Do you have any issues with not being able to hear the called party after +3
minutes? That is pretty consistent thus far.
Don't get me wrong, I am liking the phone so far. Small interface, easy to
configure. Uses an XML derived config file - nice for deployment to multiple
computers. And the porti
Dave Grey wrote:
Speaking of glaringly unworkable, like a numb-skull I edited and
tested the thing with default my green-on-black color scheme. I
happened to open something up in a raw black-on-white xterm and
realized that I had created a nightmare. I have made the appropriate
change
I'm running it on sp2 myself, never had a crash with it so far.
Jason Walker wrote:
Are you running on XP SP2just curious? How about the version of *?
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf O
Are you running on XP SP2just curious? How about the
version of *?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
VileSent: Saturday, October 22, 2005 10:03 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] iax softphone
Nop
Done -
Joachim, I cc'd you on the email so you could see what I sent.
Let me know if more info is needed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoa
Sent: Saturday, October 22, 2005 10:03 PM
To: Asterisk Users Mailing List - Non-Commercial Dis
If that's dishnetwork and they keep charging you their $5 programming
access fee or whatever they call it, just plug it in and confirm that
you get a dial-tone. Then call tech-support and have them adjust
billing -- all they check is that the receiver gets a dial-tone and they
take your word for i
> -Original Message-
> From: Joe Greco [mailto:[EMAIL PROTECTED]
>
> > juju to have all your IPs in one block.
>
> Actually, they didn't have all their DNS servers in one block.
Touche... I didn't even double-check, just assumed.
> while, we ran with sequentially numbered servers that
Nope, I do not have that issue.On 10/23/05, Jason Walker <[EMAIL PROTECTED]> wrote:
Tom - do you end up with that phone shutting down with
an error on Windows XP? I downloaded the latest. After about 3 minutes on a
call, the other end can no longer hear me and then the phone just
dies.
F
Jason, i didn't hear about that problem before (several thousand people
are using that version), could you please send a copy of your config
files + the exact version and language localisation of windows xp to
[EMAIL PROTECTED]
Does it happen with one specific version of asterisk ?
Whatever the
sure...On 10/23/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]
> wrote:Widyachacra Rajapaksha wrote:> dear friends,>> what is de best CDMA card or module made for Asterisk?
>> --> --->> the path to freedom.> --- 2.6.13-gentoo-r4>>>>_
I have an Asterisk server that peers with a VoiP provider via IAX2. I have
10 local SIP users.
I record the CDR data into a MySQL database, and use that to bill the 10
local SIP users.
The problem I have is, one of my local users (User 5, for example) has there
handset forwarded to a mobile
On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed:
>
>I just setup telasip and I'm having the same issue. I captured some RTP
>packets and realized that when I get duplicate numbers it is because an
>RTP packet has arrived out of order. In all my test cases it was just
>one packet coming 1
Tom - do you end up with that phone shutting down with
an error on Windows XP? I downloaded the latest. After about 3 minutes on a
call, the other end can no longer hear me and then the phone just
dies.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom
VileSent: Saturday,
Idefisk for me. I love how it does not clutter the screen and it works.On 10/22/05, Matt Florell <[EMAIL PROTECTED]
> wrote:We use the Firefly ThirdParty softphone on our windows laptops. It's
free, easy to configure and will do IAX2 and SIP:http://www.virbiage.com/firefly/download/firefly-thirdpa
Try changing at least on the sipura the RTP Packet Size: to 0.020 or 0.010
it should be under the admin login and then the SIP tab.On 10/22/05, Chris Mason <[EMAIL PROTECTED]
> wrote:I need my satellite receivers to call home to avoid problems with the
service. I have hooked them up through an IAXY
Sorry for the blank response -
before...
From your output below, what looks weird are the hex values
for the codecs:
[snip]
requested/capability
0x200/0xfe00 incompatible with our capability 0xf900.
From one of my servers, when I do a 'show codecs' on the
console, I get
sfsip01*CLI
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
clint_in_sydneySent: Saturday, October 22, 2005 7:15
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] Unable to negotiate
codec???
I use IAX and have a license for G729 at my end and
O
On Sat, 2005-10-22 at 21:01 -0600, Rich Adamson wrote:
> > Earlier today I lost registration with FWD, now Teliax registration is
> > down as well.
