I run unrder CentOs 4.3, and have asterisk, asterisk-addons,
asterisk-sound, zaptel, zapata and libpri installed.
I defined so far 2 accounts; sipgate works fine for incoming and
outgoing calls.
The 2. account (peoplecall) makes problems:
- The register statement works find and quering in
Wolfgang Paul Rauchholz wrote:
exten = 5550873,1,Dial,SIP/30|30|r
exten = 001,1,Dial,SIP/30|30|r
|r is evil - Don't use it.
I would be willing to bet large sums of cash this problem will go away
if you simply remove the |r (and reload)
Jeremy McNamara
--- Kamran Ahmad [EMAIL PROTECTED] wrote:
Thanks alot for your answer Florian
I have a question in this case when call is
transfered
from loadbalancing-server to server01 or server02
what
will be media Path? media will be routed through
loadbalancing-server or it will not use
Thanks
Hi,
Kamran Ahmad wrote:
I have a question in this case when call is
transfered
from loadbalancing-server to server01 or server02
what
will be media Path? media will be routed through
loadbalancing-server or it will not use
loadbalancing-server anymore
On Fri, August 11, 2006 4:26 pm, Wolfgang Paul Rauchholz said:
allow=g729
allow=g723
Do you have the g729 and g723 codecs installed? They are not installed
with Asterisk by default.
cYa,
Avi
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On Friday 11 August 2006 18:26, Wolfgang Paul Rauchholz wrote:
Aug 11 08:00:24 WARNING[2612]: channel.c:2706
ast_channel_make_compatible: No path to translate from
SIP/30-09dfbdb8(4) to SIP/3470075-09e01778(256)
Aug 11 08:00:24 WARNING[2612]: app_dial.c:1595 dial_exec_full: Had to
drop
We did this for a customer completely in the dialplan, with the Asterisk
internal database.
I don't have the coding here, but I know it involved the read command,
followed by putting the number keyed into the internal database.
(as something like ah/mobile)
later,
PaulH
On Fri, 2006-08-11 at
see http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
perhaps it can help you
2006/8/11, Rich Adamson [EMAIL PROTECTED]:
Shaun Hofer wrote: ok maybe I can explain my problem better. There two trunks both have the same
details except one is type=peer (and only does ulaw) and the other
When I put a call from an H323 phone to an asterisk box equiped with oh323
I cannot hear any ring tone on the phone (NetMeeting).
When call is answered everything is OK, i can hear and the other person too
can hear me.
I found this:
https://skylab.inaccessnetworks.com/mantis/view.php?id=79
Hi there,
i want to use another context, when i do a atxfer, but i dont know
when/where to set that magic variable. in the dialplan,
any examples?
Regards,
Kai Ober
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asterisk-users mailing
Hi
After a month or so using Asterisk we've had or first downtime period
due to a faulty RAM chip on the server, so we're starting to think about
the possible high-availability solutions.
We still haven't gone completely VoIP: we're using Asterisk in
conjunction with our old PBX and analog
I need help with SIP,RTP port forwarding , I can connect using SIP and
make calls but there is no audio even though my kernel has sip support and
I suspect that it has to do with iptables.
Siqhamo Sifo
NewLunar Technology Solutions
5th Floor
SmartXchange
5 Walnut Road
Durban
[EMAIL PROTECTED] wrote:
Hi
After a month or so using Asterisk we've had or first downtime period
due to a faulty RAM chip on the server, so we're starting to think
about the possible high-availability solutions.
Hi
If you can afford it, below will give you total fault tolerant solution.
Hi,
I have the following rules:
exten = 4441,1,NoOp(--- ${CALLERID} calling on capi-extern (${EXTEN}) ---)
exten = 4441,2,Goto(dialin-privat,s,1)
exten = 4441,3,Hangup
[dialin-privat]
; Log incoming calls
exten = s,1,LDAPget(CALLERIDNAME=daheim)
exten = s,2,NoOP(--CALLERID=-${CALLERID}-,
It's because the standard CDR engine uses the last ${EXTEN} value as the
destination number
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthias
Fechner
Sent: Friday, August 11, 2006 6:08 AM
To: asterisk-users@lists.digium.com
Subject:
just disable iptables - if use redhat/fedora#service iptables stopOn 8/11/06, Siqhamo Sifo [EMAIL PROTECTED]
wrote:I need help with SIP,RTP port forwarding , I can connect using SIP and
make calls but there is no audio even though my kernel has sip support andI suspect that it has to do with
If someone asked your for help finding their front door key, would
your proposed solution be to leave the door unlocked?
