Hi Remco
Thanks for the response. I am running Suse 9.3 kernel 2.6.11.4-20a-default.
Will check on the auto update, but I don't think so.
Cheers
Terry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: 18 August 2005 08:54 PM
To:
Steve Gladden wrote:
I have a small pile of them from customers who were too lazy to send them
back after switching to our local voice service...
Is there any hope of ever using these things with Asterisk?
Vonage does not want them back and they won't unlock them either.
A terrible shame!
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Terry Wade
Skickat: den 19 augusti 2005 07:08
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Ämne: RE: [Asterisk-Users] Zaphfc.ko module error
Hi Remco
Thanks for the response. I am
hello
why asterisk starts listening on all ports
and he is trying to listen messages from 5060.
/etc/asterisk/sip.conf
bindport=5070
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Hi all,
is it possible to monitor RTP protocol (latency, errors, ...) by
Asterisk or other software.
Thanks for answer,
Bob.
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi,
I tried to connect cisco 7910 into asterisk system using chan_sccp.so.
But I got a major issue :
- when I called from 7910 to another sip phone in the same asterisk
server, the call took place normally.
- when I called from 7910 to another sip phone in different asterisk
server, the call
Still get same:
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)
Angus
- Original Message -
From: Fábio Sakai [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 18,
John Gruber wrote:
Earlier this afternoon I had this working
exten = 2890,1,Answer
exten = 2890,2,GoTo(12)
exten = 2890,12,Wait(1)
exten = 2890,13,Festival('I can say numbers like')
exten = 2890,14,SayNumber(1230001,f)
exten = 2890,15,Wait(1)
exten = 2890,16,HangUp
I was so very proud of
On Thursday 18 August 2005 22:27, Matt Hess wrote:
Just call a milliwatt..?
you have a number?
I'm also willing to pay my regular fees to my provider for those 3-4 minutes
of testing.
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Steve Underwood wrote:
Tamas J wrote:
Hello,
I have seen that SpanDSP supports V.17 faxing, however when I tryed to
send pages, I eneded with very ugly pages (unreadable). Did anybody else
try that?
Yes, I checked frame slips and clocking on PRI, everything has to be OK.
Regards,
On Fri, Aug 19, 2005 at 09:15:02AM +0200, Christoph Eicke wrote:
On Thursday 18 August 2005 22:27, Matt Hess wrote:
Just call a milliwatt..?
you have a number?
In your dialplan:
exten=1800645549288,1,Milliwatt
MusicOnHold will also do. In fact it will probably be a better emulation
of an
Hi,
On 9:04:57 August 19, 2005 stevanus [EMAIL PROTECTED] wrote:
Hi,
I tried to connect cisco 7910 into asterisk system using
chan_sccp.so. But I got a major issue :
I've tried different versions of chan_sccp, yet the result were still
the same.
Which version of chan_sccp did you use?
But when I load Asterisk it doesn't complain. Get 2 warnings:
[chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource:
/usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed
Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module
Hi,
I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp.
Is it the same as chan_sccp from chan-sccp.berlios.de?
Best Regards,
Stevanus
Stefan Gofferje wrote:
Hi,
On 9:04:57 August 19, 2005 stevanus [EMAIL PROTECTED] wrote:
Hi,
I tried to connect cisco 7910 into
On Fri, Aug 19, 2005 at 01:18:14AM +0100, Matt King wrote:
Hello,
I'm looking to develop some custom AGI that will be MySQL intensive. It
appears Asterisk supports many different development environments. Which
would be best suited for Asterisk and MySQL?
It's generally fastest to
Remarque : message transféré en pièce jointe.
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur
Bohuslav Coufal wrote:
Hi all,
is it possible to monitor RTP protocol (latency, errors, ...) by
Asterisk or other software.
Try http://tstat.tlc.polito.it/
quote
Tstat, a passive sniffer able to provide several insight on the traffic
patterns at both the the network and transport levels.
On 10:10:54 August 19, 2005 stevanus [EMAIL PROTECTED] wrote:
Hi,
I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp.
