[asterisk-users] on reboot /var/run/asterisk owner changed to root.

[sean darcy]

I've set up runuser as asterisk. /var/run/asterisk is created 
asterisk.asterisk, but each time I reboot, ownership is changed to 
root.root.


How come?  How do I stop the ownership change?

sean


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Re: [asterisk-users] On kernel 3.16.2 : dahdi_rec: Invalid argument

[sean darcy]

On 09/13/2014 12:09 PM, sean darcy wrote:

On Fedora 20, just updated to kernel 3.16.2. Rebuilt dahdi 2.9.2 against
it. dahdi show channels works fine, but when I try to place a call:

chan_dahdi.c:9345 dahdi_read: dahdi_rec: Invalid argument

Any help appreciated.

sean



Updated to dahdi-2.10.0. No joy.

Went back to kernel 3.15.10 - it works.

sean


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[asterisk-users] On kernel 3.16.2 : dahdi_rec: Invalid argument

[sean darcy]

On Fedora 20, just updated to kernel 3.16.2. Rebuilt dahdi 2.9.2 against 
it. dahdi show channels works fine, but when I try to place a call:


chan_dahdi.c:9345 dahdi_read: dahdi_rec: Invalid argument

Any help appreciated.

sean


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[asterisk-users] France DID num2sip setup

[Sean Darcy]

Anyone using the French DID provider num2sip? Could you share the 
sip.conf setup?


Thanks,

sean


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[asterisk-users] Way off topic: gvoice and callcentric

[Sean Darcy]

To deal with google dropping xmpp for voice, I've gotten a callcentric 
number. The cc number connects to asterisk, and all works fine. Then I 
set up the cc number as the gvoice forwarding number. If I'm on the 
gvoice site, I can make a call and it will ring my cc number and then 
the outside number. That also works fine.


BUT, when an outside call comes into gvoice it forwards the call to the 
cc number. I can see the call come into asterisk. But when I answer the 
cc call, the outside call (to the gv number) continues to ring!


I've stuck an Answer() in the dial plan:

Answer("SIP/callcentric19-000c", "") in new stack
-- Executing [1777xxx@from-callcentric:4] 
Goto("SIP/callcentric19-000c", "incoming,s,9") in new stack

-- Goto (incoming,s,9)
-- Executing [s@incoming:9] Dial("SIP/callcentric19-000c", 
"DAHDI/g0&SIP/250&SIP/251&SIP/gn,60,tT") in new stack

...
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 answered SIP/callcentric19-000c
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'

I realize this is not really an asterisk issue (unless there's some 
magic I can put in the dialplan to convince gv the call is answered?) . 
Does anyone have any thoughts on how to fix this? Or where there's a 
forum/mailing list that would be helpful?


sean


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[asterisk-users] unable to transfer ???

[Sean Darcy]

On 11.9.0:



  -- Accepting AUTHENTICATED call from 111.xxx.yyy.zzz:
--> requested format = speex,
--> requested prefs = (),
--> actual format = ulaw,
--> host prefs = (silk16|ulaw|gsm|g722),
--> priority = mine
-- Executing [8447@voip-in:1] Dial("IAX2/n4-5734", "IAX2/ncal") in new stack
-- Called IAX2/ncal
-- Call accepted by 68.xxx.yyy.zzz (format ulaw)
-- Format for call is (ulaw)
-- IAX2/ncal-1777 is ringing
-- IAX2/ncal-1777 answered IAX2/n4-5734
-- Channel 'IAX2/n4-5734' unable to transfer
-- Channel 'IAX2/ncal-1777' unable to transfer
-- Channel 'IAX2/ncal-1777' unable to transfer
-- Hungup 'IAX2/ncal-1777'


What's the problem?

sean


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Re: [asterisk-users] asterisk servers down ?

[Sean Darcy]

On 04/27/2014 01:37 PM, Sean Darcy wrote:

On 04/26/2014 04:42 PM, Joshua Colp wrote:

Sean Darcy wrote:

I can't reach digium.com or asterisk.org. Did I miss the memo?


I have opened a ticket with IT. I'll keep the list apprised when the
problem is isolated and resolved.

Cheers,



Thanks.

Works fine today, FWIW.

sean


Also, here's traceroute from yesterday, for whatever help it gives IT:

traceroute  asterisk.org
traceroute to asterisk.org (216.207.245.25), 30 hops max, 60 byte packets
 ..
 6  451be0ee.cst.lightpath.net (65.19.120.238)  16.139 ms  15.037 ms 
451be0e2.cst.lightpath.net (65.19.120.226)  15.089 ms

 7  * * *
 8  be2327.mpd22.jfk02.atlas.cogentco.com (154.54.1.161)  13.075 ms 
15.157 ms be2325.ccr22.jfk02.atlas.cogentco.com (154.54.47.29)  15.163 ms
 9  be2096.ccr22.bos01.atlas.cogentco.com (154.54.30.42)  20.036 ms 
20.975 ms  20.864 ms
10  be2137.ccr41.ord01.atlas.cogentco.com (154.54.43.193)  55.116 ms 
be2138.ccr42.ord01.atlas.cogentco.com (154.54.43.201)  57.772 ms 
be2140.ccr42.ord01.atlas.cogentco.com (154.54.43.185)  56.534 ms
11  be2101.mpd22.atl01.atlas.cogentco.com (154.54.29.82)  51.199 ms 
be2100.mpd21.atl01.atlas.cogentco.com (154.54.29.66)  60.223 ms  59.764 ms
12  be2171.mpd22.dca01.atlas.cogentco.com (154.54.31.110)  46.306 ms 
be2170.mpd21.dca01.atlas.cogentco.com (154.54.31.106)  47.623 ms 
be2171.mpd22.dca01.atlas.cogentco.com (154.54.31.110)  47.557 ms

13  * * *
14  * te0-3-0-0.mpd22.dca01.atlas.cogentco.com (154.54.31.42)  45.651 ms 
 45.098 ms