On 11/08/06, Rosli Sukri [EMAIL PROTECTED] wrote:
just disable iptables - if use redhat/fedora
#service iptables stop
On 8/11/06, Siqhamo Sifo [EMAIL PROTECTED] wrote:
I
Hi guys,
i need to know if there is any gui application out there for asterisk
which provides a live report of calls, channels, agents, conferences
etc?-- RegardsRizwan HishamSoftware Engineer
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Hello,
I have a polycom 500 phone. While testing our queue and waiting to speak
with operator my phone after about
2 minutes just disconnects.
Here is sip debug.
I cannot find out what the problem might be.
Does anybody can see something strange in it :
-- SIP read from 10.60.10.109:5060:
According to the wiki page http://www.voip-info.org/wiki/view/Asterisk+IAXmodem
There are a couple of ways to integrate Asterisk and HylaFax
with IAXmodem;
IAXmodem as HylaFax modem,
both HylaFax and Asterisk on the same machine
IAXmodem in conjunction with
Check out Flash Operator Panel, gives a pretty interface for watching
parked calls, agents, etc...
On 8/11/06, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi guys,
i need to know if there is any gui application out there for asterisk which
provides a live report of calls, channels, agents,
If you are using a Linux gateway to connect your local LAN to the
Internet, then redirect as follows:
/sbin/iptables -t nat -A PREROUTING -p udp -i ethx --destination-port 5060 \ -j DNAT --to-destination xxx.xxx.xxx.xxx
/sbin/iptables -t nat -A PREROUTING -p udp -i ethx
Do you have audio running during the hold (MOH), or silence?
Could the Polycom (or asterisk) be dropping the call due to inactivity?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak
Sent: Friday, August 11, 2006 6:04 AM
Hi friends, We have installed Asterisk in our organization. We registered with Teliax and got our DID number. We are making calls to USA successfully through Asterisk. We are making outgoing calls to US. But, we are unable to receive incoming calls to our DID number. When I executed the "sip show
What do Teliax support say?
On 11/08/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi friends,
We have installed Asterisk in our organization. We registered with Teliax
and got our DID number. We are making calls to USA successfully through
Asterisk. We are making outgoing calls to US. But, we are
Ricardo, I'm looking for the same thing. Have you tried the patch? Got
any success?
--
Antonio J. S. Brandão
On 7/7/06, Ricardo Martins [EMAIL PROTECTED] wrote:
Hi all. I´m trying to disable this simple thing: I dont want an user to
put a call in hold pressing hook (or flash button). I tryied
Hi,
I am looking for a good affordable USA toll free DID provider for
asterisk.
Thank you
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
dressed in overalls and looks like work."
Thomas Alva Edison - Inventor of 1093 patents,
including the light bulb,
Note that you have:
[teliax]
context=default
but you do not have a default context in extensions.conf for this.
Change the above to:
[teliax]
context=general
**OR** in extensions.conf change
[general]
exten = 3031234567, 1, Answer()
exten = 3031234567, 2, Dial(SIP/105,15)
to:
Crazy Boy wrote:
Hi friends,
We have installed Asterisk in our organization. We registered with
Teliax and got our DID number. We are making calls to USA successfully
through Asterisk. We are making outgoing calls to US. But, we are unable
to receive incoming calls to our DID number. When I
HI lIst,
i'm a little confused about the G option of dial.
which sense hast it to send calle and caller to an context/extension and
dont bridge the calls,
is ther a way to bridge the two parties???
Has anybody a usefull example for this option?
Looking forward to your answers
KAI
Hi Guys,
Happy Friday
I have 2 problems
I run [EMAIL PROTECTED] with some Cisco 7960's
1) DTMF - When I dial a number on the 7960 it works fine. However if I dial
a number that asks 'Dial 1 for this and 2 for that' and I hit 1 or 2 (or
whatever0 the other end acts as though nothing is
This list is Asterisk Users Mailing List - Non-Commercial Discussion. Please post business inquries to the -biz list.AlexOn 8/11/06, Marnus van Niekerk
[EMAIL PROTECTED] wrote:
Hi,
I am looking for a good affordable USA toll free DID provider for
asterisk.
Thank you
Marnus van
On Fri, 2006-08-11 at 15:13 +0200, Marnus van Niekerk wrote:
Hi,
I am looking for a good affordable USA toll free DID provider for
asterisk.