Jep, it is... If you had problems with this, your chance for a solution is
higher at the chan-sccp-users list... :-)
Regards,
Stefan
oops, got it. Thanks for the info.
thanks
Somesh
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 18, 2005 7:56 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help on AGI running
i
Hi,
Haven't noticed that there exists one :P
Thanks for the pointer anyway ;). Gotta sign up pretty soon :)
Best Regards,
Stevanus
Stefan Gofferje wrote:
On 10:10:54 August 19, 2005 stevanus [EMAIL PROTECTED] wrote:
Hi,
I used chan_sccp from
Actually they have. Interrupt sharing for one. Interrupt overhead for
another. Drivers which are optimized for minimum latency instead of a balance
between latency and ability to share interrupts and overhead for a third.
-A.
Can you explain a little bit more? I thought they don't share
I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz and referenced
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices.
But I could not succeed to start ices-2.0.1 as follows;
-- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1
(Retry 1)
-- Executing
Been there, done that...
I was talking to a high level tech for an hour...
Basically, they calculate the need for throttle based on the length of
time a modem is busy, not the amount of data that is transferred.
So for example, asterisk not involved, If I view an axis camera feed
remotely,
They're using the same hosted servers with different billin schemes.
When I last looked there was a huge difference in ping times and
voipbuster when I tested it was very much up and down in
responsiveness. I thought they were in Germany (or at least Europe)?
VoipBuster is a service from
Finarea SA
Po Box 5648
Lugano 6901 CH
But you are correct. The servers are supposedly housed in germany.
Even accounting is the same as I couldn't get a voipcheap and a
voipbuster account with the same username.
-Don
-Original Message-
From: [EMAIL
VoipBuster is a service from
Finarea SA
Po Box 5648
Lugano 6901 CH
But you are correct. The servers are supposedly housed in germany.
Even accounting is the same as I couldn't get a voipcheap and a
voipbuster account with the same username.
I must have misunderstood about who is using
2005.08.19
Version 1.3
* IPManager now
supports SIP, IAX and Zap extensions and trunks.
* Music on Hold
Groups can be defined and assigned.
* MP3 files can
be uploaded directly to Asterisk
FREE download: http://ipsoftware.thorben.dk
Armin Schindler schrieb:
Hi,
this should already be fixed in current CVS version and will be part of
next release.
Maybe you want to try it. (Note: capi.conf and dial syntax has changed)
Armin
Yes, thank you. updating to cvs-version did solve the issue :)
tobias wolf
Thanks for all the replies! Looks like I was shipped the wrong
powersupply. I figured as much, cause when I first plugged it in it
took a while to boot, and started to smell something burning. :(
Time to RMA it back and get them to ship me the proper parts.
PB
Paul Belanger wrote:
Can
Hello,
I have several *
serversbehind a SER server (in a local ip range).The
SERserveris also publicy reachable. On the other site, I have SIP
clients that are behind another NAT or in the same NAT range as the * server.
Can someone give me some directions/hints etc. on how to make this
Just call a milliwatt..?
you have a number?
I'm also willing to pay my regular fees to my provider for those 3-4 minutes
of testing.
Milliwatt generators are essentially part of every telephone company's
central office switch, and typically are provided by the telco for their
installers
Very Highly Internested
Any chance you could zip or tar your content up and email it to me or give
me a link to grab it?
Maybe I could help you get it hosted again too ifyou need that.
Thanks!!!
Steve
Steve Gladden wrote:
I have a small pile of them from customers who were too lazy
Hello guys,
Can someone help me???
I was wondering to know how to point a agi message to a specific channel??
For example.
caller -- * -- agi script(Send message)---called
In this above case in my script every thing is all right, it is, I can send
the message correctly to
The catch all extension I use is _. (match everything).
That's a nono, but that is not the problem :-)
and also tried _X. (match any numeric) don't match special extensions.
Much better!
From voip-info.org on the cmd VoiceMail page:
If, during the recording the caller presses:
'#' - or
Hi Tom,
thank you. I solved my problem ... but it was really painful because of
the notify process inside the snom phones seem to crash if you send the
wrong commands :-(. So thausends ;-) of reboots were needed...
That's my solution now:
[agents-loginout]
exten = 6011,hint,DS/6011
exten =
It was my own stupid fault for installing the asterisk version available in
the SUSE distribution and then downloading and installing the latest
version. Another thing not to do!