15  * * *
16  te0-3-0-0.mpd21.dca01.atlas.cogentco.com (154.54.31.49)  45.464 ms * *
17  te4-1.mag01.dca01.atlas.cogentco.com (154.54.31.41)  79.876 ms * *
18  * * *
19  * * *
20  * * te0-3-0-0.mpd21.dca01.atlas.cogentco.com (154.54.31.49)  48.416 ms
21  * * *
22  te0-3-0-0.mpd22.dca01.atlas.cogentco.com (154.54.31.42)  47.370 ms * 
 48.270 ms

23  * * *
24  te0-3-0-0.mpd21.dca01.atlas.cogentco.com (154.54.31.49)  45.000 ms 
43.934 ms  44.853 ms

25  * * *
26  te0-3-0-0.mpd22.dca01.atlas.cogentco.com (154.54.31.42)  47.354 ms 
te0-3-0-0.mpd21.dca01.atlas.cogentco.com (154.54.31.49)  48.470 ms 
48.308 ms

27  * * *
28  te0-3-0-0.mpd21.dca01.atlas.cogentco.com (154.54.31.49)  48.240 ms 
48.163 ms  48.183 ms

29  * * *
30  te0-3-0-0.mpd22.dca01.atlas.cogentco.com (154.54.31.42)  45.446 ms 
te0-3-0-0.mpd21.dca01.atlas.cogentco.com (154.54.31.49)  45.133 ms 
te0-3-0-0.mpd22.dca01.atlas.cogentco.com (154.54.31.42)  44.993 ms



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[asterisk-users] Does CalDAV require neon-0.29 , not 0.30?

[Sean Darcy]

Asterisk-11.9.0, Fedora 20:

res_calendar_caldav.so => (Asterisk CalDAV Calendar Integration)
[Apr 27 10:49:13] ERROR[4255]: res_calendar_ews.c:911 load_module: 
Exchange Web Service calendar module require neon >= 0.29.1, but neon 
0.30.0: Library build, IPv6, Expat 2.1.0, zlib 1.2.8, GNU TLS 3.1.13. is 
installed.


Is this a bug, or do I need to downgrade to 0.29?

sean


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Re: [asterisk-users] asterisk servers down ?

[Sean Darcy]

On 04/26/2014 04:42 PM, Joshua Colp wrote:

Sean Darcy wrote:

I can't reach digium.com or asterisk.org. Did I miss the memo?


I have opened a ticket with IT. I'll keep the list apprised when the
problem is isolated and resolved.

Cheers,



Thanks.

Works fine today, FWIW.

sean

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[asterisk-users] asterisk servers down ?

[Sean Darcy]

I can't reach digium.com or asterisk.org. Did I miss the memo?

sean


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

[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?

[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] how to configure callcentric peer: no fqdn address matching?

[Sean Darcy]

On 04/15/2014 05:24 PM, Sean Darcy wrote:

On 04/14/2014 11:47 AM, Kelvin Chua wrote:

wild guess would be a conflict on host= setting.
there might be another entity on your sip.conf which have type=friend
and host=callcentric.com or host=204.11.192.161
Kelvin Chua


On Mon, Apr 14, 2014 at 8:01 AM, Sean Darcy  wrote:

On 11.9, trying to set up a callcentric peer:

sip debug:


<--- SIP read from UDP:204.11.192.161:5060 --->
INVITE sip:1777@10.10.11.180:5060 SIP/2.0
v: SIP/2.0/UDP
204.11.192.161:5060;branch=z9hG4bK-6104e46ef4249814d16a2ffb990d
f: @66.193.176.35>;tag=3606475083-968127
t: @ss.callcentric.com>
i: 18075985-3606475083-968...@msw2.telengy.net
CSeq: 1 INVITE
Max-Forwards: 8
m:


Supported: timer
c: application/sdp
l: 350

v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.161
s=sip call
c=IN IP4 204.11.192.161
t=0 0
m=audio 50960 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
<->
--- (11 headers 16 lines) ---
Sending to 204.11.192.161:5060 (NAT)
Sending to 204.11.192.161:5060 (NAT)
Using INVITE request as basis request -
18075985-3606475083-968...@msw2.telengy.net
No matching peer for '' from '204.11.192.161:5060'

<--- Reliably Transmitting (NAT) to 204.11.192.161:5060 --->
SIP/2.0 401 Unauthorized



asterisk is trying to find a peer based on the _calling number_!

Here's the callcentric peer based on its support pages:

[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
defaultuser=1777
;defaultuser=1914
secret=
fromuser=1777
;fromuser=1914
fromdomain=callcentric.com
;fromdomain=ss.callcentric.com
insecure=port,invite
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

As you can see I also tried matching to the callcentric telephone number
instead of the id. No luck. The only number I can't match is the calling
number.

Any help appreciated.

sean


Thanks for the response, but no:

# grep -R host * | grep callcentric
exts/callcentric.sip.conf:host=callcentric.com
# grep -R host * | grep 204
#

Why in the world is asterisk trying to match on the FROM header??

sean




Or, more precisely, since this is a peer context, why isn't it matching 
with the ip address 204.11.192.161?


