date:20140416

Re: [asterisk-users] Old Asterisk Release wanting to upgrade ...

2014-04-16 Thread Lee, John (Sydney)

Thanks Johan.
I think I will stick with 1.4.x and DAHDI.  Although it is a unsupported 
release, I never had any problems with them.
Some machines have never been rebooted for 5+ years.
I am a bit scared of going to 11.  I have written a lot of AEL2 script in 
Asterisk 1.4.x and I am not sure if it will still run in 11.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johan Wilfer
Sent: Tuesday, 15 April 2014 7:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Old Asterisk Release wanting to upgrade ...

2014-04-15 10:37, Lee, John (Sydney) skrev:
> Hello,
> I have been running Asterisk for the past 5+ years on RedHat and I never 
> upgraded it before.
> All my Asterisk software is of the following release:
> 1) Asterisk 1.4.21.2
> 2) Libpri-1.4.4
> 3) Zaptel-1.4.11
> I would like to move the OS to CentOS and then I thought I can at the same 
> time ponder about upgrading Asterisk releases.
> However, I am bewildered by the myriad of different releases like 1.6,
> 1.8, 10.x, 11.x, 12.x, 13.x Can someone please give me some advice as to what 
> release I should upgrade?
> Or should I just stick to 1.4.x and just upgrade DAHDI?
> Thanks.
> Regards,
> John Lee
> The contents of this e-mail are intended for the named addressee only. It 
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> and then destroy it.
>
>

1.4, and 1.6-series have no support anymore. 1.8 is an LTS and have support 
currently, but this is also true for 11 and asterisk 11 will be supported 
longer.

You have the full list here:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

I would go for Asterisk 11 in your case. You will have to think of it more like 
a migration than an upgrade thought, as a lot has happened since asterisk 1.4.

On a side-note, I still run some old installations with a current Dahdi
+ Asterisk 1.4.44 and they work great together. There is the non-support
catch however.

Good luck!

--
Johan Wilfer


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Re: [asterisk-users] how to configure callcentric peer: no fqdn address matching?

2014-04-16 Thread Richard Reina



Enviado desde mi iPad

El abr 16, 2014, a las 6:45 p.m., Sean Darcy  escribió:

> On 04/15/2014 06:52 PM, Kai-Uwe Jensen wrote:
>> Oops, had it wrong. Here's how it works for me:
>> 
>> [callcentric-template](!)
>> type=friend
>> context=from-callcentric
>> fromdomain=callcentric.com 
>> defaultuser=1777xxx
>> fromuser=1777xxx
>> secret=password
>> insecure=port,invite
>> dtmfmode=rfc2833
>> disallowed_methods=UPDATE
>> session-timers=refuse
>> videosupport=no
>> qualify=no
>> disallow=all
>> allow=ulaw
>> 
>> [alpha11](callcentric-template)
>> host=alpha11.callcentric.com 
>> 
>> [alpha12](callcentric-template)
>> host=alpha12.callcentric.com 
>> 
>> [...]
>> 
>> 
>> 
>> On Tue, Apr 15, 2014 at 4:29 PM, Kai-Uwe Jensen > > wrote:
>> 
>>So do I need 7 contexts, one for each ip address?
>> 
>>sean
>> 
>> 
>>Yes, if what you call a "context" is your peer definition in
>>sip.conf. CC routes calls through a varying number of SBCs, all
>>(also) resolving to callcentric.com , but
>>each having their own name, typically "alphaxy.callcenctric.com
>>".
>> 
>>I think you can use asterisk's configuration template syntax to
>>create the required peer definitions. This would likely look similar
>>to this (from memory):
>> 
>>[alpha11]
>>host=alpha11.callcentric.com 
>> 
>>[alpha12]
>>host=alpha12.callcentric.com 
>> 
>>[callcentric](alpha11,alpha12)
>>type=peer
>>context=from-callcentric
>>defaultuser=1777
>>secret=
>>fromuser=1777
>>fromdomain=callcentric.com 
>> 
>>insecure=port,invite
>>disallowed_methods=UPDATE
>>directmedia=no
>>videosupport=no
>>disallow=all
>>allow=ulaw
>> 
>>Add templates for all IPs that callcentric.com
>> returns. Note that this approach isn't
>>foolproof: if/when CC change their pool of SBCs, you may have to add
>>more hosts, or remove them from this config. And, as you say, I
>>believe the cause is that asterisk only uses the first returned IP
>>for a host name. (Interestingly, the DNS server authoritative for CC
>>also varies the order of IPs it returns. Guess that's their way
>>load-balancing.)
>> 
>>And again, the above is from memory. I can look it up later today
>>and will follow up if I goofed/misremembered.
> Many thanks, That worked, I set up 20 contexts for 
> alpha[1-20].callcentric.com.
> 
> sean
> 
> 
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[asterisk-users] DAHDI: Multilink PPP over T1

2014-04-16 Thread Rusty Dekema

Greetings,

I want to bond two T1 lines running between a Linux machine with a
quad port tor2 card and a router running Cisco IOS. It seems that the
way to do this is to run multilink PPP across both T1 spans, but the
nethdlc/"Linux Generic HDLC" stack does not appear to support
multilink PPP.