Uninstalled old and re-installed asterisk and it worked!
Angus
- Original Message -
From: Angus Comber
I have officially engaged in a pissing contest with the local Telco over
PRI calling name delivery.
The telco publishes their calling name delivery over PRI feature as
being bellcore gr-1367-core compliant.
The gr-1367-core spec states that the calling name is to be included as
a facility IE in
Actually they have. Interrupt sharing for one. Interrupt overhead for
another. Drivers which are optimized for minimum latency instead of a balance
between latency and ability to share interrupts and overhead for a third.
-A.
Can you explain a little bit more? I thought they don't share
On Thu, 2005-08-18 at 21:08, [EMAIL PROTECTED] wrote:
Hello All,
I was recently fighting with an optimum online connection in NY.
I finally got in touch with someone that confirmed they are throttling
my upload connection.
I know they watch for people doing peer to peer file sharing and
don't know if Asterisk can do it, but ethereal can. Ethereal is an open
source protocol analyzer. Download it from www.etheral.com
On Fri, 2005-08-19 at 02:55, Bohuslav Coufal wrote:
Hi all,
is it possible to monitor RTP protocol (latency, errors, ...) by
Asterisk or other software.
HI,
I installed a tdm440p and a monoBRI in the same Dell machine (PowerEdge
600SC) but after typing ztcfg -vv the server screen is filled with tons
of the following lines:
Aug 19 11:54:42 pippo kernel: zaphfc: bchan rx fifo not enough bytes to
receive! (z1=3200, z2=3193, wanted 8 got 7),
Just 2 questions:
1. Is there a plan for supporting mISDN, CAPI and SCCP exts. and trunks ?
2. Is it compatible with asterisk STABLE 1.0.X ?
regards,
Nenad
Message: 13
Date: Fri, 19 Aug 2005 12:40:22 +0200
From: Thorben Jensen [EMAIL PROTECTED]
Subject: [Asterisk-Users] IPManager
Tamas Jalsovszky wrote:
Steve Underwood wrote:
Tamas J wrote:
Hello,
I have seen that SpanDSP supports V.17 faxing, however when I tryed to
send pages, I eneded with very ugly pages (unreadable). Did anybody else
try that?
Yes, I checked frame slips and clocking on PRI, everything
Innocent Evil wrote:
Hello,
I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium
website)
SIP user (100) is calling another SIP user (101).
As 101 is not online, my SIP server is redirecting that call to Asterisk.
Asterisk forward it to 101's voice mail box.
SIP user
Matthew, thanks for answering me.
I think, I have found the problem.
Yes, the 2 liscenses was intalled.
If I make a phone call from SIP phone, asterisk use 1/1 encoder/decoder..
If I make a call and asterisk forward to voice mailbox.. just before it
starts recording voice mail, it use 1/1
So it was all working well and then suddenly I'm unable to get incoming
calls from BV. Outgoing is fine. I'm using AAH.
I have the following settings;
[EMAIL PROTECTED]:PASSWORD-GOES-HERE:[EMAIL PROTECTED]/2208
[broadvoice]
username=9738281625
user=phone
type=peer
secret=PASSWORD-GOES-HERE
Kevin P. Fleming wrote:
Falck Kenneth wrote:
My persistent variables (_XXX or __XXX) don't persist when I dial a
Local extension. I'm doing a forked dial where the other channel is
SIP and the other Local. Is this a known problem? Using Asterisk
1.0.9.
Variable inheritance is a CVS
Falck Kenneth wrote:
Thanks, I was misguided by
http://www.voip-info.org/wiki-Asterisk+Variables which didn't mention
this.
Yeah. Nobody ever seems to mention on the Wiki when a specific feature
became available.
___
Asterisk-Users mailing list
No memory leaks or choppy sound?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Wiley Siler
|Sent: Martes, 16 de Agosto de 2005 07:00 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Plantronics
Falck Kenneth wrote:
Thanks, I was misguided by
http://www.voip-info.org/wiki-Asterisk+Variables which didn't mention
this.
You are more than welcome to edit the page to make it obvious to the
next reader :-)
I guess there is no way to achieve what I want to do with the stable
version?