The fqdn callcentric.com has a number of ip addresses:

host callcentric.com
callcentric.com has address 204.11.192.169
callcentric.com has address 204.11.192.170
callcentric.com has address 204.11.192.171
callcentric.com has address 204.11.192.159
callcentric.com has address 204.11.192.160
callcentric.com has address 204.11.192.161
callcentric.com has address 204.11.192.163

Does asterisk only use the first one?

sip show peer callcentric


  * Name   : callcentric
  Description  :
  Secret   : 
  MD5Secret: 
  Remote Secret: 
  Context  : from-callcentric
.
  ToHost   : callcentric.com
  Addr->IP : 204.11.192.171:5080


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

sean


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Re: [asterisk-users] how to configure callcentric peer

[Sean Darcy]

On 04/14/2014 11:47 AM, Kelvin Chua wrote:

wild guess would be a conflict on host= setting.
there might be another entity on your sip.conf which have type=friend
and host=callcentric.com or host=204.11.192.161
Kelvin Chua


On Mon, Apr 14, 2014 at 8:01 AM, Sean Darcy  wrote:

On 11.9, trying to set up a callcentric peer:

sip debug:


<--- SIP read from UDP:204.11.192.161:5060 --->
INVITE sip:1777@10.10.11.180:5060 SIP/2.0
v: SIP/2.0/UDP
204.11.192.161:5060;branch=z9hG4bK-6104e46ef4249814d16a2ffb990d
f: @66.193.176.35>;tag=3606475083-968127
t: @ss.callcentric.com>
i: 18075985-3606475083-968...@msw2.telengy.net
CSeq: 1 INVITE
Max-Forwards: 8
m:

Supported: timer
c: application/sdp
l: 350

v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.161
s=sip call
c=IN IP4 204.11.192.161
t=0 0
m=audio 50960 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
<->
--- (11 headers 16 lines) ---
Sending to 204.11.192.161:5060 (NAT)
Sending to 204.11.192.161:5060 (NAT)
Using INVITE request as basis request -
18075985-3606475083-968...@msw2.telengy.net
No matching peer for '' from '204.11.192.161:5060'

<--- Reliably Transmitting (NAT) to 204.11.192.161:5060 --->
SIP/2.0 401 Unauthorized



asterisk is trying to find a peer based on the _calling number_!

Here's the callcentric peer based on its support pages:

[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
defaultuser=1777
;defaultuser=1914
secret=
fromuser=1777
;fromuser=1914
fromdomain=callcentric.com
;fromdomain=ss.callcentric.com
insecure=port,invite
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

As you can see I also tried matching to the callcentric telephone number
instead of the id. No luck. The only number I can't match is the calling
number.

Any help appreciated.

sean


Thanks for the response, but no:

# grep -R host * | grep callcentric
exts/callcentric.sip.conf:host=callcentric.com
# grep -R host * | grep 204
#

Why in the world is asterisk trying to match on the FROM header??

sean


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[asterisk-users] how to configure callcentric peer

[Sean Darcy]

On 11.9, trying to set up a callcentric peer:

sip debug:


<--- SIP read from UDP:204.11.192.161:5060 --->
INVITE sip:1777@10.10.11.180:5060 SIP/2.0
v: SIP/2.0/UDP 
204.11.192.161:5060;branch=z9hG4bK-6104e46ef4249814d16a2ffb990d
f: @66.193.176.35>;tag=3606475083-968127
t: @ss.callcentric.com>
i: 18075985-3606475083-968...@msw2.telengy.net
CSeq: 1 INVITE
Max-Forwards: 8
m: 
Supported: timer
c: application/sdp
l: 350

v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.161
s=sip call
c=IN IP4 204.11.192.161
t=0 0
m=audio 50960 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
<->
--- (11 headers 16 lines) ---
Sending to 204.11.192.161:5060 (NAT)
Sending to 204.11.192.161:5060 (NAT)
Using INVITE request as basis request - 
18075985-3606475083-968...@msw2.telengy.net
No matching peer for '' from '204.11.192.161:5060'

<--- Reliably Transmitting (NAT) to 204.11.192.161:5060 --->
SIP/2.0 401 Unauthorized


asterisk is trying to find a peer based on the _calling number_!

Here's the callcentric peer based on its support pages:

[callcentric]
type=peer
context=from-callcentric
host=callcentric.com
defaultuser=1777
;defaultuser=1914
secret=
fromuser=1777
;fromuser=1914
fromdomain=callcentric.com
;fromdomain=ss.callcentric.com
insecure=port,invite
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

As you can see I also tried matching to the callcentric telephone number 
instead of the id. No luck. The only number I can't match is the calling 
number.


Any help appreciated.

sean


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[asterisk-users] originate woes: extension never executes

[Sean Darcy]

Here's my cmd:

originate MOTIF/8447/+12122064...@voice.google.com extension  s@greeting

Greeting:

[greeting]
exten=> s,1,Wait(2)
  same=>n,Background("hello")
  same=>n,Wait(3)


I can see the call go out (also in, since testing on one our own 
numbers), but [greeting] never executes.


I'm expecting to see that when the motif call comes in and is answered, 
the dialplan would connect [greeting] with the answered call.


sean


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[asterisk-users] how can I get authenticate from my own server?

[Sean Darcy]

I'm  used to seeing fraudulent attempts to authenticate, But now I'm 
getting them from the server itself.


I have an asterisk server behind a firewalled router. The local subnet 
is 10.10.10.0/24, the server is 10.10.10.100.


Now I'm seeing in the log lots of:

Failed to authenticate device 
<*>@10.10.10.100:5060>;tag=9c565e6e


How can this happen?

sean


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Re: [asterisk-users] iax2: no authentication, but still peer?

[Sean Darcy]

On 10/08/2013 03:29 PM, Adrian Serafini wrote:

The qualify is on for the peer.  It is failing to reply to the requested
SIP status.  Maybe it is on wifi, screen goes off, wifi follows, zoiper
iax stack doesn't re-reg with the asterisk.


[Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper:
Peer 'n4' is now REACHABLE! Time: 441
[Oct 8 18:15:58] NOTICE[519]: chan_iax2.c:8153 register_verify: Host
 failed MD5 authentication for 'n4'
(c374d0a70c72e6e9bd359aa6a0f1a6c2 != 2c76c104bbfc3d54f566490f40cd12bd)
[Oct 8 18:19:17] NOTICE[517]: chan_iax2.c:11077 socket_process_helper:
Peer 'n4' is now TOO LAGGED (1002 ms)!
[Oct 8 18:19:29] NOTICE[512]: chan_iax2.c:11071 socket_process_helper:
Peer 'n4' is now REACHABLE! Time: 300
[Oct 8 18:26:02] NOTICE[519]: chan_iax2.c:11077 socket_process_helper:
Peer 'n4' is now TOO LAGGED (1017 ms)!
ip-172-31-29-115*CLI> iax2 show peers
Name/Username Host Mask Port Status Description
n4  (D) 255.255.255.255 4569 LAGGED (1017 ms)

is it still registered, or do we really have an authentication problem?

sean





Got it thanks.  So the issue is that zoiper should re-register when the 
wifi comes back on.


sean


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[asterisk-users] iax2: no authentication, but still peer?

[Sean Darcy]

Using zoiper on a nexus 4, asterisk 11.5.1, sometimes we see failed 
authentication. The secret seems correct, so we can't figure out why 
we're getting failed authentication. But at the same time the device 
shows as registered:



[Oct  8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper: 
Peer 'n4' is now REACHABLE! Time: 441
[Oct  8 18:15:58] NOTICE[519]: chan_iax2.c:8153 register_verify: Host 
 failed MD5 authentication for 'n4' 
(c374d0a70c72e6e9bd359aa6a0f1a6c2 != 2c76c104bbfc3d54f566490f40cd12bd)
[Oct  8 18:19:17] NOTICE[517]: chan_iax2.c:11077 socket_process_helper: 
Peer 'n4' is now TOO LAGGED (1002 ms)!
[Oct  8 18:19:29] NOTICE[512]: chan_iax2.c:11071 socket_process_helper: 
Peer 'n4' is now REACHABLE! Time: 300
[Oct  8 18:26:02] NOTICE[519]: chan_iax2.c:11077 socket_process_helper: 
Peer 'n4' is now TOO LAGGED (1017 ms)!

ip-172-31-29-115*CLI> iax2 show peers
Name/UsernameHost Mask PortStatus 
Description

n4  (D)  255.255.255.255  4569   LAGGED (1017 ms)

is it still registered, or do we really have an authentication problem?

sean


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Re: [asterisk-users] iax: unable to transfer - one way audio

[Sean Darcy]

On 09/30/2013 12:09 PM, Sean Darcy wrote:

On 09/28/2013 11:11 AM, Asghar Mohammad wrote:

Hi,
If you post your configuration someone may help you.


On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy mailto:seandar...@gmail.com>> wrote:

On 09/27/2013 09:08 PM, Sean Darcy wrote:

We have zoiper connected over iax to asterisk in Sydney. The
call is to
asterisk in New York. The caller in NZ can hear clearly. Nothing
in NY.

Here's the sydney server:

-- Accepting AUTHENTICATED call from :
 > requested format = speex,
 > requested prefs = (),
 > actual format = ulaw,
 > host prefs = (silk16|ulaw|gsm|g722),
 > priority = mine
  -- Executing [8447@nz-in:1] Dial("IAX2/n4-270",
"IAX2/sydney") in
new stack
  -- Called IAX2/sydney
  -- Call accepted by  (format ulaw)
  -- Format for call is (ulaw)
  -- IAX2/sydney-8819 is ringing
  -- IAX2/sydney-8819 answered IAX2/n4-270
  -- Channel 'IAX2/n4-270' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer

The NY server:

 -- Accepting AUTHENTICATED call from :
  --> requested format = ulaw,
  --> requested prefs = (ulaw|silk16|gsm|g722),
  --> actual format = ulaw,
  --> host prefs = (ulaw|gsm|g722),
  --> priority = mine
  -- Executing [s@incoming-nz:1] Goto("IAX2/home-2152",
"incoming,s,nz-in") in new stack
  -- Goto (incoming,s,5)
  -- Executing [s@incoming:5] Dial("IAX2/home-2152",
"DAHDI/g0&SIP/250&SIP/251,60,__tT") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
  -- Called DAHDI/g0
  -- Called SIP/250
  -- Called SIP/251
  -- DAHDI/1-1 is ringing
  -- SIP/251-001d is ringing
  -- SIP/250-001c is ringing
  -- DAHDI/1-1 is ringing
  -- DAHDI/1-1 answered IAX2/home-2152
  -- Channel 'IAX2/home-2152' unable to transfer
  -- Hanging up on 'DAHDI/1-1'

Any help appreciated.

sean



FWIW, sydney server is 11.5.1, ny server 11.6.0-rc1.

sean




Thanks for the reply.

Here's sydney iax.conf:

[general]
bandwidth=medium

trunkmtu=1240
disallow=all
allow=silk16
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no
trunktimestamps=yes

authdebug=yes

tos=ef
cos=5
autokill=yes
codecpriority=caller

[default](!)
type=friend
auth=md5
host=dynamic
context=nz-in
qualify=1000
setvar=Protocol=IAX2

[n4](default)
secret=
callerid=""

[sydney](default)
secret=
username=home-sydney


home iax.conf:

[general]
bandwidth=medium
disallow=all
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no

tos=0x10
autokill=yes

register => sydney:@

[nz](!)
type=friend
secret=
context=incoming-nz

[home-sydney](nz)
host=
username=sydney
callerid="House"

sean





Any thoughts? Anybody?

sean


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Re: [asterisk-users] iax: unable to transfer - one way audio

[Sean Darcy]

On 09/28/2013 11:11 AM, Asghar Mohammad wrote:

Hi,
If you post your configuration someone may help you.


On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy mailto:seandar...@gmail.com>> wrote:

On 09/27/2013 09:08 PM, Sean Darcy wrote:

We have zoiper connected over iax to asterisk in Sydney. The
call is to
asterisk in New York. The caller in NZ can hear clearly. Nothing
in NY.

