I would be very interested in using sipsak for something like this. What
have you tried so far?
-Thufir
On Mon, 16 Jan 2017, Olivier wrote:
Thinking over my previous, I wonder if sipsak could be used to send
outgoing SIP NOTIFY messages.
Would both Asterisk and sipsak be able to share
On Wed, 11 Jan 2017, Doug Lytle wrote:
On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote:
Can I dial directly from the asterisk console with the Dial() application?
console dial number@context
Thanks, that's much more intuitive :)
-T
0UNKNOWN
demo_bob (Unspecified)D Yes
Yes0UNKNOWN
piter (Unspecified)D Yes
Yes0UNKNOWN
thufir(Unspecified)D Yes
Can I dial directly from the asterisk console with the Dial() application?
or, is channel originate preferred:
channel originate SIP/thufir extension 18003569377@outbound
thanks,
Thufir
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I appreciate that the console lets you see the details for a peer with
"sip show peer foo". Certainly, I can look in sip.conf to see the
[general] context, but can I output those settings, and only those
settings, to the console?
thank
ows as coming from a single number, the account number?
thanks,
Thufir
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ok, that's really all I need to know. Of course, if anyone else wants to
throw in their two cents, don't let me stop you :)
-Thufir
On Wed, Jul 6, 2016 at 1:36 AM, Frank Vanoni
wrote:
> I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with
> Ubuntu Server
never be sent unless a call, or other communication with
an endpoint, was being attempted?
thanks,
Thufir
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I'm debating between a cloud PBX or, perhaps, rasberry pi. For a SOHO,
maybe three hardphones, rasberry pi would suffice? I would be amazed, but,
if so, great.
thanks,
Thufir
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so, when I'm receiving an inbound call, the direction would be telnyx
first, then me. Regardless of whether the ?message? is from me or the
provider.
-Thufir
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hat am I looking for with
regards to receiving calls?
thanks,
Thufir
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h
g instance of Asterisk. I would have to handle the invite
through Asterisk and keep it running to make and receive calls? Presumably
this invite is interacting with Asterisk, or something similar, at
telnyx.com -- which seems overkill.
thanks,
Thufir
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_
the channel variable is
testcarrier, which has a context of "default".
Each channel variable maps to at most one context? Many channel
variables can map to a single context?
thanks,
Thufir
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200)
in local_200, that just seems suspect. Yes, dial out, but shouldn't it
be using BABY? I don't understand why it's using sub-string with the 1.
thanks,
Thufir
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you this:
H = Hash["a" => 100, "b" => 200]
The => is a mapping, or at least that's my understanding. What does it
mean in Asterisk? I didn't
fully appreciate that Asterisk is, apparently, its own language.
thanks,
Thufir
--
at
net.java.sip.communicator.service.protocol.AccountManager.loadStoredAccounts(AccountManager.java:446)
at
net.java.sip.communicator.service.protocol.AccountManager.runInLoadStoredAccountsThread(AccountManager.java:562)
at
net.java.sip.communicator.service.protocol.AccountMan
when it's
> encapsulated into MAC traffic).
so how does a client pc find the server if there's no NAT? by IP
address?? That makes no sense, to me, if the switch isn't assigning
addresses.
-Thufir
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_
-
oblem Solved"
This is the router/modem gateway the ISP supplied:
http://www.cisco.com/web/consumer/support/modem_DPC3825.html
When I connect one of these switches to the router, that doesn't create a
double-NAT problem?
thanks,
Thufir
--
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c.
turn on debug like so:
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
I'll try that, thanks :)
-Thufir
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New to A
31:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
linux-k7qk*CLI> exit
linux-k7qk:~ #
linux-k7qk:~ #
thanks,
Thufir
--
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On 15-03-20 6:42 AM, thufir wrote:
I wasn't able to get much out of babytel, beyond the fact that I was,
apparently, sending options which is why I'm not getting 200 OK.