BTW, any sip or iax softphones with skin support, for example, for putting
you logo in for semi-branding for internal use?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Anton Krall
|Sent: Viernes, 19 de Agosto de 2005 09:54 a.m.
|To: 'Asterisk
Hello,
I'm again there
I have also a Problem with Overlap
Digits...
I'm getting a Call from my Telco to the extension
1234 and i will forward it with exten = 1234,1,Dial(Zap/g1/987654), but
asterisk is not dialing 987654, asterisk is dialing 987654 and as overlap digits
1234.
so i see on
On 8/19/05, Falck Kenneth [EMAIL PROTECTED] wrote:
I guess there is no way to achieve what I want to do with the stable
version? I.e. pass call-specific variables when dialling through a Local
channel. Now I can only see the original Caller ID and the destination
extension, but not the other
Hi,
In this link is the script Suse
http://www.leals.com/~mm/asterisk/asterisk_suse.sh
On 8/18/05, James Oakley [EMAIL PROTECTED] wrote:
On Wednesday 17 August 2005 3:04 pm, Tzafrir Cohen wrote:
On Wed, Aug 17, 2005 at 01:27:08PM +0300, [EMAIL PROTECTED] wrote:
Hi!
I'm trying to
Kevin P. Fleming wrote:
Falck Kenneth wrote:
Thanks, I was misguided by
http://www.voip-info.org/wiki-Asterisk+Variables which
didn't mention
this.
You are more than welcome to edit the page to make it obvious
to the next reader :-)
You're quite right - I added a little note there
During the install [EMAIL PROTECTED] will add all the
lines on your card to group 0. Make a trunk for g0 and
it will use all the lines.
--- Sascha Ferley [EMAIL PROTECTED] wrote:
Hi,
I am just trying to figure out how to setup a TDM04B
card for
incoming/outgoing calls. I have 4 lines,
Hello,
I am hoping someone might be able to help me with this issue. Right
now I am testing a X100P card in asterisk connected to a Lucent
Partner ACS 3.0 PBX. The card, obviously, acts as an SLT to the
system. The card interacts wonderfully with the system. The problem I
am having,
Matt Florell wrote:
We struggled with this a couple years back before CVS_head
had that function. What we ended up doing was using the
CallerIDName field for a 20 character unique identifier and
we used the callerIDnum as usual (telcos in the US only use
callerIDnum anyway).
Thanks! I
Have you restarted Asterisk to see if that helps?
What does 'sip show registry' show?
Tom
On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote:
So it was all working well and then suddenly I'm unable to get
incoming calls from BV. Outgoing is fine. I'm using AAH.
I have the following
OK, this seems to be their side at first look. My friend whom has the
same setup as me is also having the same problem.
Opinions?
Mark Phillips wrote:
So it was all working well and then suddenly I'm unable to get incoming
calls from BV. Outgoing is fine. I'm using AAH.
I have the
Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is on
a call, other incoming calls will ring through to her phone and can be
answered. So far I have only been able to get this working by using the
call-waiting function, which is cumbersome and does not properly allow
On 12:07, Fri 19 Aug 05, John Mensel wrote:
Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is
on
a call, other incoming calls will ring through to her phone and can be
answered. So far I have only been able to get this working by using the
call-waiting function,
If you register 2 line buttons with the same SIP account, then the
second call will go to the second button. Also search the list for
this.
On 8/19/05, John Mensel [EMAIL PROTECTED] wrote:
Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is on
a call, other incoming
Using contexts, and making sure which device is coming in to where.
On 8/19/05, Angus Comber [EMAIL PROTECTED] wrote:
Hello
If callerid is not available on an external line, how can you tell if call
is incoming or outgoing?
Angus
___
Sean Rima wrote:
Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto the asterisk box
Sean
I just bought 12 of them to
Too many people with misconfigured autoresponders...
Latest is Make Zuzlak, who has announced he'll be annoying everyone
until August 22nd.