Here's the sydney server:

-- Accepting AUTHENTICATED call from :
 > requested format = speex,
 > requested prefs = (),
 > actual format = ulaw,
 > host prefs = (silk16|ulaw|gsm|g722),
 > priority = mine
  -- Executing [8447@nz-in:1] Dial("IAX2/n4-270",
"IAX2/sydney") in
new stack
  -- Called IAX2/sydney
  -- Call accepted by  (format ulaw)
  -- Format for call is (ulaw)
  -- IAX2/sydney-8819 is ringing
  -- IAX2/sydney-8819 answered IAX2/n4-270
  -- Channel 'IAX2/n4-270' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer
  -- Channel 'IAX2/sydney-8819' unable to transfer

The NY server:

 -- Accepting AUTHENTICATED call from :
  --> requested format = ulaw,
  --> requested prefs = (ulaw|silk16|gsm|g722),
  --> actual format = ulaw,
  --> host prefs = (ulaw|gsm|g722),
  --> priority = mine
  -- Executing [s@incoming-nz:1] Goto("IAX2/home-2152",
"incoming,s,nz-in") in new stack
  -- Goto (incoming,s,5)
  -- Executing [s@incoming:5] Dial("IAX2/home-2152",
"DAHDI/g0&SIP/250&SIP/251,60,__tT") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
  -- Called DAHDI/g0
  -- Called SIP/250
  -- Called SIP/251
  -- DAHDI/1-1 is ringing
  -- SIP/251-001d is ringing
  -- SIP/250-001c is ringing
  -- DAHDI/1-1 is ringing
  -- DAHDI/1-1 answered IAX2/home-2152
  -- Channel 'IAX2/home-2152' unable to transfer
  -- Hanging up on 'DAHDI/1-1'

Any help appreciated.

sean



FWIW, sydney server is 11.5.1, ny server 11.6.0-rc1.

sean




Thanks for the reply.

Here's sydney iax.conf:

[general]
bandwidth=medium

trunkmtu=1240
disallow=all
allow=silk16
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no
trunktimestamps=yes

authdebug=yes

tos=ef
cos=5
autokill=yes
codecpriority=caller

[default](!)
type=friend
auth=md5
host=dynamic
context=nz-in
qualify=1000
setvar=Protocol=IAX2

[n4](default)
secret=
callerid=""

[sydney](default)
secret=
username=home-sydney


home iax.conf:

[general]
bandwidth=medium
disallow=all
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no

tos=0x10
autokill=yes

register => sydney:@

[nz](!)
type=friend
secret=
context=incoming-nz

[home-sydney](nz)
host=
username=sydney
callerid="House"

sean



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Re: [asterisk-users] iax: unable to transfer - one way audio

[Sean Darcy]

On 09/27/2013 09:08 PM, Sean Darcy wrote:

We have zoiper connected over iax to asterisk in Sydney. The call is to
asterisk in New York. The caller in NZ can hear clearly. Nothing in NY.

Here's the sydney server:

-- Accepting AUTHENTICATED call from :
> requested format = speex,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (silk16|ulaw|gsm|g722),
> priority = mine
 -- Executing [8447@nz-in:1] Dial("IAX2/n4-270", "IAX2/sydney") in
new stack
 -- Called IAX2/sydney
 -- Call accepted by  (format ulaw)
 -- Format for call is (ulaw)
 -- IAX2/sydney-8819 is ringing
 -- IAX2/sydney-8819 answered IAX2/n4-270
 -- Channel 'IAX2/n4-270' unable to transfer
 -- Channel 'IAX2/sydney-8819' unable to transfer
 -- Channel 'IAX2/sydney-8819' unable to transfer
 -- Channel 'IAX2/sydney-8819' unable to transfer

The NY server:

-- Accepting AUTHENTICATED call from :
 --> requested format = ulaw,
 --> requested prefs = (ulaw|silk16|gsm|g722),
 --> actual format = ulaw,
 --> host prefs = (ulaw|gsm|g722),
 --> priority = mine
 -- Executing [s@incoming-nz:1] Goto("IAX2/home-2152",
"incoming,s,nz-in") in new stack
 -- Goto (incoming,s,5)
 -- Executing [s@incoming:5] Dial("IAX2/home-2152",
"DAHDI/g0&SIP/250&SIP/251,60,tT") in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Called DAHDI/g0
 -- Called SIP/250
 -- Called SIP/251
 -- DAHDI/1-1 is ringing
 -- SIP/251-001d is ringing
 -- SIP/250-001c is ringing
 -- DAHDI/1-1 is ringing
 -- DAHDI/1-1 answered IAX2/home-2152
 -- Channel 'IAX2/home-2152' unable to transfer
 -- Hanging up on 'DAHDI/1-1'

Any help appreciated.

sean




FWIW, sydney server is 11.5.1, ny server 11.6.0-rc1.

sean


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[asterisk-users] iax: unable to transfer - one way audio

[Sean Darcy]

We have zoiper connected over iax to asterisk in Sydney. The call is to 
asterisk in New York. The caller in NZ can hear clearly. Nothing in NY.


Here's the sydney server:

-- Accepting AUTHENTICATED call from :
   > requested format = speex,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (silk16|ulaw|gsm|g722),
   > priority = mine
-- Executing [8447@nz-in:1] Dial("IAX2/n4-270", "IAX2/sydney") in 
new stack

-- Called IAX2/sydney
-- Call accepted by  (format ulaw)
-- Format for call is (ulaw)
-- IAX2/sydney-8819 is ringing
-- IAX2/sydney-8819 answered IAX2/n4-270
-- Channel 'IAX2/n4-270' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer

The NY server:

   -- Accepting AUTHENTICATED call from :
--> requested format = ulaw,
--> requested prefs = (ulaw|silk16|gsm|g722),
--> actual format = ulaw,
--> host prefs = (ulaw|gsm|g722),
--> priority = mine
-- Executing [s@incoming-nz:1] Goto("IAX2/home-2152", 
"incoming,s,nz-in") in new stack

-- Goto (incoming,s,5)
-- Executing [s@incoming:5] Dial("IAX2/home-2152", 
"DAHDI/g0&SIP/250&SIP/251,60,tT") in new stack

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called DAHDI/g0
-- Called SIP/250
-- Called SIP/251
-- DAHDI/1-1 is ringing
-- SIP/251-001d is ringing
-- SIP/250-001c is ringing
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 answered IAX2/home-2152
-- Channel 'IAX2/home-2152' unable to transfer
-- Hanging up on 'DAHDI/1-1'

Any help appreciated.

