Displaying 20 results from an estimated 10000 matches similar to: "Latest CVS (7/20/2004) stops answering SIP calls after 5 min"
2004 Aug 11
2
Asterisk & MyPhoneCompany.com (aka Talk(n))
They say on their website that they allow you to use your own device
provided you give them the MAC address. Has anyone tried using * with it?
Looks like they have quite a few rate centers and also phone support...
Their website is horrible though...
Just wondering, it'd be good to get user experiences from different
providers other than IconnectHere and BroadVoice...
-Chris
2004 Jul 09
1
IVR Menu and VoiceMail quality
I have really tried to do my best googling and wiki-reading before asking
this question. I couldn't find the answers there so I throw myself at the
mercy of the list...
I get excellent quality for SIP -> PSTN and PSTN -> SIP calls, however when
I or anyone else calls from PSTN -> * the voice menus are oftentimes very
choppy. Sometimes they are absolutely perfect and I cannot tell
2004 Jul 28
3
Workaround for BroadVoice and possibly others...
I have an idea, tell me if this wouldn't work... I know it's really ugly,
but it might help some people until we can get round robin DNS checks for
peers...
Since * does not do GetHostByName() again until you reload your config, and
BroadVoice and I'm sure other sip providers are using round robin DNS, why
not create 2 [<your server here>-out] contexts in sip.conf, and then in
2004 Aug 19
3
GrandStream BT101 Attended Transfers
I know this must have been asked before, but I was just wondering, the
manual says it can do attended transfers, has anyone gotten this to work
successfully? How did they do it?
Is it possible to do attended transfers with the 'T' dial option? If so,
how?
-Chris
Chris Shaw
IS Manager
Water Tech Industries
Phone: (888)-254-8412
Fax: (503)-261-9118
E-Mail: chriss@watertech.com
2004 Sep 07
4
Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled?
> -----Original Message-----
> From: Chris Shaw [mailto:chriss@watertech.com]
> Sent: September 7, 2004 4:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2
> w/ojitterbuffer enabled?
>
{clip}
>
> If you can reproduce it, this smells like a bug... IAX runs over TCP and
TCP
>
2007 Apr 18
4
[Bridge] MTU Question
I have a bridge that has gigabit interfaces. The machine in question has the
fun job of being a Bridge, Firewall and SMB server. Both of the Gigabit
interfaces are connected to workstations directly via Xover cable (well
MDI-X to be exact). My question is, if I enable jumbo frames on the gigabit
interfaces will that make any difference in overall transfer rate of the
bridge? I was thinking it
2004 Jun 23
3
Voicemail Password Changes Lost on Asterisk Restart
Ok I have googled and googled and combed through the wiki for an answer to
this and have come up empty. What I'm finding is that when a user changes
their
VM password, it is saved somewhere like maybe the CSV database or something
because when you log in, the new password works fine, but it's not saving to
voicemail.conf. So new passwords are lost when asterisk is restarted and
people
2004 Aug 11
2
StanaPhone and Asterisks
I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them.
Here is what I have in my sip.conf:
[stanaphone]
type=friend
secret=pAsSwOrD ; skewed for this message.
username=3475341914
host=sip.stanaphone.com
2004 Aug 13
0
Broadvoice User hung up on voicemail
don't quote me on this but I believe the earlier assumtion is correction. I
think you need to have RTP going bothways otherwise the call will
disconnect.
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Shaw
Sent: Friday, August 13, 2004 12:40 PM
To: asterisk-users@lists.digium.com
Subject: Re:
2004 Sep 17
3
how to get caller ID
i cannot see caller ID of the call originated from outside zap channel.
i hv configured both zapata.conf and extensions.conf.
i m right now in india
i think asterisk only supports Bellcore enable caller ID.
so is it the same bug of BT caller ID problem in UK?
or it is the bug of my asterisk configuration?
i hv enabled callerID from my TELCO.
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An HTML
2006 Apr 12
2
* 1.2.4 & 1.2.6: "Ringing" anamoly
I was alerted the other day by of all people, my mom, that she wasn't
hearing a "ring" when she dialed my number. Puzzled, I tried calling myself.
The call connects, but there's dead silence until voicemail picks up.
Calling internally, extensions worked perfectly. So, I figured, "another
damned Broadvoice issue."
For kicks, I upgraded to 1.2.6 today, and the end
2004 Aug 31
4
T100P No D-channels
Hi
Last week I installed Asterisk (release1) with digium t100p single span T1
(wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is
currently the only user of the t1 line. All worked well for about a half a
day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I
consistently get multitudes of blue alarms on all b-channels followed by a
loss of d-channel:
Aug 31
2004 Jul 19
1
MAC OS X Panther :?
Just wondering if anyone has tried MAC OS X and panther.
I will like to do SIP to H323, not sure if this will be possible on the MAC
because of the Libraries PWlib and OPenh32 for Linux..
Just curious..
Anyway, anyone has an easy guide (step by step) to setup oh323 with
asterisk. I saw a guide but i am not very savy on linux.
thanks,
Francisco
----- Original Message -----
From:
2004 Jul 22
4
VSP? Looking for advice.
Has anyone tried using BroadVoice for VSP? I have Asterisk configured
for a home office & I've been trying to decide which VoIP provider to go
with for a little while now. I had heard you could get sub $.01 calls
but I have not found that to be true yet (not saying it's not possible,
I just haven't found it!).
Also I'm not sure if BV will support multiple lines. Any
2012 Jun 04
2
paquete SPEI función thornthwaite
Hola eRReros.
Os lo explico con un ejemplo:
# Cargamos los paquetes y el ejemplo
install.packages("SPEI")
library("SPEI")
data(wichita)
# los primeros 12 datos
head(wichita,12)
# mi subset de los primeros 12 datos
meu<-wichita[c(1:12),]
meu
# como veis los valores de TMED son iguales en ambos dataframes.
# ahora viene el problema
# calculamos
2004 Jul 22
6
D-Link DPH-80S vs *
List,
The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk.
Seshu Kanuri
"G
2009 Jun 22
5
Convert "ragged" list to matrix
Hi,
I have a list made up of character strings with each item a different
length (each item is between 1and 6 strings long). Is there a way to
convert a "ragged" list to a matrix such that each item is its own row?
Here is a simple example:
a=list();
a[[1]] = c("a", "b", "c");
a[[2]] = c("d", "e");
a[[3]] = c("f",
2002 Nov 05
0
FW: rsync-cvs mail list archiving broken??
Thanks!
If I can be of any help, please let me know. While I don't have a lot of
free time, either, I enjoy working on open-source software and have several
years of experience porting it to our system. I'd be happy to try to lend a
hand from time to time, applying well-formed patches or looking at bugs.
Maybe my slack and busy periods would happen at different times from the
other
2004 Sep 10
1
Can I STOP decoding at an exact sample?
Hi guys,
Thanks Matt for your prompt response about the C++ problem, its working
great now!
Again for my "virtual cdplayer", I am wondering if its possible to stop
decoding within a large file, at the end of an exact sample?
ie. I wish to play a song in the middle of a flac'd cd. Using the file
decoder, I can seek to the start of the track(index 01 of the cds TOC),
but I then want
2005 Feb 02
2
How to download CVS with attended transfers
Hi
I know that attended transfers are only available in the CVS Head.
I downloaded the asterisk-update.sh script from voip-info.com and ran it
with these parameters
./asterisk-update.sh update dev
It looked as tho CVS HEAD was downloading and compiling, although it
couldn't download the addons.
However, now it's up and running, only blind transfers work with "#", and I