If people are going on holiday please do one of 3 things:
1. Don't use an autoresponder
or 2. Use one that isn't broken.. ie. knows what the Precedence:
Dennis Gilmore wrote:
Sean Rima wrote:
Does anyone have any experience of these, I have been offered one and am
thinking of adding sticking it onto the back of my Asterisk box and just
ignore the WAN port if possible, It would be to stick my exisiting
phones onto the asterisk box
Sean
I
Il Neofita wrote:
I put these lines on features.conf in asterisk CVS-v1-0-08/16/05
[featuremap]
blindxfer= ##
automon = *1
atxfer = *2
You need to use CVS-HEAD for those features. You are using 1.0.x CVS
___
Asterisk-Users mailing list
Has anyone tried attaching calling a Dialogic D/41JTC-LS (analog) device on another system from an asterisk system with TDM10B?
Calling to asterisk from the outside, asterisk correctly dials the
internal line and makes the connection to the Dialogic system. A
few seconds later Asterisk debug info
Hi,
I am using Asterisk cmd VoiceMailMain to manage voice mail.
Problem is, voice mail box can't read password sent from SIP phone, but I
don't have any problem to read password from the handset attached to my
asterisk box.
Your help will be greatly appreciated.
Steve Gladden wrote:
Very Highly Internested
Any chance you could zip or tar your content up and email it to me or give
me a link to grab it?
Maybe I could help you get it hosted again too ifyou need that.
Thanks!!!
Steve
I would love to have a tarball of my web stuff. I didnt know
It sounds to me like an issue of transmitting DTMF tones from the SIP
phones.
There are several methods that can be used to accomplish DTMF from SIP
phones. Of course, you may ask why it isn't just sent as audio (like a
regular POTS phone would.) What happens if you are using a SIP phone,
Rollin Weeks wrote:
Has anyone tried attaching calling a Dialogic D/41JTC-LS (analog) device on
another system from an asterisk system with TDM10B?
Calling to asterisk from the outside, asterisk correctly dials the internal
line and makes the connection to the Dialogic system. A few seconds
I ran into the same problem the other day and just went back to non head
version It would be nice to figure out why it does this.
- Original Message -
From: Nico Giefing
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, August 19, 2005 9:20 AM
Subject:
my sip phone have dtmf relay: rfc2833
asterisk sip.conf have dtmf relay: rfc2833 in associated context.
I tried with Inband.. but g729 doesn't support it. I have g729 liscence from
digium
I havn't try with INFO yet.
I prefer to have rfc2833 as dtmf relay.
Is there any other thing that can cause
Fellow list members,
I have run into an issue where I encounter a delay
at the beginning of a phone conversation when I make outgoing calls through
Asterisk with an Aastra 9133i hardphone. This is most noticable when I
call a voicemail system with the Aasta and then with a land line or
Follow-up on this
We have tried several things without success.
Digium responded that the problem was NMI (non-maskable interrupts) and
told me to boot linux with the nmi_watchdog=0 option
It did not solve the problem.
Finally I replaced the TDM card with an older one (revision F) 2FXO 2FXS
Hi,
I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to
place and receive
calls. I have successfully configured Asterisk to route inbound calls and send
them to the correct extension, but I can't get outbound calls to work. I have
Asterisk successfully registering
Brian Deep wrote:
[from-sip]
exten = _9.,1,Dial(SIP/[EMAIL PROTECTED])
exten = 201,1,Dial(SIP/201)
exten = 202,1,Dial(SIP/202)
try exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
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Asterisk-Users@lists.digium.com
If you can get an rtp debug while your pressing digits I can see if
maybe your device is sending the digits incorrectly.
/b
On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote:
my sip phone have dtmf relay: rfc2833
asterisk sip.conf have dtmf relay: rfc2833 in associated context.
I tried with
You are calling a sip host that you do not have defined in sip.conf.
I think the line should look like this
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
This will force * to look in its sip.conf file for a stanza called
[vonage] which you have rather than [atlas-east.vonage.net] which
Well, some smarty pants lady at broadvoice, claim that the problem is in our
end, well, I have taking my box out of the picture, I went to bv control
panel and have forwarded the calls to my home phone number, she STILL
insist that the problem is my asterisk box, the one I deleted the
Broadvoice
I have a TDM400P with some FXO ports, and I wanted to connect the two POTS
lines from my Pipeline-75 ISDN router into the FXO interfaces on my Asterisk
server.