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/10/2013 12:15 PM, Joshua Colp wrote:

Sean Darcy wrote:

Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?


Asterisk does not assign ports. The IAX2 channel driver, by default,
binds to a single UDP port (4569). As UDP is connectionless there are no
connections. What you see on the console is the *source* IP address and
port of the packets. It's possible that the Amazon stuff is sort of
NATting things to do connection tracking... but that's Amazon land, so
no clue really.



Since no horse is dead enough not to take another beating:

If the console is showing the *source* port of the packets. then:

does the server send iax packets to that source port, or to 4569?

"home" (which is another asterisk server) shows 4569, while the androids 
running zoiper show random ports. I assume zoiper puts the source port 
in an iax packet. But regardless of how zoiper describes its source 
port, asterisk will only send iax packets on 4569. correct?


I ask all this because Amazon EC2 uses a firewall that doesn't have a 
connection state. All incoming ports are blocked unless they are 
explicitly opened. Just having a packet go out to an ip address and 
port, doesn't open the source port.


But if iax is always and only using 4569 to send and receive, I don't 
have to worry about opening any other ports.


Thanks,

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/10/2013 05:27 PM, Joshua Colp wrote:

Sean Darcy wrote:

On 09/10/2013 12:15 PM, Joshua Colp wrote:

Sean Darcy wrote:

Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?


Asterisk does not assign ports. The IAX2 channel driver, by default,
binds to a single UDP port (4569). As UDP is connectionless there are no
connections. What you see on the console is the *source* IP address and
port of the packets. It's possible that the Amazon stuff is sort of
NATting things to do connection tracking... but that's Amazon land, so
no clue really.



Since no horse is dead enough not to take another beating:

If the console is showing the *source* port of the packets. then:

does the server send iax packets to that source port, or to 4569?


It sends to the source port if using the registration.



"home" (which is another asterisk server) shows 4569, while the androids
running zoiper show random ports. I assume zoiper puts the source port
in an iax packet. But regardless of how zoiper describes its source
port, asterisk will only send iax packets on 4569. correct?


It does not put the source port in an IAX packet. It's in the IP header
itself, outside of IAX. Asterisk will send IAX packets *from* port 4569
but *to* any host/port.



OK, so I only need to open up 4569 incoming, But I need to allow a range 
of outgoing udp ports since zoiper is choosing other udp ports in the IP 
header of the iax registration.


Thanks. Sorry it's taken so long for me to get this.

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 07:48 PM, Eric Wieling wrote:

Try this as an example of why it doesn't matter.

1) On windows open a cmd prompt or on linux open up a local terminal.
2) open a web browser and connect to a web site like cnn.com
3) on windows type "netstat -n" in the command prompt, in linux type netstat -n 
--ip

For example on my system, the local IP is 172.17.3.111.  Notice below how the 
port on my local system is NOT 80, even though the port on the remote system 
is?   This is simply how TCP and UDP work.  When you are looking at your iax 
peers you are seeing the REMOTE IP and REMOTE port, which seldom matters.  It 
is the port on the client you are connecting TO which matters, not the port 
which you are connecting FROM. TCP and UDP do not allow more than one 
connection using the same source IP/source port/destination IP/destination port 
(called a tuple).  For most things the source port does not matter so the 
operating system assigns whatever source port it wants to.   NAT routers will 
often change the source port when the connection is NAT'd.  These are 
fundamental IP networking concepts whi
  ch all people doing VoIP should know, but most don't. I'm sure there are 
many books on TCP/IP networking which explain it better than I have explained 
it.

Active Connections

   Proto  Local Address  Foreign AddressState
TCP172.17.3.111:22020 157.166.226.25:80  ESTABLISHED
  TCP172.17.3.111:22021 157.166.249.10:80  ESTABLISHED
  TCP172.17.3.111:22022 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22023 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22024 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22025 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22026 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22027 23.203.4.211:80ESTABLISHED
  TCP172.17.3.111:22028 23.63.227.185:80   ESTABLISHED
  TCP172.17.3.111:22029 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22030 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22031 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22032 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22033 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22034 4.27.18.126:80 ESTABLISHED
  TCP172.17.3.111:22035 74.217.240.83:80   ESTABLISHED
  TCP172.17.3.111:22036 23.63.227.123:80   ESTABLISHED
  TCP172.17.3.111:22037 12.130.81.225:80   ESTABLISHED
  TCP172.17.3.111:22038 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22039 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22040 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22041 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22042 4.26.252.126:80ESTABLISHED
  TCP172.17.3.111:22043 4.26.252.126:80ESTABLISHED

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 7:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip 
address

On 09/09/2013 03:37 PM, Eric Wieling wrote:

Again, that port is assigned by your NAT router.  Asterisk cannot control the 
source port if the incoming packet.   That is set by your NAT router and client 
and likely has nothing to do with your problem.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean
Darcy
Sent: Monday, September 09, 2013 3:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from
same ip address

Dial("IAX2/home-14358", "IAX2/gn") in new stack
   -- Called IAX2/gn
CLI> iax2 show peers
Name/UsernameHost Mask Port
Status  Description
gn (D)  255.255.255.255  9007  OK
(179 ms)

[Sep  9 19:11:36] WARNING[530]: chan_iax2.c:3552 __attempt_transmit: Max retries 
exceeded to host  on IAX2/gn-11311 (type = 6, subclass = 11, 
ts=10018, seqno=1)
   -- Hungup 'IAX2/gn-11311'

Again, what's with this port 9007? Is asterisk assigning it? I thought all iax 
traffic went over 4569.