Is there a way that I can get DAHDI to make the two T1 spans in
question show up as serial/tty type devices (or some such) so that I
can run regular Linux pppd on both of them? I see that there is a
plugin for pppd in dahdi-tools, but I don't know how to configure the
spans in system.conf, nor what device to point pppd to, in order to do
this.

Sincerely,
Rusty Dekema

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Re: [asterisk-users] DAHDI loading issue on Asterisk

2014-04-16 Thread Sean Darcy


On 04/16/2014 05:42 PM, Josh Metzger wrote:

Try starting Asterisk with the -f option.  It will NOT fork into the
background so you will see all messages on startup (including any that
might not end up in the log file).  Search for DAHDI errors which will
likely be there.

Also, if you configure everything and start DAHDI but don't start
Asterisk and run "dahdi_tool", is it showing you the circuits in an "OK"
state?

Josh


On Wed, Apr 16, 2014 at 5:25 PM, mailto:st...@vanwambeck.net>> wrote:

Hi all,
I have a fresh install of Asterisk 11.8.1 and am putting a Digium
TE435 4 T1 card in it for ISDN PRI. I can get the card to be
recognised by the DAHDI utilities but when I put in the file
"chan_dahdi.conf" with either the generated file from samples with
what seem to be appropriate settings or with the basic config as
outlined on the DAHDI install guide the Asterisk "core show help"
display is missing all the "dahdi" and "pri" commands.

If I remove the "chan_dahdi.conf" file and restart Asterisk the
commands magically reappear. I have gone back and checked on
menuselect but don't see anything obvious that I have missed to
support this function.

I have run out of ideas on how to integrate this. The documentation
makes it sound pretty simple but I have been fighting this for a
week now with no success.__ __
I am not seeing any parse errors from the module reload command:

asteriskpbx*CLI> module reload chan_dahdi.so 
asteriskpbx*CLI>

The truncated output from "core show help" is:

core stop when convenient Shut down Asterisk at empty call volume
core waitfullybooted Wait for Asterisk to be fully booted
data get Data API get
data show providers Show data providers
... 
resencestate change Change a custom presence state
presencestate list List currently know custom presence states
realtime destroy Delete a row from a RealTime database
realtime load Used to print out RealTime variables.

I can restart the asteriskpbx process without the "chan_dahdi.conf"
file and all the dahdi and pri commands are present. The
"chan_dahdi.conf" file I am loading is a basic file from the DAHDI
instructions. Even the sample file will not correctly load up either.

asteriskpbx@asteriskpbx:/etc/asterisk$ cat chan_dahdi.conf
[trunkgroups]

[channels]
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
echocancel = yes
echocancelwhenbridged = yes
relaxdtmf = yes
rxgain = 0.0
txgain = 0.0
group = 1
callgroup = 1
pickupgroup = 1
immediate = no
switchtype = 5ess
signalling = pri_cpe
context = incoming
echocancel = yes
channel = 1-23

Any suggestions on what I am missing would be greatly appreciated.
Steve VanWambeck



Do you have the kernel module loaded?

lsmod | grep dahdi

sean


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Re: [asterisk-users] how to configure callcentric peer: no fqdn address matching?

2014-04-16 Thread Sean Darcy


On 04/15/2014 06:52 PM, Kai-Uwe Jensen wrote:

Oops, had it wrong. Here's how it works for me:

[callcentric-template](!)
type=friend
context=from-callcentric
fromdomain=callcentric.com 
defaultuser=1777xxx
fromuser=1777xxx
secret=password
insecure=port,invite
dtmfmode=rfc2833
disallowed_methods=UPDATE
session-timers=refuse
videosupport=no
qualify=no
disallow=all
allow=ulaw

[alpha11](callcentric-template)
host=alpha11.callcentric.com 

[alpha12](callcentric-template)
host=alpha12.callcentric.com 

[...]



On Tue, Apr 15, 2014 at 4:29 PM, Kai-Uwe Jensen mailto:kujen...@gmail.com>> wrote:

So do I need 7 contexts, one for each ip address?

sean


Yes, if what you call a "context" is your peer definition in
sip.conf. CC routes calls through a varying number of SBCs, all
(also) resolving to callcentric.com , but
each having their own name, typically "alphaxy.callcenctric.com
".

I think you can use asterisk's configuration template syntax to
create the required peer definitions. This would likely look similar
to this (from memory):

[alpha11]
host=alpha11.callcentric.com 

[alpha12]
host=alpha12.callcentric.com 

[callcentric](alpha11,alpha12)
type=peer
context=from-callcentric
defaultuser=1777
secret=
fromuser=1777
fromdomain=callcentric.com 

insecure=port,invite
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

Add templates for all IPs that callcentric.com
 returns. Note that this approach isn't
foolproof: if/when CC change their pool of SBCs, you may have to add
more hosts, or remove them from this config. And, as you say, I
believe the cause is that asterisk only uses the first returned IP
for a host name. (Interestingly, the DNS server authoritative for CC
also varies the order of IPs it returns. Guess that's their way
load-balancing.)