Thank you
Have a look at Teliax, Asterlink and Junction Networks. Don't know about
their pricing. They just seem to have a pretty good reputation
It could be NAT. Either way you need ports 5060-5090 and 1-2 in UDP
open.
- Original Message -
From: Siqhamo Sifo [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, August 11, 2006 5:48 AM
Subject: [asterisk-users] Port Forwarding SIP rtp
I need help with
Try iaxlite at www.iaxtalk.com
- Original Message -
From:
Alyed
Tzompa
To: asterisk-users@lists.digium.com
Sent: Monday, July 24, 2006 11:43
AM
Subject: Re: [asterisk-users] G729
Softphone
As far as I there is no free softphone
that can handle G729
We are still in the testing state for our asterisk box. We currently have a
PRI connected to the box for Local Inbound calling. We are looking for
someone that we can do SIP termination with for our outbound. We have been
looking at BroadVox, but I figured I would ask the list to get opinions
You get what you pay for in life
- Original Message -
From:
dorn
hetzel
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, August 10, 2006 12:11
PM
Subject: Re: [asterisk-users] Warning -
Voiplink.com doesn't deliver - stuckin a
If he suspects the problem is iptables, but isn't sure, disabling it is a surefire way to be certain. That doesn't mean he has to leave it disabled, but it's certainly narrowed down the source of the issue. If at that point he's able to say that iptables is his issue, he can then go and determine
He gave us something that makes us a lot of money. IMOH he has changed the
VOIP world. Lots of good cheap companies out there that only exist because
of Mark. I have paid people on the developers list to create functionality
that I needed and had him submit it to the bug tracker. Why should I
Title: Re: [asterisk-users] Quick One - PHP Script to restart Asterisk
?php
execute "asterisk -rx
'restart when convienent";
?
Not the exact syntax but should be enough
to get you going.
From: [EMAIL PROTECTED] on
behalf of Paul HalesSent: Fri 8/11/2006 3:51 AMTo:
Asterisk Users
I run
HylaFAX on a separate box from my dual PRI Asterisk box, and Asterisk relays the
call to HylaFAX when it detects the fax. It relays the call on a private subnet
with a crossover Ethernet cable with the slin codec. I have over 200 IAXmodems
running on the HylaFAX box, which is an
If you ever see the chance to get the coding again I'm also
interested. I'm setting up something very similar here.
greetings
Tijl Van den Broeck
On 8/11/06, Paul Hales [EMAIL PROTECTED] wrote:
We did this for a customer completely in the dialplan, with the Asterisk
internal database.
I
Has anyone got a working analog connection to POTS
wherein DTMF, *not* FSK is used to send caller id by
the telco switch towards asterisk?
I've tried Asterisk 1.2.10, SVN trunk, and SVN branch
but so far has been unsuccesful.
When asterisk receives a call, I can see from DEBUG
that chan_zap is
Here's the relevent section of my extensions.conf:
### Handle voicemail
exten = _1XX,1,SayDigits(${EXTEN})
exten = _1XX,2,MailboxExists(${EXTEN})
exten = _1XX,3,Playback(vm-nobox)
exten = _1XX,4,Goto(teliax,5013584196,3)
exten = _1XX,103,VoiceMail(b${EXTEN})
exten =
Hi Stephen
I started off using the SPA-3000 for FXO and FXS but could not kill the echo
enough to pass the wife test so I invested in a Diguim TDM400p with 2 FXO
ports, ran the fxotune app and the echo was gone! I now use the SPA-3000 as
FXS and have the pots line plugged in as a life line if
Hi,
I am curious to know whether anyone else can replicate my issue, or
can suggest an avenue of investigation. I have tried it with a number
of Aastra phones, and they all seem to do it (all SIP firmware 1.4.0).
If the SIP call takes the form:
INVITE - TRYING - BUSY HERE
then everything seems
Use j option in those apps.
pbx*CLI show application MailboxExists
-= Info about application 'MailboxExists' =-
[Synopsis]
Check to see if Voicemail mailbox exists
[Description]
MailboxExists([EMAIL PROTECTED]|options]): Check to see if the specified
mailbox exists. If no voicemail context
Or, since you're only touching extensions.conf,
?php system(asterisk -rx 'extensions reload'); ?
instead of a whole restart
Moj
Jonathan k. Creasy wrote:
?php
execute asterisk -rx 'restart when convienent;
?