Of course, this could be a zoiper problem.

sean



But the problem is it's not MY nat router; it's amazon's. And if you only have only have 
one iax device registered, it's always 4569, So why does amazon assign a different port 
to the second iax device? How would it even "know"?

sean



Well, I may be confused, but iax show peers is showing the remote port, 
the port it will connect TO, right?


netstat doesn't show the asterisk connections at all, ju

Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 03:37 PM, Eric Wieling wrote:

Again, that port is assigned by your NAT router.  Asterisk cannot control the 
source port if the incoming packet.   That is set by your NAT router and client 
and likely has nothing to do with your problem.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 3:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip 
address

Dial("IAX2/home-14358", "IAX2/gn") in new stack
  -- Called IAX2/gn
CLI> iax2 show peers
Name/UsernameHost Mask Port
Status  Description
gn (D)  255.255.255.255  9007  OK
(179 ms)

[Sep  9 19:11:36] WARNING[530]: chan_iax2.c:3552 __attempt_transmit: Max retries 
exceeded to host  on IAX2/gn-11311 (type = 6, subclass = 11, 
ts=10018, seqno=1)
  -- Hungup 'IAX2/gn-11311'

Again, what's with this port 9007? Is asterisk assigning it? I thought all iax 
traffic went over 4569.

Of course, this could be a zoiper problem.

sean



But the problem is it's not MY nat router; it's amazon's. And if you 
only have only have one iax device registered, it's always 4569, So why 
does amazon assign a different port to the second iax device? How would 
it even "know"?


sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 01:54 PM, Joshua Colp wrote:

Sean Darcy wrote:


home is from the home machine, which registers with the server:

register => home:@

[home]
type=friend
insecure=port,invite
secret= ; same secret as on server
context=incoming
host=


You aren't specifying what username to authenticate as here. Add:

username=home

And give it a go.



Excellent! It's so easy to overlook the obvious.

But now I can't call "gn". I can call out from gn, but calling to gn 
dies with Max retries...


Dial("IAX2/home-14358", "IAX2/gn") in new stack
-- Called IAX2/gn
CLI> iax2 show peers
Name/UsernameHost Mask Port 
Status  Description
gn (D)  255.255.255.255  9007  OK 
(179 ms)


[Sep  9 19:11:36] WARNING[530]: chan_iax2.c:3552 __attempt_transmit: Max 
retries exceeded to host  on IAX2/gn-11311 (type = 6, subclass 
= 11, ts=10018, seqno=1)

-- Hungup 'IAX2/gn-11311'

Again, what's with this port 9007? Is asterisk assigning it? I thought 
all iax traffic went over 4569.


Of course, this could be a zoiper problem.

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 11:08 AM, Joshua Colp wrote:

Sean Darcy wrote:


On the server each device has type=friend.

I do notice that peer "home" has the standard iax port 4569. The other
peers are assigned 1026, 1027 and 1028. How are these ports assigned?


The actual configuration entries (minus password) for each one involved
would be useful... if you aren't being explicit with what username to
use for outgoing authentication then stuff like this can happen.



On the server:

[default](!)
type=friend
auth=md5
host=dynamic
context=nz-in
qualify=1000
setvar=Protocol=IAX2

[gn](default)
secret=
callerid="GN"

[home](default)
secret=
username=home

I'm using Zoiper on Android for gn,

home is from the home machine, which registers with the server:

register => home:@

[home]
type=friend
insecure=port,invite
secret=; same secret as on server
context=incoming
host=

I'm wondering if it's a result of the amazon ec2 firewall (not 
iptables). I may need to open up those lower udp ports. Maybe the amazon 
firewall doesn't use ctstate; it may block any port not explicitly 
opened even if a connection is established.


Thanks for the help.

sean


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Re: [asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

On 09/09/2013 08:04 AM, Julian Beach wrote:

Hello Sean,

Sunday, September 8, 2013, 11:25:24 PM, you wrote:


The problem is that once a phone has used the server, no other phone can
use it. The servers sees all the phones as having the same ip address
(though different ports).


This  sounds  like  the  Peer v Friend problem I have had in the past.
Try  setting  user=friend which will match on the username and not IP
address.  I  found  that asterisk was matching to the first account in
the  list  in  IAX.CONF  and  authentication  was  then failing (or in
the case of incoming calls, ending up in the wrong context).

http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

Julian



Thanks for the response.

On the server each device has type=friend.

I do notice that peer "home" has the standard iax port 4569. The other 
peers are assigned 1026, 1027 and 1028. How are these ports assigned?


sean


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[asterisk-users] iax2: two users can't authenticate from same ip address

[Sean Darcy]

I'm trying set up asterisk on an amazon instance in Sydney. It's to use 
for our kids in Sydney to connect with their friends in the States.


We've found iax works better than sip with these distances. But we now 
have weird problem: everybody has a cell phone, and it's much 
cheaper/better to use the house internet connection over the phones 
wifi. Each cell phone has it's own peer. Each cell phone registers with 
the server.


The problem is that once a phone has used the server, no other phone can 
use it. The servers sees all the phones as having the same ip address 
(though different ports).


iax2 show peers
Name/UsernameHostMask Port  Status 
Description
gn  (D)  255.255.255.255  1026  OK (101 
ms)
home/home   (D)  255.255.255.255  4569  OK (85 
ms)


but "home" can't make a call:
chan_iax2.c:11157 socket_process_helper: Host  failed to 
authenticate as gn


gn can make calls:

-- Registered IAX2 'gn' (AUTHENTICATED) at :1026
-- Accepting AUTHENTICATED call from :
   > requested format = speex,
  


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[asterisk-users] permission problems on amazon ec2

[Sean Darcy]

I'm marching forward trying to get asterisk running on a amazon EC2 
instance, Fedora 19.


