date:20140418

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

2014-04-18 Thread Gervasio Marchand Cassataro

I'm not exactly nailing it in participation on my thread, so I swear this
is my last message if no one replies ;)

I went ahead and enabled full debugging, and got some interesting results
(available at http://pastebin.com/KiY6DMHi)

What I see is:

   1. The registration works fine
   2. When the client tries to establish a call to the server, the server
   looks into either the From or Contact headers... but the client sends the
   caller id in that place and that's when the INVITE gets rejected

I guess a proper question would be "Is there any way on the sip.conf to
specify the contact (I think that s@ip would work, as that's registered on
the client) or on the pjsip.conf to whitelist the ip of a registered
contact? something like insecure=invite"

Thanks!


On Wed, Apr 16, 2014 at 8:02 PM, Gervasio Marchand Cassataro wrote:

> 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] Connecting 2 asterisks, one with PJSIP and other SIP returning 401

2014-04-18 Thread Gervasio Marchand Cassataro

I know I promised it... but... I FIGURED IT OUT!!!

So, I'll just reply so that the next one with this issue can find this

The only bit of information was using fromuser instead of defaultuser...
that only thing did the trick... so

[vpsserver]
type=friend
host=vpsserver.domain.com
context=inbound
fromuser=myuser
secret=mypass

on the client's sip.conf did the trick


On Fri, Apr 18, 2014 at 9:24 AM, Gervasio Marchand Cassataro wrote:

> I'm not exactly nailing it in participation on my thread, so I swear this
> is my last message if no one replies ;)
>
> I went ahead and enabled full debugging, and got some interesting results
> (available at http://pastebin.com/KiY6DMHi)
>
> What I see is:
>
>1. The registration works fine
>2. When the client tries to establish a call to the server, the server
>looks into either the From or Contact headers... but the client sends the
>caller id in that place and that's when the INVITE gets rejected
>
> I guess a proper question would be "Is there any way on the sip.conf to
> specify the contact (I think that s@ip would work, as that's registered
> on the client) or on the pjsip.conf to whitelist the ip of a registered
> contact? something like insecure=invite"
>
> Thanks!
>
>
> On Wed, Apr 16, 2014 at 8:02 PM, Gervasio Marchand Cassataro 
> wrote:
>
>> 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] Connecting 2 asterisks, one with PJSIP and other SIP returning 401

2014-04-18 Thread Joshua Colp


Gervasio Marchand Cassataro wrote:

I know I promised it... but... I FIGURED IT OUT!!!

So, I'll just reply so that the next one with this issue can find this

The only bit of information was using fromuser instead of defaultuser...
that only thing did the trick... so

[vpsserver]
type=friend
host=vpsserver.domain.com 
context=inbound
fromuser=myuser
secret=mypass

on the client's sip.conf did the trick


I don't know why I didn't see this earlier but yes, the currently 
available endpoint identifiers do not include a mechanism to match based 
on the information a device may have registered with to an AOR. As 
you've figured out using fromuser does allow the user endpoint 
identifier to find the endpoint and then it works happily.


Cheers,

--
Joshua Colp
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] Dimensioning

2014-04-18 Thread Joshua Colp


Steve Edwards wrote:

On Thu, 17 Apr 2014, Jerry Geis wrote:


I was thinking transcoding was through PRI card - not gsm to ulaw. :)


You can convert the GSM files to ULAW using sox. I tend to transcode
everything to WAV (PCM not that funky 'GSM in WAV') because it is
relatively cheap (CPU cycles) to transcode from WAV to ULAW and
everything else in the world understands WAV just fine. If you really
need to squeeze out every last cycle, you can schedule a script to
transcode WAVs to ULAWs as needed.


So if all I am doing is originating calls, and using playback() in the
dialplan - then a system() call on completion I can expect upwards or
3000 concurrent calls?


Based on my unsubstantiated testing on my hosts, that seems like a
reasonable conclusion.


I'll just chime in slightly here. At that number of calls you've got 
many factors going on - and they aren't just Asterisk. If playing back 
files you've got disk I/O going on which can slow stuff down (unless the 
system caches it enough or you throw them into a ramdisk yourself). You 
are also sending and receiving a *ton* of small packets. This can make 
network equipment and NICs unhappy.


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

2014-04-18 Thread Joshua Colp


Josh Metzger wrote:

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.