> > I can ping both networks but can not register.
> >
> > Can anybody check on their end?
>
> Of the two servers:
> host=voip-co1.teliax.com
I use IAX and have a license for G729 at my end and
OZTell, my provider, use G729 as their main codec.
My box rejects connections from my provider due to
incompatible codecs and vice versa.
I'm waiting for them to get back to me on
this.
Clint.
- Original Message -
Fro
I need my satellite receivers to call home to avoid problems with the
service. I have hooked them up through an IAXY and tried a SPA2002 set
to G711 and made sure the transport is 711 all the way. However, it does
not work at all, the receivers cannot make the connection work.
Has anyone made t
Good lord, it's part asterisk part goiax.com
If you have an issue with it ignore the thread. At first I thought I had asterisk config'd wrong.
Now I know better than to waste my time with a list that has people like you on it.On 10/21/05, Robert Webb <
[EMAIL PROTECTED]> wrote:On Fri, 21 Oct 2005
> Earlier today I lost registration with FWD, now Teliax registration is
> down as well.
> I can ping both networks but can not register.
>
> Can anybody check on their end?
Of the two servers:
host=voip-co1.teliax.com
host=voip-co2.teliax.com
the co2 was not pro
Earlier today I lost registration with FWD, now Teliax registration is
down as well.
I can ping both networks but can not register.
Can anybody check on their end?
--
#Joseph
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users m
What codec are you using on the client and the server? From
my understanding, you have to have a license for both ends of the G.729 call.
Are you passing this through one server to another and the call is being
rejected at the server level?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi-
I am attempting to setup Asterisk for the first time, and I think I am about
99% there. I am using vonage softphone, and want to use asterisk to
redirect incoming calls to my cell phone primarily, and maybe other remote
lines.
Right now, I am able to register with vonage, and trap inco
Hi All,I get the following when trying to dial in to my asterisk
box.Oct 22 13:58:11 WARNING[3599]: Don't know any of 0xf800
formatsOct 22 13:58:11 ERROR[3599]: No best format in 0xf800???Oct 22
13:58:11 NOTICE[3599]: Rejected connect attempt from 203.98.83.19,
requested/capability 0x200/0xf
We use the Firefly ThirdParty softphone on our windows laptops. It's
free, easy to configure and will do IAX2 and SIP:
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
MATT---
On 10/22/05, Zoa <[EMAIL PROTECTED]> wrote:
>
> Idefisk is currently only for windows, but a native linu
On Sat, 22 Oct 2005, Jean-Michel Hiver wrote:
> I have a potential client who has legacy alarm systems which use modems
> to transmit encoded data to a remote location through the PSTN. They
> wish to replace the 'PSTN' bit with an IP link.
>
> I am aware that it would be best if the data was tra
On Oct 22, 2005, at 12:18 AM, Dave Grey wrote:
On Oct 21, 2005, at 5:50 PM, JP Carballo wrote:
Dave Grey wrote:
I hacked together an emacs general/minor mode for basic font-
locking (syntax shading) support. Feel free to grab it here:
http://homepage.mac.com/lydanynom/asterisk-mode.el.zip
Hey jean,
I have had to deal with the same situation many times what we have come to
the conclusion is you can actully get them to work only under the 4-2
signaling that the alarm companys use. Just about all alarms now are set to
use contact id which we have found out that we send the data ok bu
I have asterisk running with sipura 3000 connect to PSTN and
sipura 2000 connected to phones.
On inbound calls I am getting what sounds like DTMF tone when
someone is talking on the PSTN side of the phone. It sound like
someone is hitting key on the phone while talking.
Is there any way to stop
Idefisk is currently only for windows, but a native linux version is
nearly ready and will be released soon,
others i can also recommend
:
- iaxphone by ipsando
- firefly by virbiage.
Time Bandit wrote:
can someone tell me about a good iax softphone ??
Shameless plug : http://www.marccharbo
> can someone tell me about a good iax softphone ??