And again, the above is from memory. I can look it up later today
and will follow up if I goofed/misremembered.


Many thanks, That worked, I set up 20 contexts for 
alpha[1-20].callcentric.com.


sean


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Re: [asterisk-users] Connecting 2 asterisks, one with PJSIP and other SIP returning 401

2014-04-16 Thread Gervasio Marchand Cassataro

Just a heads up... Enabled NOTICEs on the server and I see this every 10
seconds or so

[Apr 16 18:58:28] NOTICE[2138]: res_pjsip/pjsip_distributor.c:246
log_unidentified_request: Request from '"asterisk"
' failed for '179.25.158.95:5060' (callid:
477ca2fd0db3a5542dcf2afd50673b89@179.25.158.95:5060) - No matching
endpoint found

Thanks in advance for any help / ideas / clues or something! I'm scratching
my head around this and at this point


On Wed, Apr 16, 2014 at 6:26 PM, Gervasio Marchand Cassataro wrote:

> It's my first post here, so I'll cut to the chase
>
> I have 2 Asterisk servers and want to connect them using sip on one and
> pjsip on the other one. One is running at home and another at a VPS. The
> first one will be the client (with dynamic ip) and the 2nd the server.
>
> The client uses sip and the server pjsip.
>
> This is the client's sip.conf
>
> [general]
> context = default
> allowguest = no
> realm = myrealm.com
> udpbindaddr = 0.0.0.0
> qualify = yes
> subscribecontext = default
> localnet=192.168.1.0/255.255.255.0
> externhost=myhost.com 
> externrefresh=30
> dtmfmode = auto
> canreinvite = no
> jbenable = no
> sendrpid = yes
> trustrpid = no
> disallow=all
> allow=ulaw
> allow=alaw
> register => myuser:mypass@vpsserver
>
> [vpsserver]
> type=friend
> secret=myuser
> defaultuser=mypass
> host=vpsserver.domain.com
> context=inbound
> canreinvite=no
> insecure=port,invite
>
> And this is the server's pjsip.conf
>
> [transport-udp]
> type=transport
> protocol=udp
> bind=0.0.0.0
>
> [home]
> type=endpoint
> context=trusted
> disallow=all
> allow=ulaw
> allow=alaw
> transport=transport-udp
> auth=home
> aors=home
>
> [home]
> type=auth
> auth_type=userpass
> password=mypass
> username=myuser
>
> [home]
> type=aor
> max_contacts=10
>
> When I check on the client, executing sip show registry I get
>
> Hostdnsmgr Username   Refresh State   
>  Reg.Time
> vpsserver:5060  N  myuser   104 
> Registered   Tue, 15 Apr 2014 22:57:34
>
> which I guess means everything is ok... on the client side, I have on my
> extensions.conf
>
> exten => 66,1,Dial(SIP/1@vpsserver)
>
> and on the server's extensions.conf (in the trusted context) I have
>
> exten => 1,1,Playback(hello-world)
>
> So far so good... but when I dial 66 on my client Asterisk, I see the
> following SIP dialog on the server... the only weird thing is that I see
> some From: 192.168.1.112 (that's my home Asterisk's internal IP... the
> externhost works fine for all the providers I'm using, so I doubt that's an
> issue)
>
> http://pastebin.com/hkFezB8j
>
> Thanks in advance!
>
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Re: [asterisk-users] DAHDI loading issue on Asterisk

2014-04-16 Thread steve

(resend in plain text)...

Josh,

Yes, I only have one span currently connected, the other 3 are looped.
With the Asterisk process stopped I do see the OK on the "dahdi_tool"
screen.

I am not seeing any sort of errors in the /var/log/asterisk directory
but when I start asterisk manually with the -f option I do get the
following:

Unable to open '/dev/dahdi/channel': Permission denied
Unable to open channel 1: Permission denied
here = 0, tmp->channel = 1, channel = 1
Unable to register channel '1-23'
poll() failed: Interrupted system call

Looking at the /dev/dahdi directory I see the following:
snip
lrwxrwxrwx  1 root root   12 Apr 15 11:30 95 -> chan/004/023
lrwxrwxrwx  1 root root   12 Apr 15 11:30 96 -> chan/004/024
drwxr-xr-x  6 root root  120 Apr 15 11:30 chan
crw-rw  1 root root 196, 254 Apr 15 11:30 channel
crw-rw  1 root root 196,   0 Apr 15 11:30 ctl
drwxr-xr-x  2 root root   80 Apr 15 11:30 devices
crw-rw  1 root root 196, 255 Apr 15 11:30 pseudo
crw-rw  1 root root 196, 253 Apr 15 11:30 timer
root@asteriskpbx:

I compiled the dahdi package under "sudo su", perhaps that is what is
wrong???