To provide some clarification of why it is up to date:

The documentation for many things (applications, dialplan functions, 
some configuration files, manager events, etc) exists as XML within the 
Asterisk source code itself near the implementation. There are scripts 
which take this XML and produce some of the wiki pages you see. We've 
also made it so that documentation for some things has to be specified 
when adding stuff or else Asterisk will not allow it to work. This makes 
documentation as close to a first class citizen as it can be (albeit 
that doesn't mean one can't write bad documentation).


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] DAHDI loading issue on Asterisk

2014-04-18 Thread steve

Hello all,
At this point I am at a complete loss as to how to fix my problem. I
have gone back and rebuilt the whole system from the ground up but am
getting the same results. When I start asterisk with the -f option I do
see what the problem appears to be.

Unable to load config skinny.conf, Skinny disabled.
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'

This is because the file permissions are not set correctly on the
/dev/dahdi/channel:

crw-rw  1 root root 196, 254 Apr 17 15:07 channel

Can anyone provide me with a concise step by step guide, or point me to
one, on building Asterisk 11, DADHI and all? I have pieced together
different components of the process but they seem to be in error. Other
than this one problem the Asterisk system works like a champ. I really
need to get my wtce43x card working and ISDN PRI setup on this platform.

Thanks for your assistance.
Steve

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

2014-04-18 Thread Russ Meyerriecks

On Fri, Apr 18, 2014 at 12:22 PM,  wrote:
>
> This is because the file permissions are not set correctly on the
> /dev/dahdi/channel:
>

Steve,
  The default dahdi.rules file specifies the /dev/dahdi/ directory to be
owned by asterisk:asterisk. What's the contents of your
/etc/udev/rules.d/dahdi.rules file?

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] DAHDI loading issue on Asterisk

2014-04-18 Thread steve

Russ,
Here is what I am showing:

asteriskpbx@asterisk-pbx:/etc/udev/rules.d$ cat dahdi.rules

ACTION!="add",  GOTO="dahdi_add_end"

# DAHDI devices with ownership/permissions for running as non-root
SUBSYSTEM=="dahdi", OWNER="asterisk", GROUP="asterisk",
MODE="0660"

# Backward compat names: /dev/dahdi/
SUBSYSTEM=="dahdi_channels",SYMLINK+="dahdi/%m"

# Add persistant names as well
SUBSYSTEM=="dahdi_channels", ATTRS{hardware_id}!="",   
SYMLINK+="dahdi/devices/%s{hardware_id}/%s{local_spanno}/%n"
SUBSYSTEM=="dahdi_channels", ATTRS{location}!="",  
SYMLINK+="dahdi/devices/@%s{location}/%s{local_spanno}/%n"

LABEL="dahdi_add_end"

# hotplug scripts
SUBSYSTEM=="dahdi_devices",
RUN+="%E{DAHDI_TOOLS_ROOTDIR}/usr/share/dahdi/dahdi_handle_device"
SUBSYSTEM=="dahdi_spans",  
RUN+="%E{DAHDI_TOOLS_ROOTDIR}/usr/share/dahdi/dahdi_span_config"
 
 
 Original Message 
Subject: Re: [asterisk-users] DAHDI loading issue on Asterisk
From: Russ Meyerriecks 
Date: Fri, April 18, 2014 12:09 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion



On Fri, Apr 18, 2014 at 12:22 PM,  wrote:This is
because the file permissions are not set correctly on the
/dev/dahdi/channel:


Steve,
  The default dahdi.rules file specifies the /dev/dahdi/ directory to be
owned by asterisk:asterisk. What's the contents of your
/etc/udev/rules.d/dahdi.rules file?
 

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com & www.asterisk.org



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

2014-04-18 Thread Russ Meyerriecks

On Fri, Apr 18, 2014 at 2:58 PM,  wrote:

> Russ,
> Here is what I am showing:
>
> asteriskpbx@asterisk-pbx:/etc/udev/rules.d$ cat dahdi.rules
>
> ACTION!="add",  GOTO="dahdi_add_end"
>
> # DAHDI devices with ownership/permissions for running as non-root
> SUBSYSTEM=="dahdi", OWNER="asterisk", GROUP="asterisk",
> MODE="0660"
>

Hmm, this looks right. Does chan_dahdi.so load up properly if you "chown -R
asterisk:asterisk /dev/dahdi" ?