Not the exact syntax but should be enough to get you going.
pbx-1*CLI show application MailboxExists
pbx-1*CLI
-= Info about application 'MailboxExists' =-
[Synopsis]
Check to see if Voicemail mailbox exists
[Description]
MailboxExists([EMAIL PROTECTED]|options]): Check to see if the specified
mailbox exists. If no voicemail context is specified,
We encountered a problem recently where, if an agent who received a call from a
queue, tried to transfer the caller with the Polycom transfer keys, it would
cause the Queue application to completely lock up.
This bug seems to be similar.
http://bugs.digium.com/view.php?id=7458
and this seems
Yes, this will work, because with the 50x the 'special' cable has a
female power socket in it to plug the wall transformer in to... between
your aforementioned female/female rj45 coupler and the phone itself.
The 301 has the power socket on the back of the phone instead, but it
MAY work with
Hi,
I would like some advice on how to configure my asterisk such that
local users with IAX and SIP extensions can utilize a SIP account from another
server, to make international calls.
In my understanding this would be possible by them dialing a certain
prefix say 888, followed
Hi All,
Asterisk can bind to the IP set with bindip=x.x.x.x in the global
section, but this sets the outbound IP for all outbound SIP traffic.
If I wanted to bind one SIP context to a specific IP address, a
virtual one on the server or a different Ethernet port on the server,
how can this be
Hi Paul, Happy Friday back.
In the config of the extension change the dtmf=XXX
Basically there are three ways dtmf can be transmitted by a sip handset,
choose another or search the voip-info for the options and you'll solve
your problem pretty quickly.
Re: sipgatesorry cant help, you'll
On 7/24/06, Thomas Laurids Pedersen [EMAIL PROTECTED] wrote:
I have the same card, but in my zaptel.conf I have the following linespan=1,1,0,hdb3,crc4as you can see from the status your line is down.BR Thomas Lincoln Zuljewic Silva
Hello all. I have a Digium TE110P board and when I do a 'dial
Is it a bug or is it me?
For the longest time I have been using the feature within voicemail
to call back a number by caller ID.
Never had a problem with it at all.
I just updated to the latest (stable) asterisk from asterisk.org
Option 3 (advanced) then 2 then 1
caller number 7347292615
and
Does anyone know how
I can set/increase the inter digit timeout on Asterisk assisted
transfers?
Doug.
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Hey everyone,
Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk
1.2.10. It has been reported to me when doing an attended transfer the
audio drops out. I ran a few different tests and here is what I noticed.
1. Blind transfers work with no problem.
2. Attended transfers
Colin Anderson wrote:
The only thing that keeps me from ditching Zetafax is it has a slick
client that people like to use, unlike the barebones / kooky /
barely-working assortment of clients available for Hylafax (Yes, I
looked at Cypheus. It's crap. Yes I looked at WinHFC. It's good, but
Hello,
Does anyone know about open source wholesale billing for Asterisk?
Thanks
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Hey all. I pretty much need to satisfy my own curiosity on something.
Someone makes an incoming call to my Asterisk box, number never shows up
in any log whatsoever. I call the T1 provider, the number never shows up
on their end. According to the caller, every single instance shows
they're
Damon Estep wrote:
According to the wiki page
http://www.voip-info.org/wiki/view/Asterisk+IAXmodem
There are a couple of ways to integrate Asterisk and HylaFax with
IAXmodem;
* IAXmodem as HylaFax modem, both HylaFax and Asterisk on the same
machine
In my opinion this
Hi,
Does anybody have an easy solution for this.
I want something that will keep trying a busy number every 30 seconds
until it gets through.
I've tried retrydial, but can't get it to work.
Any suggestions?
Thanks,
-N
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Actually lately I have been poking around my Visual Studio .NET IDE and
ruminating about writing a client since the FTP protocol HylaFAX uses is
quite well documented; the other advantage with using VS instead of Python
or gtk or whatever is you can leverage the excellent Microsoft ODBC/DAO
Colin Anderson wrote:
The Zetafax client essentially uses port 139 and submits the .tif to a
Windows share so I don;t think it would be really possible to modify it,
interesting idea though.
I wasn't talking about modifying the client, but rather working *with*
the client.
So if the
I have a client with about 24 GXP-2000. Everything seems to be
working fine except one particular behavior of the blind transfer.
Whenever anyone makes an outbound call, they can transfer the call
between extensions either blind or attended with no problems.
However, whenever an incoming
Do you have audio running during the hold (MOH), or silence?
Could the Polycom (or asterisk) be dropping the call due to inactivity?
Yes is running...
I can listen to the music (MOH) and then suddenly I get disconnected.