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Re: [asterisk-users] DAHDI loading issue on Asterisk

2014-04-16 Thread steve

Josh,
Yes, I only have one span currently connected, the other 3 are looped. With the Asterisk process stopped I do see the OK on the "dahdi_tool" screen.
 
I am not seeing any sort of errors in the /var/log/asterisk directory but when I start asterisk manually with the -f option I do get the following:
 
Unable to open '/dev/dahdi/channel': Permission deniedUnable to open channel 1: Permission deniedhere = 0, tmp->channel = 1, channel = 1Unable to register channel '1-23'
Looking at the /dev/dahdi directory I see the following:
 snip
lrwxrwxrwx  1 root root   12 Apr 15 11:30 95 -> chan/004/023lrwxrwxrwx  1 root root   12 Apr 15 11:30 96 -> chan/004/024drwxr-xr-x  6 root root  120 Apr 15 11:30 chancrw-rw  1 root root 196, 254 Apr 15 11:30 channelcrw-rw  1 root root 196,   0 Apr 15 11:30 ctldrwxr-xr-x  2 root root   80 Apr 15 11:30 devicescrw-rw  1 root root 196, 255 Apr 15 11:30 pseudocrw-rw  1 root root 196, 253 Apr 15 11:30 timerroot@asteriskpbx:/dev/dahdi# cd
I compiled the dahdi package under "sudo su", perhaps that is what is wrong???
 
Steve

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Re: [asterisk-users] DAHDI loading issue on Asterisk

2014-04-16 Thread Josh Metzger

Try starting Asterisk with the -f option.  It will NOT fork into the
background so you will see all messages on startup (including any that
might not end up in the log file).  Search for DAHDI errors which will
likely be there.

Also, if you configure everything and start DAHDI but don't start Asterisk
and run "dahdi_tool", is it showing you the circuits in an "OK" state?

Josh


On Wed, Apr 16, 2014 at 5:25 PM,  wrote:

> Hi all,
> I have a fresh install of Asterisk 11.8.1 and am putting a Digium TE435 4
> T1 card in it for ISDN PRI. I can get the card to be recognised by the
> DAHDI utilities but when I put in the file "chan_dahdi.conf" with either
> the generated file from samples with what seem to be appropriate settings
> or with the basic config as outlined on the DAHDI install guide the
> Asterisk "core show help" display is missing all the "dahdi" and "pri"
> commands.
>
> If I remove the "chan_dahdi.conf" file and restart Asterisk the commands
> magically reappear. I have gone back and checked on menuselect but don't
> see anything obvious that I have missed to support this function.
>
> I have run out of ideas on how to integrate this. The documentation makes
> it sound pretty simple but I have been fighting this for a week now with no
> success.
> I am not seeing any parse errors from the module reload command:
>
> asteriskpbx*CLI> module reload chan_dahdi.so
> asteriskpbx*CLI>
>
> The truncated output from "core show help" is:
>
> core stop when convenient Shut down Asterisk at empty call volume
> core waitfullybooted Wait for Asterisk to be fully booted
> data get Data API get
> data show providers Show data providers
> ... 
> resencestate change Change a custom presence state
> presencestate list List currently know custom presence states
> realtime destroy Delete a row from a RealTime database
> realtime load Used to print out RealTime variables.
>
> I can restart the asteriskpbx process without the "chan_dahdi.conf" file
> and all the dahdi and pri commands are present. The "chan_dahdi.conf" file
> I am loading is a basic file from the DAHDI instructions. Even the sample
> file will not correctly load up either.
>
> asteriskpbx@asteriskpbx:/etc/asterisk$ cat chan_dahdi.conf
> [trunkgroups]
>
> [channels]
> usecallerid = yes
> hidecallerid = no
> callwaiting = yes
> usecallingpres = yes
> callwaitingcallerid = yes
> threewaycalling = yes
> transfer = yes
> canpark = yes
> cancallforward = yes
> callreturn = yes
> echocancel = yes
> echocancelwhenbridged = yes
> relaxdtmf = yes
> rxgain = 0.0
> txgain = 0.0
> group = 1
> callgroup = 1
> pickupgroup = 1
> immediate = no
> switchtype = 5ess
> signalling = pri_cpe
> context = incoming
> echocancel = yes
> channel = 1-23
>
> Any suggestions on what I am missing would be greatly appreciated.
> Steve VanWambeck
>
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[asterisk-users] Connecting 2 asterisks, one with PJSIP and other SIP returning 401

2014-04-16 Thread Gervasio Marchand Cassataro

It's my first post here, so I'll cut to the chase

I have 2 Asterisk servers and want to connect them using sip on one and
pjsip on the other one. One is running at home and another at a VPS. The
first one will be the client (with dynamic ip) and the 2nd the server.

The client uses sip and the server pjsip.