How can I, generally speaking, ping/telnet or otherwise test the
connection to get more data?
A connection to
t:
Reg. Contact :
Qualify Freq : 6 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption :
in
Asterisk? Or, how do I establish that the parameters are incorrectly
entered? Because Jitsi is able to call out and in, I believe that
eliminates NAT, firewall or other networking issues.
thanks,
Thufir
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've run out of troubleshooting steps, that's the
one to use.
HTH,
Thufir
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p://superuser.com/questions/880705/
IAX might be useful in this circumstance :)
-Thufir
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On Thu, 12 Mar 2015 12:52:46 +, Thufir wrote:
> Heh, well, I guess it's dead:
>
> http://www.digium.com/en/products/software/skype-for-asterisk
is this current?
http://www.remsys.com/blog/skype-connect-to-asterisk
it doesn't solve, I think, the prob
iga, etc, without using Asterisk, I
get "too many hops" errors. While I have another computer on the LAN I
can connect to, it's not quite the same.
Any thoughts?
thanks,
Thufir
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personal preference.
>From a naive perspective, why SQL statements at all? Why not just
database config and data instead?
thanks,
Thufir
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New to Aster
ome) of the ports aren't gigabit?
Small office, about five agents.
thanks,
Thufir
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On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote:
> For a very basic setup it would work, but I would suggest POE at a
> minimum, and vlan support if possible.
thanks for the recomendations :)
-
On Sun, 22 Feb 2015 08:32:26 +0530, Mitul Limbani wrote:
> READ READ READ
I know, I have the 4th edition and I've been reading it. Personally, I
find it more general than specific, but I'll go back through that
chapter, absolutely.
th
I'm looking into the dialplan specifics:
tleilax:~ #
tleilax:~ # cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=DAHDI/r1; Trunk interface
TRUN
who's help me out. I'm sure I'll have other
problems, but huge milestone.
-Thufir
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nnectivity because of too many hops:
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:thu...@ekiga.net -m "hi"
No SRV record: _sip._tcp.ekiga.net
No SRV record: _sip._udp.ekiga.net
using A record: ekiga.net
Max-Forwards set to 0
message received:
SIP/2.0 483 Too Many Hops
Via:
all sorts of complex
network setups, yes, but not something basic like this.
thanks,
Thufir
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PTIONS)
[Feb 20 21:06:38] Really destroying SIP dialog '1876256264@127.0.1.1'
Method: OPTIONS
[Feb 20 21:06:51] Really destroying SIP dialog '705273564@127.0.1.1'
Method: OPTIONS
tleilax*CLI> exit
tleilax:~ #
I would've liked to
t; to "345" and get success (well, at this at least). This
probably has something to do with my dialplan..
Is the message, "hi", logged anywhere?
-Thufir
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r.com/product/fs116
Is this a reasonable choice? Would I be wrong in thinking that most any
Fast Ethernet switch would be fine for Asterisk?
thanks,
Thufir
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leilax*CLI>
on doge:
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:devries@tleilax -m "hi"
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
Max-Forwards set to 0
message received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
127.0.1.1:
lt;999>
ACL : No
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Sess-Min-SE : 90 secs
RTP Engine : asterisk
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
tleilax*CLI>
tleilax*CLI> exit
tleilax:~ #
tleilax:~ # exit
logout
Connection to
: uas
Sess-Expires : 1800 secs
Sess-Min-SE : 90 secs
RTP Engine : asterisk
Codec Order : (ulaw:20,gsm:20)
Auto-Framing: No
tleilax*CLI>
which would make the URI sip:thufir...@tleilax.bounceme.net ?
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:thufir101@tleilax
No S
spond with a 200 OK.
In general, this 200 OK status code can be used for troubleshooting? Is
there a log of status codes sent, or that's just done live through the
console?
thanks,
Thufir
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own)
Do I have a firewall problem which would impact a soft phone from
establishing a connection?
thufir@doge:~$
thufir@doge:~$
thufir@doge:~$ nmap 192.168.1.1
Starting Nmap 6.46 ( http://nmap.org ) at 2015-02-18 06:10 PST
Nmap scan report for 192.168.1.1
Host is up (0.0086s latency).