-Original Message-
From: [EMAIL PROTECTED]
Why don't you just test for the dial status after the dial command
completes? I don't really see why you want something to keep dialing
until it gets through, but this would work.
[something]
1,1,Dial(zap/,sip/, etc/whatever, 10)
1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
1,n(LINEBUSY),
I'm trying to get inbound DIDs working via SIP.
I have 20 DIDs coming in via a single SIP profile in sip.conf.
I was hoping to have these matched in extensions.conf, so I have setup
lines like this:
exten=949271,1, Goto(mainmenu,s,1)
Unfortunately these aren't getting matched and I'm
Try changing it to:
exten = _949271,1,Goto(whatever)
another way to troubleshoot and figure out what you are getting for
the DID would be:
exten = _X.,1,Noop(Exten is: ${EXTEN})
watch the CLI to see what DID is coming in.
On 8/11/06, Mr. Jones [EMAIL PROTECTED] wrote:
I'm trying to get
Perhaps the context in sip.conf doesn't match the context in the dial plan.
From: [EMAIL PROTECTED] on behalf of Mr. Jones
Sent: Fri 8/11/2006 2:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
I'm trying
The reason he might want it is because it's a feature offered by many POTS
and Mobile Telcos. I know that's why I've played with it, the ITSPs and SIP
Termination providers I consult for want to have as many if not more
features to offer than the POTS and Mobile guys.
Cheers,
Rushowr - Sherwood
I have seen this before, it a routing issue on CustomerAs providers side.
Here is my story:
T1 from Focal with 400 DIDs, every single phone in the world can reach
it, but Verzion wireless customers. After 2 weeks of back and forth
with Verizon wireless, they bumped the ticket number to higher
If I am following you right, for extension matching you need to have a
_ in front of the number.
So your example should be like this:
exten = _949927,1,Goto(mainmenu,s,1)
Also I don't know if you did this on purpose or not but N will only
match for numbers 2-9, if you want 0-9 you will
I'm trying asterisk 1.2.9.1 with rtp jitterbuffer patch from
http://asterisk-backports.org
and seems, that this working only for sip-sip calls (probably also for
sip-zap),
I have jb enabled and forced in sip.conf, I can see debug log messages
from jitterbuffer, but only for sip-sip calls, not
Thanks -
Just to be clear - I just replaced the real digits with - I want
to direct these to specific extensions. So maybe I should have used
or something else?
I tried this:
exten=_9495551212,1, Goto(mainmenu,s,1)
But still to no avail.
On 8/11/06, Vadim Berezniker [EMAIL
Kevin,
Thanks for the suggestion. I can't seem to get it to work.
This is what I put in my extensions.conf
We only have one number that we want to keep trying right now, so I
tried to set it so by calling extension 777, it would start the
system retrying. (The actual number isn't
I double checked the context.
But the Looking for s is a bit confusing - not sure what s is?
On 8/11/06, Vadim Berezniker [EMAIL PROTECTED] wrote:
Perhaps the context in sip.conf doesn't match the context in the dial plan.
From: [EMAIL PROTECTED] on behalf of
Interesting. I guess unchecked (which my sample had no error checking)
it would lead me to think it would just use up resources. But I suppose
with the correct implementation I could see a use for it.
Kevin
Rushowr wrote:
The reason he might want it is because it's a feature offered by many
This seems to work just great, my configuration changes are maintained even
after rebooting this live CD, really cool ! I was able to get my softphone
working in just a couple of minutes.
However, being a complete asterisk beginner, does anyone know how I can
configure my net.PBX for use with
Also many so-called legacy hybrid PBX switches have had this for many
a year
Hard to compete when well used features that have been around for 20
years are lacking
John Novack
Rushowr wrote:
The reason he might want it is because it's a feature offered by many POTS and
Mobile Telcos. I know
Thanks Kevin -
I realized afterwards that the was a bad example.
It should be a specific number, I was just masking it.
I have 20 DIDs, some I want to send to a menu, most directly to an extension.
I've tried:
exten=9492711234,1, Macro(druiexten,3711,SIP/3711)
and:
exten=_9492711234,1,
You might want to take a look at 'sip debug' to see what your provider
is actually sending you. Its likely they aren't sending you the
9495551212 sting as you are expecting.
Thanks -
Just to be clear - I just replaced the real digits with - I want
to direct these to specific
s, means that it got an incoming call, but no exten came with it.
On 8/11/06, Mr. Jones [EMAIL PROTECTED] wrote:
I double checked the context.