This is the client's sip.conf

[general]
context = default
allowguest = no
realm = myrealm.com
udpbindaddr = 0.0.0.0
qualify = yes
subscribecontext = default
localnet=192.168.1.0/255.255.255.0
externhost=myhost.com
externrefresh=30
dtmfmode = auto
canreinvite = no
jbenable = no
sendrpid = yes
trustrpid = no
disallow=all
allow=ulaw
allow=alaw
register => myuser:mypass@vpsserver

[vpsserver]
type=friend
secret=myuser
defaultuser=mypass
host=vpsserver.domain.com
context=inbound
canreinvite=no
insecure=port,invite

And this is the server's pjsip.conf

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[home]
type=endpoint
context=trusted
disallow=all
allow=ulaw
allow=alaw
transport=transport-udp
auth=home
aors=home

[home]
type=auth
auth_type=userpass
password=mypass
username=myuser

[home]
type=aor
max_contacts=10

When I check on the client, executing sip show registry I get

Hostdnsmgr Username   Refresh
StateReg.Time
vpsserver:5060  N  myuser   104
Registered   Tue, 15 Apr 2014 22:57:34

which I guess means everything is ok... on the client side, I have on my
extensions.conf

exten => 66,1,Dial(SIP/1@vpsserver)

and on the server's extensions.conf (in the trusted context) I have

exten => 1,1,Playback(hello-world)

So far so good... but when I dial 66 on my client Asterisk, I see the
following SIP dialog on the server... the only weird thing is that I see
some From: 192.168.1.112 (that's my home Asterisk's internal IP... the
externhost works fine for all the providers I'm using, so I doubt that's an
issue)

http://pastebin.com/hkFezB8j

Thanks in advance!
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[asterisk-users] DAHDI loading issue on Asterisk

2014-04-16 Thread steve

Hi all,
I have a fresh install of Asterisk 11.8.1 and am putting a Digium TE435 4 T1 card in it for ISDN PRI. I can get the card to be recognised by the DAHDI utilities but when I put in the file "chan_dahdi.conf" with either the generated file from samples with what seem to be appropriate settings or with the basic config as outlined on the DAHDI install guide the Asterisk "core show help" display is missing all the "dahdi" and "pri" commands.If I remove the "chan_dahdi.conf" file and restart Asterisk the commands magically reappear. I have gone back and checked on menuselect but don't see anything obvious that I have missed to support this function. I have run out of ideas on how to integrate this. The documentation makes it sound pretty simple but I have been fighting this for a week now with no success. 
I am not seeing any parse errors from the module reload command:asteriskpbx*CLI> module reload chan_dahdi.soasteriskpbx*CLI>The truncated output from "core show help" is:core stop when convenient Shut down Asterisk at empty call volumecore waitfullybooted Wait for Asterisk to be fully booteddata get Data API getdata show providers Show data providers... resencestate change Change a custom presence statepresencestate list List currently know custom presence statesrealtime destroy Delete a row from a RealTime databaserealtime load Used to print out RealTime variables.I can restart the asteriskpbx process without the "chan_dahdi.conf" file and all the dahdi and pri commands are present. The "chan_dahdi.conf" file I am loading is a basic file from the DAHDI instructions. Even the sample file will not correctly load up either.asteriskpbx@asteriskpbx:/etc/asterisk$ cat chan_dahdi.conf[trunkgroups][channels]usecallerid = yeshidecallerid = nocallwaiting = yesusecallingpres = yescallwaitingcallerid = yesthreewaycalling = yestransfer = yescanpark = yescancallforward = yescallreturn = yesechocancel = yesechocancelwhenbridged = yesrelaxdtmf = yesrxgain = 0.0txgain = 0.0group = 1callgroup = 1pickupgroup = 1immediate = noswitchtype = 5esssignalling = pri_cpecontext = incomingechocancel = yeschannel = 1-23
 
Any suggestions on what I am missing would be greatly appreciated.
Steve VanWambeck

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Re: [asterisk-users] WebRTC and JsSIP

2014-04-16 Thread Rusty Newton

On Wed, Apr 16, 2014 at 1:35 PM, Consultor VOIP  wrote:
> Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.
>
> I configure my Asterisk 11.7.0 to work wit WEBRTC.
>
> Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at
> the Asterisk, but when we try to make a call they send a 488 response and
> finish it.
>
> here is the part of the SIP DEBUG

We can't do much with part of your debug. You'll want to post a
pastebin link to your full SIP trace, and be sure that it includes at
least VERBOSE messages turned up to 5.[1]

Work on WebRTC support is on-going, so you'll want to test in the very
latest Asterisk version in your branch (11 or above). That means you
need to be on 11.9.0-rc2[2] at this moment.