Hooked it up, seemed fine, called in and it answered. The problem is when
the call is hung up on, the FXO port never drops. So of
Do you need a hangup in your dialplan?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard
Leadmon
Sent: Friday, August 19, 2005 4:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Ascend Pipeline
OK, I nailed it, it's working now. If any are curious, seems in the P75
there is an option called Forward Disconnect and by default it's set to NO,
and needed to be set to YES so it sends the disconnect to the TDM card.
---
Howard Leadmon
http://www.leadmon.net
-Original Message-
Same here. When I turn on their forwarding or their VM my inbound calls
complete. Thing is though, why suddenly now does it not work.
I can't believ that 3 of us have been messing with our boxes at the same
time? I reckon they made a change at their end and won't fess up.
Manny A. Wise
OK, now I know of 5 peeps that suddenly are having this problem.
It has to be them right?
Mark (in the rainy end of NNJ)
Manny A. Wise wrote:
Well, some smarty pants lady at broadvoice, claim that the problem is in our
end, well, I have taking my box out of the picture, I went to bv control
I don't believe that this issue is with Asterisk.
The issue is that the phone does not set up the RTP stream until it receives the
"200 OK". Asterisk sets up the RTP stream when it receives, or sends,the
message with SDP (either INVITE message, 180 response or 183 response), as per
the
Mark,
Thank you very much!!! That was exactly it. My config files now look like the
following and I can send and receive calls using Vonage.
sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
context=vonage-out
disallow=all
allow=ulaw
allow=alaw
nat=yes
register=:[EMAIL
Does anybody know what would be causing the errors
below?
I get these errors continuously until asterisk finally
quits. This happens when I make 20 simultaneous SIP
calls with the Dial Command.
chan_oss.c:291 sound_thread: Failed to write sound
chan_oss.c:200 send_sound: Unable to read output
Other than below:
Got RTP packet from x.y.z.sip_phone:10006 (type 18, seq 25407, ts 191360,
len 40)
Sent RTP packet to x.y.z.asterisk:10006 (type 18, seq 63928, ts 193744, len
20)
I dont see any message while sending digits.
-Original Message-
From: [EMAIL PROTECTED]
Sent: Fri,
Hello list,
We have some kind of a problem with our Asterisk installation. We
want to be able to publish different caller id when placing outbound
calls through the PSTN. We have Asterisk with TE410P and T1 from FDN
Communications. The problem is that all our outbound calls show our
main
Thanks Eric,
I tried the changes to zapata.conf. I still get the hangup. It makes me wonder
if the Dialogic card is sending a hangup tone to the FXO module. It seems to
work OK if I use an analog phone instead of linking to the Dialogic card.
RollinOn 8/19/05, Eric Wieling aka ManxPower [EMAIL
i am install asterisk in gentoo linux,
#emerge zaptel
#emerge asterisk
#modprobe zaptel
#modprobe wcfxo
#asterisk -vvvc
localhost ~ # asterisk -vvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.8, Copyright (C)
Waldo Rubinstein wrote:
switchtype = dms100
signalling = em_w
group = 1
context=inbound
callerid=asreceived
channel = 1-24
You cannot set your own Caller ID on anything except PRI.
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OK, now that you have it working put it into the WIKI!!
Mark
Brian Deep wrote:
Mark,
Thank you very much!!! That was exactly it. My config files now look like the
following and I can send and receive calls using Vonage.
sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
I'm using their dca proxy and have not had any problems at all today
with them.
I've got 201 and 212 DID's with them and both have completed incoming
calls throughout the day today.
On 8/19/05, Mark Phillips [EMAIL PROTECTED] wrote:
OK, now I know of 5 peeps that suddenly are having this
I may be wrong here, so if anyone else here knows contrary, please feel
free to jump in and correct me. ::dons his asbestos armor::
When we first deployed * we were coming from an analog channel bank
setup (hooked into our old PBX as analog lines.) I was able to connect
* to the T1 and use
Working fine here in the Northwest. Actually I haven't had a single problem
with them since the dreaded Global Crossing fiasco...
-Chris
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There are 2 possibilities:
1) Your PRI provider does not have the overide settings correctly set on
your PRI.
2) you are not setting the callerid correctly in your dialplan.
You indicate that your provider indicates that they have it set up
correctly. You have a 50/50 chance that this is indeed
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