-- 
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Digium, Inc. | Linux Kernel Developer
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direct: +1 256-428-6025
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Re: [asterisk-users] DAHDI loading issue on Asterisk

2014-04-18 Thread steve

Russ,
I think that got it... I had to make a slight change to the command
though:
root@asterisk-pbx:/etc/dahdi# chown -R asteriskpbx:asteriskpbx
/dev/dahdi
 
Astersisk is now showing the dahdi and pri commands. I can see channels
now from the "pri show channels" command. I think I am good to go now.

Thanks for the guidance, have a wonderful weekend!
Steve
  
 Original Message 
Subject: Re: [asterisk-users] DAHDI loading issue on Asterisk
From: Russ Meyerriecks 
Date: Fri, April 18, 2014 1:15 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion


On Fri, Apr 18, 2014 at 2:58 PM,  wrote:
Russ,
Here is what I am showing:

asteriskpbx@asterisk-pbx:/etc/udev/rules.d$ cat dahdi.rules

ACTION!="add",  GOTO="dahdi_add_end"

# DAHDI devices with ownership/permissions for running as non-root
SUBSYSTEM=="dahdi", OWNER="asterisk", GROUP="asterisk",
MODE="0660"


Hmm, this looks right. Does chan_dahdi.so load up properly if you "chown
-R asterisk:asterisk /dev/dahdi" ?
  

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com & www.asterisk.org



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

2014-04-18 Thread Russ Meyerriecks

On Fri, Apr 18, 2014 at 3:36 PM,  wrote:

> Russ,
> I think that got it... I had to make a slight change to the command
> though:
> root@asterisk-pbx:/etc/dahdi# chown -R asteriskpbx:asteriskpbx
> /dev/dahdi
>
> Astersisk is now showing the dahdi and pri commands. I can see channels
> now from the "pri show channels" command. I think I am good to go now.
>
> Thanks for the guidance, have a wonderful weekend!
> Steve


Oh I see, udev must have been setting the ownership to root because the
asterisk user/group doesn't exist on your system.

# DAHDI devices with ownership/permissions for running as non-root
> SUBSYSTEM=="dahdi", OWNER="asterisk", GROUP="asterisk",
> MODE="0660"
>

In the dahdi.rules file, change the owner and group on this line to
"asteriskpbx" so this fix will persist through a module reload.

-- 
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Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] DAHDI loading issue on Asterisk

2014-04-18 Thread steve

Russ,
OK, I see where the disconnect is now. As I mentioned somewhere along
the line there doesn't seem to be a definative guide on building all of
the things from scratch outside of the "Asterisk. The Definitive Guide
3rd Edition" which does touch on the Linux install and making the
Asterisk user "asteriskpbx". As we now see the DAHDI implementation is
looking for the "asterisk" user. This sure can be a touch confusing. 

I did push the server through a restart to see if it would come up and
indeed it did not. Looking at the /dev/dahdi directory everything is set
back to root/root! Time to rebuild the system with the username of
asterisk and go home!!!

Thanks again for the assist.
Steve
 


Oh I see, udev must have been setting the ownership to root because the
asterisk user/group doesn't exist on your system.


# DAHDI devices with ownership/permissions for running as non-root
SUBSYSTEM=="dahdi", OWNER="asterisk", GROUP="asterisk",
MODE="0660"




In the dahdi.rules file, change the owner and group on this line to
"asteriskpbx" so this fix will persist through a module reload.
 

-- 
Russ Meyerriecks
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com & www.asterisk.org



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

2014-04-18 Thread Russ Meyerriecks

On Fri, Apr 18, 2014 at 4:06 PM,  wrote:

> Russ,
>
> I did push the server through a restart to see if it would come up and
> indeed it did not. Looking at the /dev/dahdi directory everything is set
> back to root/root! Time to rebuild the system with the username of
> asterisk and go home!!!
>

My only guess here would be that either the asteriskpbx user or group
doesn't actually exist in one of the two system files /etc/shadow or
/etc/group. I might check there to be sure. If they were missing you could
just do the usual "useradd" and "groupadd" to add them.

-- 
Russ Meyerriecks
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] DAHDI loading issue on Asterisk

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

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

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

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

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

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

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

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