[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
[2]: 
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.9.0-rc2.tar.gz


-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

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[asterisk-users] WebRTC and JsSIP

2014-04-16 Thread Consultor VOIP

Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.I configure my Asterisk 11.7.0 to work wit WEBRTC.Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish it.here is the part of the SIP DEBUG<--- SIP read from WS:177.64.122.237:49217 --->BYE sip:500@187.122.82.197:0;transport=ws SIP/2.0Via: SIP/2.0/WS e8ilhkrhlup2.invalid;branch=z9hG4bK7306188Max-Forwards: 69To: ;tag=as52a1a298From: "G" ;tag=ue84kn6rkuCall-ID: u5hkiispkvn9g841oedeCSeq: 9338 BYEReason: SIP ;cause=488; text="Not Acceptable Here"Supported: path, outbound, gruuUser-Agent: JsSIP 0.3.7Content-Length: 0Some one can help me with this problem?Thanks Gerald -- 
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Re: [asterisk-users] DUNDi with SIP Mapping

2014-04-16 Thread Kevin Larsen

> I wanted to move to DUNDi to simplify the setup. It looks like I 
> need to switch to IAX trunks to be able to do this.

You are a bit outside of what I have done, but this looks like it might be 
what you want to do with SIP:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP-- 
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Re: [asterisk-users] DUNDi with SIP Mapping

2014-04-16 Thread Ryan Wagoner

On Wed, Apr 16, 2014 at 9:06 AM, Kevin Larsen <
kevin.lar...@pioneerballoon.com> wrote:

>
> I am using DUNDi with SIP to do some least cost routing amongst my various
> locations. My mapping is close to what you have:
>
> priv => dundi-extens,0,SIP,/
>
> Where  is replaced with the actual name of my trunk as defined
> in sip.conf and  is the number they should dial on that
> trunk. I have not tried to define the SIP username/password in the DUNDi
> config itself, so I don't know if what you are trying to do is possible or
> not.
>

I was trying to avoid having to define the SIP trunks on all systems. I
currently have three FreePBX systems connected by SIP trunks with 800 DIDs.
Each system has SIP trunks defined to both other systems and routes
defining the extensions / DIDs. As I add more DID blocks and FreePBX
systems maintaining the trunks and routes is going to become cumbersome.

I wanted to move to DUNDi to simplify the setup. It looks like I need to
switch to IAX trunks to be able to do this.

Thanks,
Ryan
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Re: [asterisk-users] FW: clients unable to auth

2014-04-16 Thread Josh Metzger

I would modify the suggestions slightly and either add to it, or replace
the reference to voip-info with a link to https://wiki.asterisk.org

I was just thinking yesterday that voip-info seems out of date for many
things I search for, but at this point I'm usually looking up things that
are probably not used as frequently, so maybe the more "regular" stuff is
more up-to-date on voip-info.  In any case, the Asterisk Wiki seems quite
up-to-date and useful.  Also, you can quickly get some good info by doing
"core show application " or "core show function " from the
Asterisk console, like "core show application dial" gives you all the
possible arguments for "Dial", including some useful notes.  Quite handy.

Josh




On Wed, Apr 16, 2014 at 9:22 AM, Kevin Larsen <
kevin.lar...@pioneerballoon.com> wrote:

> > Thank you guys – your advice was spot on.  I will now reach out
> > earlier and not struggle with issues like this for 2 weeks J
>
> You sound like you are just getting started with Asterisk. A couple pieces
> of advice that helped me when I was starting out:
>
> 1. Get a copy of Asterisk: The Definitive Guide. Work through the examples
> and understand the concepts it teaches. I still use it all the time.
> 2. When you run into problems, http://voip-info.org is a great Asterisk
> resource. It isn't always perfectly up to date, but is very useful.
> 3. Search on the error messages you are given by Asterisk. It is common
> enough that many (but not all) of the error messages will have hits that
> explain your problem in greater detail.
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Re: [asterisk-users] FW: clients unable to auth

2014-04-16 Thread Kevin Larsen

> Thank you guys – your advice was spot on.  I will now reach out 
> earlier and not struggle with issues like this for 2 weeks J 

You sound like you are just getting started with Asterisk. A couple pieces 
of advice that helped me when I was starting out: 

1. Get a copy of Asterisk: The Definitive Guide. Work through the examples 
and understand the concepts it teaches. I still use it all the time.
2. When you run into problems, http://voip-info.org is a great Asterisk 
resource. It isn't always perfectly up to date, but is very useful.
3. Search on the error messages you are given by Asterisk. It is common 
enough that many (but not all) of the error messages will have hits that 
explain your problem in greater detail.
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Re: [asterisk-users] DUNDi with SIP Mapping

2014-04-16 Thread Kevin Larsen

> From the reading and testing I have done it doesn't look like SIP 
> supports a username and password in the Dial string. I currently 
> have the following mapping.
> 
> priv => dundi-extens,0,SIP,dundi:pass@1.1.1.1/$
> {NUMBER},nounsolicited,nocomunsolicit,nopartial

> On the sending side I see
> 
> NOTICE[31598] chan_sip.c: Conflicting extension values given. Using 
> 'dundi' and not '1001'

> On the receiving side it will not match the SIP dundi user and tries
> to call dundi instead of 1001.
> 
> -- Executing [dundi@from-sip-external:1] NoOp("SIP/1.1.1.
> 2-", "Received incoming SIP connection from unknown peer to 
> dundi") in new stack
> 

> Is there a way to configure DUNDi to use SIP or does it only work with 
IAX?