Not
le to an iogear wifi adaper.
the netgear router uses DHCP and gets an IP address of 192.x.x.x from the
iogear device.
The iogear device gets its IP address wirelessly from the another
router. That upstream router is from the ISP (has their branding), and
has a firewall.
So,
11 D N
5060 UNREACHABLE
gs102/gs102 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0
offline]
tleilax*CLI>
thanks,
Thufir
--
_
(Unspecified) D N 0UNKNOWN
babytel/1 198.38.7.11 D
N 5060 UNREACHABLE
gs102/gs102 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0
offline]
when running asterisk -r, is there a way to turn off the messages? I
didn't find the answer in the man page.
thanks,
Thufir
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New to Asterisk?
asy to implement.
I also saw a module or plugin which did this. Darn, can't find it now.
-Thufir
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http://www.voip-info.org/wiki/view/Asterisk+CRM+Integration
lists a few options. I'm looking for, literally, the simplest FOSS CRM
for "click to dial" functionality, but don't know where to start
;contact center" software
for asterisk, not asterisk itself. There's another variant (or perhaps
the same thing) at:
http://astercc.org/products/astercrm
thanks,
Thufir
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or the responses, I'm off to the races now.
-Thufir
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http://w
On Tue, 17 Jun 2014 12:14:05 +0200, Rainer Piper wrote:
> git clone https://github.com/asterisk/pjproject pjproject
At the very least, thank you for pjsip. I'm not sure what it is yet, but
seems intriguing :)
I'm on Ubunutu 14.04, but will look over your script and adapt i
at kind of phones are?
>
>
>
> On Tue, Jun 17, 2014 at 1:14 PM, Rainer Piper
> wrote:
>
>> Am 17.06.2014 09:04, schrieb thufir:
>>
>> I have the Asterisk book, it's enormous, the 4th edition as per
>> http://www.asteriskdocs.org/.
>>
>>
Pardon. My home PC is Ubuntu, 14.04.
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http://www.asterisk.org/hello
a
ask?
I'm looking for the simplest litmus test for functionality possible.
thanks,
Thufir
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How do you load the contact list? It's a database? Sqlite3?
https://wiki.asterisk.org/wiki/display/AST/SQLite3+astdb+back-end
I'm not clear on what this specific database does. If it's not this
specific database which has contact information, which database does?
Are the nslu2 folks describing hacking the
<http://www.yealink.com/english/prodetail_p1k.htm> phone, or using that
phone _with_ a slug? If I can run asterisk on my computer, and not hack
any hardware, that'd be preferable.
thanks,
Thufir
..]
Heh, I did miss it. Yes, for windows, it specifies X-Lite software. That
x-Lite isn't mentioned for Linux implies that it'll only work for windows.
Curious, but not unusual, state of affairs.
In any event, x-Lite doesn't support IAX, which I require.
-Thufir
__
It seems that xlite doesn't support IAX? Too bad.
While xlite does, apparently, run under linux it's not clear to me whether
or not the a-link device will work with the linux version of xlite.
-Thufir
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On Sun, 05 Nov 2006 09:53:52 +, Peter Bowyer wrote:
> It''s a USB Sound card / keypad / display, not a phone. It contols a
> softphone on the PC it's plugged into - they say it works with XLite -
> the SIP setup will be done in Xlite, not the 'phone'.
&
with gizmo project or free world dialup, or even Skype?
thanks,
Thufir
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der
linux, as it responds to HTTP POST requests. No driver necesarry.
They also make phones, or adapters with multiple jacks for phones, if you
need more than one jack.
-Thufir
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ssible to directly connect any
hardware to the router. No, it's not possible to use a switch.
thanks,
Thufir
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