the archive, that's not surprising.
Jaswinder Singh wrote:
> That's strange , i am able to see the *url* in Martin's reply .
>
> On Sun, Mar 23, 2008 at 6:14 PM, Eric Wieling <[EMAIL PROTECTED]> wrote:
>> The only messages I h
Without knowing the line type, card model, etc, I doubt anyone can help
you. FXO signaled ports do not support receiving the dialed number.
mark morreny wrote:
> Hi all,
>
> I am using Digium PCI board to receive PSTN call through regular phone
> line. It is no problem for me to receive calls,
Looks to me like you changed the ethernet controller. The G729 copy
protection is based on the MAC of the interfaces in the system.
Guilherme Loch Waltrick Góes wrote:
> I'm trying to use the Digium suplied G.729 Codec, I have ran the register
> utility, and got my licenses written to /var/lib/a
Watkins, Bradley wrote:
> Being able to pass variables around between systems is by *definition*
> channel-specific, since the channel driver is the module responsible for
> speaking a given protocol. Besdies, SIP already has (and has had for a
> long time) a method for doing this (SIP headers).
Inband only works with the ulaw and alaw codecs.
David Nedved wrote:
> Hi All,
>
> Anyone have any idea what could cause incoming calls on a SIP channel
> to no longer be able to use DTMF? DTMF on incoming calls on zaptel and
> on local SIP softphones and ATAs all work fine. Nothing gets
> regi
responses inline
bilal ghayyad wrote:
> So I would like to ask, did any one try it later and
> wether it is good or not? I am asking this because I
> need to use it as it is NAT Transparent (as I read
> also, and I did not try it to see how much it is
> transparent).
Thousands and thousands and t
You will never get latency on a network low enough for echo to be
perceived as sidetone (like on analog). If you want to get rid of echo
you must cancel echo.
Mojo with Horan & Company, LLC wrote:
> Sean Dennis wrote:
>> bilal ghayyad wrote:
>>
>>> Hi All;
>>>
>>> I have been chocked just wh
This is a reasonably common problem. ztdummy uses the Linux kernel Real
Time Clock (RTC) and something is wrong with it. The solution is to
recompile your kernel, you should search the mailing list archives.
Prepend "site:lists.digium.com" to your Google search to limit your
search to the m
Rizwan Hisham wrote:
> Hi,
> Does anyone know the purpose of "/n" attached at the end of the dial
> command below
>
> Dial(Local/[EMAIL PROTECTED]/n )<
Yes, and you will too when you read localchannel.txt in your Asterisk
source code docs directory.
--
Consulting for Asterisk, Polycom, Sa
A "w" in the D() string will wait .5 second. Example:
Dial(Zap/g1/5551212,,D(ww668))
If you are dialing out of an FXO or FXS signaled port, you can add "w"
to the dial string to wait .5 second. Example: Dial(Zap/g1/ww5551212)
Pete Kay wrote:
> Is there anyway to have Asterisk to wait for 1 se
Yes, some kernels don't work with ztdummy. This is discussed over and
over and over again on this mailing list. Check the archives.
Tzafrir Cohen wrote:
> On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote:
>>> On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
>>>
/ I hav
"core show application retrydial"
You need to do a "core show applications" and look at what apps are
included with Asterisk.
Tony Mountifield wrote:
> Has anyone here implemented "Ring back when free" in Asterisk?
>
> The way it works in the UK is as follows:
>
> 1. A calls B. B is engaged (b
Any time you have this kind of hard lockup with a Digium card you should
run, not walk to the nearest phone and call them.
broadband Voice wrote:
> I installed the Digium T1 card on Dell Poweredge 2950 and the system crashed
> several times, we got a Kernel Panic and first though it was the OS so
This is a very common issue with Asterisk. There is no good fix, but if
you make sure ALL IP addresses of the server are listed in /etc/hosts on
the server it may help.
Marius Muja wrote:
> It is using a local DNS server.
>
> On Fri, Apr 11, 2008 at 10:43 AM, Steven Kurylo <[EMAIL PROTECTED]>
The PSTN only allows ulaw or alaw (depending on your location). You
CANNOT send calls in any other codec over a PSTN line. Generally, if
you want to use G729 then you must buy a G729 license (with a few
exceptions).
Jeremy Mann wrote:
> But I want my polycom to attempt g729 on SIPPEER-SIPPEER
he
> phone supports the codec it needs natively.
>
> Is there any dialplan logic that can coerce the transaction to be ulaw only?
> Setting something in the SIP header perhaps?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieli
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