I am using DUNDi with SIP to do some least cost routing amongst my various 
locations. My mapping is close to what you have:

priv => dundi-extens,0,SIP,/

Where  is replaced with the actual name of my trunk as defined 
in sip.conf and  is the number they should dial on that 
trunk. I have not tried to define the SIP username/password in the DUNDi 
config itself, so I don't know if what you are trying to do is possible or 
not.-- 
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[asterisk-users] DUNDi with SIP Mapping

2014-04-16 Thread Ryan Wagoner

>From the reading and testing I have done it doesn't look like SIP supports
a username and password in the Dial string. I currently have the following
mapping.

priv => dundi-extens,0,SIP,
dundi:pass@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial

On the sending side I see

NOTICE[31598] chan_sip.c: Conflicting extension values given. Using 'dundi'
and not '1001'

On the receiving side it will not match the SIP dundi user and tries to
call dundi instead of 1001.

-- Executing [dundi@from-sip-external:1] NoOp("SIP/1.1.1.2-",
"Received incoming SIP connection from unknown peer to dundi") in new stack


Is there a way to configure DUNDi to use SIP or does it only work with IAX?

Thanks,
Ryan
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Re: [asterisk-users] FW: clients unable to auth

2014-04-16 Thread Peter Reid

Hi All,

 

Thank you guys - your advice was spot on.  I will now reach out earlier and
not struggle with issues like this for 2 weeks J 

 

Best Regards, 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: Wednesday, April 16, 2014 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FW: clients unable to auth

 

asterisk-users-boun...@lists.digium.com wrote on 04/16/2014 05:56:32 AM:

> From: "Peter Reid"  
> To: , 
> Date: 04/16/2014 05:56 AM 
> Subject: [asterisk-users] FW: clients unable to auth 
> Sent by: asterisk-users-boun...@lists.digium.com 
> 
> Hi Guys, 
>   
> Just new to Asterisk and am completely stumped.  I have created two 
> accounts as instructed.  Please see below for the config of the user
> accounts.   
>   
> [Peter] 
> type=friend 
> host=IP address 
> disallow=all 
> allow=ulaw 
> allow=alaw 
> callerid=Peter <6004> 
> secret=XXX 
> context=default 
> port=9060 
> nat=force_rport,comedia 
> deny=0.0.0.0 
> permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0 
>   

Your phone is registering with the name 6004 and not Peter. You either need
to change [Peter] to [6004] in Asterisk or update your phone config to make
it use Peter for your authorization name.



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Re: [asterisk-users] FW: clients unable to auth

2014-04-16 Thread Kevin Larsen

asterisk-users-boun...@lists.digium.com wrote on 04/16/2014 05:56:32 AM:

> From: "Peter Reid" 
> To: , 
> Date: 04/16/2014 05:56 AM
> Subject: [asterisk-users] FW: clients unable to auth
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Hi Guys,
> 
> Just new to Asterisk and am completely stumped.  I have created two 
> accounts as instructed.  Please see below for the config of the user
> accounts. 
> 
> [Peter]
> type=friend
> host=IP address
> disallow=all
> allow=ulaw
> allow=alaw
> callerid=Peter <6004>
> secret=XXX
> context=default
> port=9060
> nat=force_rport,comedia
> deny=0.0.0.0
> permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, 
PrvIP/255.255.0.0
> 

Your phone is registering with the name 6004 and not Peter. You either 
need to change [Peter] to [6004] in Asterisk or update your phone config 
to make it use Peter for your authorization name.-- 
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Re: [asterisk-users] FW: clients unable to auth

2014-04-16 Thread Joshua Colp


Peter Reid wrote:

Hi Guys,


Kia ora,


Just new to Asterisk and am completely stumped. I have created two
accounts as instructed. Please see below for the config of the user
accounts.

[Peter]

type=friend

host=IP address

disallow=all

allow=ulaw

allow=alaw

callerid=Peter <6004>

secret=XXX

context=default

port=9060

nat=force_rport,comedia

deny=0.0.0.0

permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0

When attempting to register there appears to be something not allowing
the authentication of the client against Asterisk. I am getting a 401
Unauthorized on first attempt and then 403 (Bad auth) on second. Is
there any ACL config that I have missed which is not allow a good
authentication. I have enabled nat and allowed all public and private
IP’s of the two clients with masks.

The CLI console returns the following:

ignore - Apr 16 11:09:57] NOTICE[24359]: chan_sip.c:27724
handle_request_register: Registration from
'"6004"' failed for 'IP:57836' - No matching
peer found

[Apr 16 11:09:57] NOTICE[24359]: chan_sip.c:27724
handle_request_register: Registration from '"6004"'
failed for 'IP:57836' - No matching peer found


Your client is configured to use "6004" as the username while your above 
configuration uses "Peter". Since the usernames differ, it fails.