But the Looking for s is a bit confusing - not sure what s is?
On 8/11/06, Vadim Berezniker [EMAIL PROTECTED] wrote:
Perhaps the context in sip.conf
I didn't understand :(
On Tue, 2006-08-08 at 17:31 -0400, C F wrote:
Yes, if you see the blindxfer veriable has something in it, send it
thru the same exten that the channel contained in blindxfer would go
thru.
On 8/8/06, Juan Pablo Abuyeres [EMAIL PROTECTED] wrote:
When a call is
You need to change:
exten = 777,4,goto(trunkretry,1,1)
to
exten = 777,4,goto(trunkretry,777,1)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Silverman
Sent: Friday, August 11, 2006 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial
Stephen,
FWIW I have not used the SPA-3000 but we just took delivery of an
A200d yesterday and installed it in our asterisk server. Had it up
and running in a couple of hours. We were previously using a PRI
with a Lucent TNT gateway that has hardware echo suppression built in.
We are
Mr. Jones wrote:
I have 20 DIDs, some I want to send to a menu, most directly to an
extension.
sip debug is (really) your friend. It should give you the [context]
where your DID is being send to and the 404 not found error also.
A particular line to look for: Looking for
Yeah...
I tried the NoOp function someone gave me above and I'll I'm getting is s
I'll go back to the provider
On 8/11/06, C F [EMAIL PROTECTED] wrote:
s, means that it got an incoming call, but no exten came with it.
On 8/11/06, Mr. Jones [EMAIL PROTECTED] wrote:
I double checked the
OK, just let me know what other language you want it in, I'm fluent in
more than english, just let me know what language. For the rest use
your creativity.
On 8/11/06, Juan Pablo Abuyeres [EMAIL PROTECTED] wrote:
I didn't understand :(
On Tue, 2006-08-08 at 17:31 -0400, C F wrote:
Yes, if
Actually it looks like I am getting the number but its coming through weird:
This is what sip debug gives me:
Looking for s in test-context (domain 9495551212)
So clearly I am getting the number, just not sure if its formated ok?
On 8/11/06, Mr. Jones [EMAIL PROTECTED] wrote:
Yeah...
I
Yep, my point exactly.
Since I'm in the middle of another ITSP project, I'll be hitting this again,
and will share anything I come up with. I've had thoughts, but never tested
it.
SHerwood
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
I have that
Looking for s in test-context (domain 9495551212)
-- Executing NoOp(SIP/5060-b7a1aa50, Exten is: s) in new stack
== Auto fallthrough, channel 'SIP/5060-b7a1aa50' status is 'UNKNOWN'
SIP/2.0 603 Declined
I'm just not sure how to use the domain BLAH to match an extension.
Uh, what's your Register statement for those SIP DIDs look like? If you
don't specify the number after a /, you'll be handed calls for that line,
but specifying 's' as the extension.
register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
I consider that last argument required anymore
Title: RE: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
Hmmm...
Appears as if the SIP invite request is ill-formed. Can you send the SIP debug
of the session to the list, so we may examine it?
Nir S
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
Ok - now maybe we're getting somewhere.
I didn't know I had to register them?
This is inbound only and the provider doesn't require that - so do I
just makeup a username?
I currently have the provider as a SIP peer.
On 8/11/06, Rushowr [EMAIL PROTECTED] wrote:
Uh, what's your Register
Here you go:
-- SIP read from 1.2.3.4:5060:
INVITE sip:3125551212;[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK08mh9j107ol1tace2500.1
From: sip:[EMAIL
PROTECTED]:5060;user=phone;testplex=TESTPLEX-h7l88bktjdhsf;tag=1000-0-1617457931
To: sip:[EMAIL
Ok, let's give it a try in Spanish please :)
On Fri, 2006-08-11 at 16:08 -0400, C F wrote:
OK, just let me know what other language you want it in, I'm fluent in
more than english, just let me know what language. For the rest use
your creativity.
On 8/11/06, Juan Pablo Abuyeres [EMAIL
I have asterisk set up for testing right now. I want
to roll it out in production to utilize it in several offices. I have a
few questions so this email might seem all over the place.
The topology is pretty simple. There are a total of
three offices. The main office has the asterisk
although a little example would be of great help here..
On Fri, 2006-08-11 at 17:08 -0400, Juan Pablo Abuyeres wrote:
Ok, let's give it a try in Spanish please :)
On Fri, 2006-08-11 at 16:08 -0400, C F wrote:
OK, just let me know what other language you want it in, I'm fluent in
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