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php
works only on windows
for one that works on Windows and Linux :
http://iaxclient.sourceforge.net/iaxcomm/index.html
there is also DIAX : http://www.laser.com/dante/diax/d
> Thanks for the wealth of information. I knew they were off-air
> DNS-wise, and this happened before a couple of times. It's just bad
> juju to have all your IPs in one block.
Actually, they didn't have all their DNS servers in one block.
It's also a fallacy that having DNS servers in a single
On Fri, 2005-10-21 at 09:26 -0700, trixter aka Bret McDanel wrote:
> On Fri, 2005-10-21 at 09:27 -0700, [EMAIL PROTECTED] wrote:
> > I received some postings back, as I was trying to do the same thing.
> >
> > it' is a problem with Kernel 2.6... 2.4 works fine .. this is the summary
> > I got from
can someone tell me about a good iax softphone ??
thanks
Daniel___
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To UNSUBSCRIB
strace?
valgrind?
There aren't any profiling tools or the sort within the Asterisk suite that will deliver the information you're looking for about what each thread is doing.
On 10/20/05, Jason Walker <[EMAIL PROTECTED]> wrote:
When I run 'ps aux' I get this:
root 964 0.0
can anyone recomend a good iax softphone??___
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We have a turn-key solution available that does exactly what you are asking
for. You can reach someone for more information at 415.442.4010.
TKS
Paul
[EMAIL PROTECTED]
>
> trixter aka Bret McDanel wrote:
>
> >I am tasked with evaluating ready made solutions for a voip provider.
> >Does anyon
Jay Milk wrote:
I'm having the following recurring problem with asterisk:
When for any reason one of my SIP providers fails to register (i.e.
internet connection dropped), ALL SIP channels fail. This means that,
for example, when my internet connection is out, none of my internal sip
phones reg
Widyachacra Rajapaksha wrote:
dear friends,
what is de best CDMA card or module made for Asterisk?
--
---
the path to freedom.
--- 2.6.13-gentoo-r4
___
--Bandwidth and Coloca
Hi,
I have a potential client who has legacy alarm systems which use modems
to transmit encoded data to a remote location through the PSTN. They
wish to replace the 'PSTN' bit with an IP link.
I am aware that it would be best if the data was transmitted directly
over IP rather than modulated
>Hello all,
>yes there is a lot of information about this on the wiki and in past posts on
>this list but have not found anything that has solved my problem.
>setup is:
>phone-->PAP2-na-->asterisk v1.0.9(in house on local subnet dtmf is
>inband)--->PSTN--->Telisip>asterisk box at colo v1.0.9
OK, yet another thread is closing, where I am the only poster :). For the
record, I find out that starting asterisk with -p option (realtime) gives
the following table, which now makes more sense. But what is surprising for
me is that the load of the server was close to 0 in my original post, th
On Saturday 22 October 2005 14:07, Jason Lixfeld wrote:
> Is there any difference in the amount of hardware timing
> something like a Wildcard X100P can provide over something like a
> Wildcard TE411P? If someone has a machine that pretty much just does
> very low volume MeetMe, Voicemail, SI
dear friends,
what is de best CDMA card or module made for Asterisk?-- ---the path to freedom.--- 2.6.13-gentoo-r4
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digiu
Hi,
Is there any difference in the amount of hardware timing
something like a Wildcard X100P can provide over something like a
Wildcard TE411P? If someone has a machine that pretty much just does
very low volume MeetMe, Voicemail, SIP + IAX and 2 or 3 channels
worth of codec translat
Sorry, I've worked this out if anybody is scratching their heads on my
behalf.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> David Craigon
> Sent: 21 October 2005 15:46
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Queue_lo
Adtran 750
d4,ami
loop start
fxs 5x4 port
fxo 1x4 port
TE405P
Asterisk 1.0.9 stable
span 1 -> CAC channel bank all FXS
sand 2 -> Adtran 750 20 FXS 4 FXO (new FXO card)
span 3&4 -> outgoing only E&M
A call comes in from the telco on the FXO port & answered on a FXS
(fxoks). If the FXS hangs up, o
I apologize if
this is posted again as I sent it last night, but I don’t see it anywhere
in the list as of now.
I did a quick Google search of the lists and I hope that I
am not asking a question that has already been answered recently.