Cheers,

--
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] FW: clients unable to auth

2014-04-16 Thread Asghar Mohammad

Hello,
Try this

[6004]

type=friend

host=dynamic

disallow=all

allow=ulaw

allow=alaw

callerid=6004 

secret=XXX

context=default

port=9060

nat=force_rport,comedia

deny=0.0.0.0

permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0





On Wed, Apr 16, 2014 at 12:56 PM, Peter Reid wrote:

> Hi Guys,
>
>
>
> Just new to Asterisk and am completely stumped.  I have created two
> accounts as instructed.  Please see below for the config of the user
> accounts.
>
>
>
> [Peter]
>
> type=friend
>
> host=IP address
>
> disallow=all
>
> allow=ulaw
>
> allow=alaw
>
> callerid=Peter <6004>
>
> secret=XXX
>
> context=default
>
> port=9060
>
> nat=force_rport,comedia
>
> deny=0.0.0.0
>
> permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0
>
>
>
> When attempting to register there appears to be something not allowing the
> authentication of the client against Asterisk.  I am getting a 401
> Unauthorized on first attempt and then 403 (Bad auth) on second.   Is there
> any ACL config that I have missed which is not allow a good
> authentication.  I have enabled nat and allowed all public and private IP’s
> of the two clients with masks.
>
>
>
> The CLI console returns the following:
>
>
>
> ignore - Apr 16 11:09:57] NOTICE[24359]: chan_sip.c:27724
> handle_request_register: Registration from '"6004"<
> sip:6...@xx.xx.xx.xx:9060>' failed for 'IP:57836' - No matching peer found
>
> [Apr 16 11:09:57] NOTICE[24359]: chan_sip.c:27724 handle_request_register:
> Registration from '"6004"' failed for 'IP:57836' - No
> matching peer found
>
> serverIP*CLI>
>
>
>
> Have searched under No matching peer found and although there is some info
> on this it does not appear to satisfy my situation.
>
>
>
> Please help.
>
>
>
> *Best Regards, *
>
>
>
> *Peter *
>
>
>
>
>
>
>
>
> --
>
>
> This email is free from viruses and malware because avast! 
> Antivirusprotection is active.
>
>
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[asterisk-users] FW: clients unable to auth

2014-04-16 Thread Peter Reid

Hi Guys,

 

Just new to Asterisk and am completely stumped.  I have created two accounts
as instructed.  Please see below for the config of the user accounts.  

 

[Peter]

type=friend

host=IP address

disallow=all

allow=ulaw

allow=alaw

callerid=Peter <6004>

secret=XXX

context=default

port=9060

nat=force_rport,comedia

deny=0.0.0.0

permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0

 

When attempting to register there appears to be something not allowing the
authentication of the client against Asterisk.  I am getting a 401
Unauthorized on first attempt and then 403 (Bad auth) on second.   Is there
any ACL config that I have missed which is not allow a good authentication.
I have enabled nat and allowed all public and private IP's of the two
clients with masks.  

 

The CLI console returns the following: 

 

ignore - Apr 16 11:09:57] NOTICE[24359]: chan_sip.c:27724
handle_request_register: Registration from
'"6004"' failed for 'IP:57836' - No matching peer
found

[Apr 16 11:09:57] NOTICE[24359]: chan_sip.c:27724 handle_request_register:
Registration from '"6004"' failed for 'IP:57836' - No
matching peer found

serverIP*CLI> 

 

Have searched under No matching peer found and although there is some info
on this it does not appear to satisfy my situation.  

 

Please help. 

 

Best Regards, 

 

Peter 

 

 

 



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Re: [asterisk-users] Asterisk and OSX

2014-04-16 Thread Pete Mundy

Hi Gents

I thought I'd pop up and add my 2cents when I just happened across this e-mail 
glancing through the list.

I actually do run a production service on Asterisk running on Mac OS X server. 
But of course I do it within a VM environment on Linux (Ie Asterisk on Ubuntu 
under VirtualBox on Mac OS X Server). This gives me all of the niceties of a 
Mac server, along with the ease of administration of Linux for the native 
Asterisk config.

The best part though is that we can easily run up 4 independent Asterisk VMs on 
the one Mac Mini server, then mount it in one of these cases along with another 
one configured the same for hot-standby and we have a fantastic low-power 
(<200W) small 1U, 300m-deep rack-mount solution. Stopping the VMs on one Mini 
and launching them on the other (without any change of IPs) can be done in 
<9seconds when booted from SSDs.

http://www.sonnettech.com/product/rackmacmini.html

Macs do have their place running Asterisk. Just not natively! :)

Pete Mundy
Technical Director
Fiberphone Limited
Nelson, New Zealand
www.fiberphone.co.nz

On 15/04/2014, at 10:40 PM, Thomas Rechberger  wrote:

> Am 14.04.2014 16:19, schrieb Eric Wieling:
>> So few people use Asteisk on OSX that I doubt anyone will answer.
> 
> Look how many answers he got, i got none
> 
> 
> 
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