I have been working on a interface to u
Thanks for the wealth of information. I knew they were off-air
DNS-wise, and this happened before a couple of times. It's just bad
juju to have all your IPs in one block.
I don't think they were reselling, and I actually thought I had a pretty
good report with them -- just have been unable to ge
I was able to resolve the problem to some extent. My out going calls are
working fine when I included the external IP adress in sip_nat.conf file. But
my incoming calls are going voice mail instead of ringing the telephone
attached to my sipura device.
Any help is appreciated.
--kotesh
On 10
I just tried to place a call thru fwdout, works fine.
On Sat, 2005-10-22 at 06:40 -0700, trixter aka Bret McDanel wrote:
Initially I thought this may have been the fiasco last night or the
night before (I forget now) where level3 did a software upgrade and it
went awry. With the pings respo
KRTorio wrote:
>"Is there an easy way to modify the filename of an incoming call's
>recording, or are we stuck to agent-- format given
to us by Asterisk?
Your answer was in queues.conf that's why you only got 1 reply.
Kevin
___
--Bandwidth a
Hey ho,
We have something like that (tailored for huge installations), contact
me off list for more info.
zoa.
trixter aka Bret McDanel wrote:
I am tasked with evaluating ready made solutions for a voip provider.
Does anyone have any recommendations for software, specifically the
environment
I have a similar setup... I set the canceller on the incoming PSTN
lines, but turn it off on the FXS.
I have no local internal echo over the t1, but moderate over the PSTN.
I managed to tweak it a little and most of my outbound (local side)
echo is minimized, but still there a little. I have no
Bret,
See my recent post:
http://lists.digium.com/pipermail/asterisk-users/2005-October/130542.html
I'll send you an email off list with the features and future roadmap.
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
trixter aka Bret McDanel wrote:
I am tasked
I had this, my problem turned out to be in zapata.conf on the receiving
end.
I'll do the KS, right now I am using LS. Any particular reason to use
KS? The LSCPD on the adit seems to work fairly decently.
Now I just need to work out some echo, although I have done milliwatt
tests to a local lin
Hello,
I usually use
exten =>
s,1,SetVar(MONITOR_FILENAME=/var/spool/asterisk/q/QSAMPLE-${UNIQUEID})
exten => s,2,Queue(q-sample|nt|||60)
and it seems to work, then use QueueMetrics to keep track of who was
talking to whom, instead of using the Agents monitoring.
Bye
l.
On Sat, 22 Oct
> Initially I thought this may have been the fiasco last night or the
> night before (I forget now) where level3 did a software upgrade and it
> went awry. With the pings responding I now wonder.
>
> It still could be this (all symptoms from the same problem). I am
> thinking about signing up for
Jay Milk wrote:
> I'm having the following recurring problem with asterisk:
>
> When for any reason one of my SIP providers fails to register (i.e.
> internet connection dropped), ALL SIP channels fail. This means that,
> for example, when my internet connection is out, none of my internal sip
>
Initially I thought this may have been the fiasco last night or the
night before (I forget now) where level3 did a software upgrade and it
went awry. With the pings responding I now wonder.
It still could be this (all symptoms from the same problem). I am
thinking about signing up for FWD-out an
Hi all,
I have a public ip in Linksys RT31 (2 FXS port + 3 swtich port + 1
uplink port). I want to add behind it, a Linksys pap2 (uplink port + 2
FXS port) with private ip.
I understand that I have to configure Port forwarding or port
triggering (really I'm not sure which one).
Is someone already
> Currently (and since yesterday evening), sipmedia.com/myphonecompany.com
> is completely off the radar. No DNS entry found -- not even a
> name-server. They've had this sort of massive failure before, but this
> is one of the longest for all I can tell. While that's a major problem,
> it also
When get sip respond 6xx ( such as 603 decline), I want asterisk to
play a voice file to the caller, how to do this in extensions ?
for example, when get 603 respond, play decline.gsm to caller
when get 604 respond, play doesnot-exit.gsm to caller
when
> I'm having the following recurring problem with asterisk:
>
> When for any reason one of my SIP providers fails to register (i.e.
> internet connection dropped), ALL SIP channels fail. This means that,
> for example, when my internet connection is out, none of my internal sip
> phones register,
> > Mine stopped working sometime back in Feb. I just made the changes
> > so everything points to fwdOUT.net now, but it still seems to fail.
> >
> > Using a sniffer, I see packets going out, but none coming back. I
> > have a firewall, but 4569 has been opened, and I'm not seeing denys
I'm having the following recurring problem with asterisk:
When for any reason one of my SIP providers fails to register (i.e.
internet connection dropped), ALL SIP channels fail. This means that,
for example, when my internet connection is out, none of my internal sip
phones register, and I'm una
Chris Bagnall ha scritto:
What's the best way to link them up to * ? SIP or SCCP? I've trawled through
the mailing list and it seems opinion is divided on the topic, but I
understand there's been quite a lot of work on *'s SCCP module over the last
few months.
Yes, the chan_sccp (http://chan
I am tasked with evaluating ready made solutions for a voip provider.
Does anyone have any recommendations for software, specifically the
environment will be a chargable voip provider (ie broadvoice, vonage,
etc). They wanted me to see what was there and write something if
nothing they like exists
Hi all,
Today i try to use asterisk to make SIP call between two office A and B.
At the office A, i use [EMAIL PROTECTED]. testA is softphone
(for testing, i use sjphone) which is running in PC with IP: 192.168.4.100.
At the office B, i use [EMAIL PROTECTED]. testB is softphone
(for testing,
Hello, we can help you with this. Setting up large Asterisk and SER
clusters is our speciality. We've done 4 high availability and fully
redundant systems this year, as well as quite a few smaller ones.
NAT traversal is no problem; we can do this on SER. We generally don't
use STUN, as it's no
Regarding my previous post:
>"Is there an easy way to modify the filename of an incoming call's
>recording, or are we stuck to agent-- format given to us by Asterisk?
>
> There seems to be neither an equivalent ChangeMonitor() application for incoming, nor you can tweak the recording's filen
Hi:
I am using a voip provider that has both sip and iax
in their system. I can make calls using sip with no
problems. But I can't make calls from the same voip
provider using IAX. I get the following message while
the call is in progress:
dial [EMAIL PROTECTED]
-- Executing Dial("OSS/dsp"
Hello all,
I'm about to source a pair of 7960Gs to test with Asterisk prior to a demo
to a new client next month. I've never used Cisco phones, let alone tried to
make them play nice withly with *.
According to our supplier, they either come with a SIP licence or a CCM
licence (which from what I'
On Sat, 2005-10-22 at 11:28 +0200, Olle E. Johansson wrote:
> A lot of this can be done by non-programmers and will greatly help us
> moving forward with 1.3 after the release of 1.2. Join us in
> #asterisk-dev or #asterisk-bugs on IRC if you have any questions.
you may want to mention that this i
Please ignore my message. Problem solved. Using a call-by-call vendor in
Germany caused this long period of silence. Without it everything is
working as expected.
Have a great weekend.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
___
-
Someone wrote me off list:
> I would like to be able to help, but I'm not a C programmer - is there
> any other way I can assist the project?
>
There are many ways! Testing new patches, making sure they are
documented properly, that they work as expected. Making sure the Wiki is
up to date with 1
Hi,
I am using asterisk and chan-capi_cm CVS as of yesterday, but the
problem has been for a long time.
After dialing a number via
dial(CAPI/G1/0123-122)
it takes roughly 10 seconds to hear the first ringing tone. Adding
option "b" is not feasible, as it does not fix the dialout
Hi,
this is what I continuously see into the logs:
Oct 22 10:26:07 NOTICE[26614]: chan_sip.c:6924 handle_response: Failed
to authenticate on REGISTER to
';tag=as77222f33' (tries '2')
Oct 22 10:26:26 NOTICE[26614]: chan_sip.c:4055 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out
Hi Peter,
> Does anybody know how I could make contact with them other than the
> published phone/email on their webpage?
>
I can offer you the following details of Mr. Junghanns himself:
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +
Hello gurus,
After successful installation of Areski. I am
having few problem before I can do any test dial-outs.
When I try to create sip/iax
friend from web interface it says"Could not open buddy file
'/etc/asterisk/additional_areskicc_sip.conf'
I tried creating the file manually with
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