If I start it from the terminal all works. I can login as  user 
"asterisk" and start asterisk.


But if I try to use systemctl to start it automatically I get the error 
it doesn't have the permission to open asterisk.conf.


cat /usr/lib/systemd/system/asterisk.service
[Unit]
Description=Asterisk PBX and telephony daemon.
After=network.target

[Service]
Type=simple
Environment=HOME=/var/lib/asterisk
WorkingDirectory=/var/lib/asterisk
User=asterisk
Group=asterisk
ExecStart=/usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf
ExecStop=/usr/sbin/asterisk -rx 'core stop now'
ExecReload=/usr/sbin/asterisk -rx 'core reload'

and if I login as asterisk:

[asterisk@myip ~]$ /usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf

it works. I can login from another terminal.

But if I do systemctl start asterisk I get:

systemd[1]: Started Asterisk PBX and telephony daemon..
asterisk[711]: Unable to open specified master config file 
'/etc/asterisk/asterisk.conf', using built-in defaults


ls -l /etc/asterisk/asterisk.conf
-rw-r--r--. 1 asterisk asterisk 3938 Sep  7 22:19 
/etc/asterisk/asterisk.conf


Going nuts here.

sean


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Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

[Sean Darcy]

On 09/07/2013 01:26 PM, Tony Mountifield wrote:

In article ,
Sean Darcy  wrote:

On 09/07/2013 10:33 AM, Tony Mountifield wrote:

In article <522a934d.8010...@gmail.com>,
Sean Darcy  wrote:

On 09/06/2013 07:08 PM, Steve Edwards wrote:

On Fri, 6 Sep 2013, Sean Darcy wrote:


I'm not sure asterisk is even listening for the packets:

[root@asterisk ~]# netstat -apnt | grep 4569
[root@asterisk ~]#


'-t' meand TCP. IAX is UDP.



My bad:

netstat -apnu | grep 4569
udp0  0 0.0.0.0:45690.0.0.0:*
   3176/asterisk

But why isn't asterisk seeing/acting upon the registration request?
Wireshark finds the packet to 4569, so it's not a firewall problem.


Are you sure about that? I have found in the past that tcpdump sees inbound
packets before they get to the iptables filter.

What happens if you do:
iptables -I INPUT 1 -p udp --dport 4569 -j ACCEPT

Cheers
Tony



Wow! Look:

   iptables -L
Chain INPUT (policy ACCEPT)
target prot opt source   destination
ACCEPT all  --  anywhere anywhere ctstate
RELATED,ESTABLISHED
ACCEPT icmp --  anywhere anywhere
ACCEPT all  --  anywhere anywhere
ACCEPT tcp  --  anywhere anywhere ctstate
NEW tcp dpt:ssh
REJECT all  --  anywhere anywhere
reject-with icmp-host-prohibited

Chain FORWARD (policy ACCEPT)
target prot opt source   destination
REJECT all  --  anywhere anywhere
reject-with icmp-host-prohibited

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination


Which means to me that the INPUT chain will ACCEPT all protocols from
anywhere to anywhere.


I suspect there's something that is not being shown there. Try:

# iptables -vnL

(and if pasting it, to post here, try to avoid line-wrapping if possible).


But no, iptables -I INPUT 1 -p udp --dport 4569 -j ACCEPT solves the
problem and asterisk now registers my device.

Now I have to find a way to make it persistent across reboots.


If your system is RH or CentOS-like, you can do:

# service iptables save

That creates the file /etc/sysconfig/iptables, which is loaded on boot.

Cheers
Tony




iptables -vnL
Chain INPUT (policy ACCEPT 0 packets, 0 bytes)
 pkts bytes target prot opt in out source 
destination
 125K  171M ACCEPT all  --  *  *   0.0.0.0/0 
0.0.0.0/0  ctstate RELATED,ESTABLISHED
0 0 ACCEPT icmp --  *  *   0.0.0.0/0 
0.0.0.0/0
0 0 ACCEPT all  --  lo *   0.0.0.0/0 
0.0.0.0/0
   13   768 ACCEPT tcp  --  *  *   0.0.0.0/0 
0.0.0.0/0  ctstate NEW tcp dpt:22

140 REJECT all  --  *  *   0.0.0.0/0 0.0.0.0/0

So this means the packet is accepted only if it comes from the loopback 
interface?


I've disabled iptables altogether, now relying on the amazon security group.

Thanks for your help.

sean


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Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

[Sean Darcy]

On 09/07/2013 10:33 AM, Tony Mountifield wrote:

In article <522a934d.8010...@gmail.com>,
Sean Darcy  wrote:

On 09/06/2013 07:08 PM, Steve Edwards wrote:

On Fri, 6 Sep 2013, Sean Darcy wrote:


I'm not sure asterisk is even listening for the packets:

[root@asterisk ~]# netstat -apnt | grep 4569
[root@asterisk ~]#


'-t' meand TCP. IAX is UDP.



My bad:

netstat -apnu | grep 4569
udp0  0 0.0.0.0:45690.0.0.0:*
  3176/asterisk

But why isn't asterisk seeing/acting upon the registration request?
Wireshark finds the packet to 4569, so it's not a firewall problem.


Are you sure about that? I have found in the past that tcpdump sees inbound
packets before they get to the iptables filter.

What happens if you do:
iptables -I INPUT 1 -p udp --dport 4569 -j ACCEPT

Cheers
Tony



Wow! Look:

 iptables -L
Chain INPUT (policy ACCEPT)
target prot opt source   destination
ACCEPT all  --  anywhere anywhere ctstate 
RELATED,ESTABLISHED

ACCEPT icmp --  anywhere anywhere
ACCEPT all  --  anywhere anywhere
ACCEPT tcp  --  anywhere anywhere ctstate 
NEW tcp dpt:ssh
REJECT all  --  anywhere anywhere 
reject-with icmp-host-prohibited


Chain FORWARD (policy ACCEPT)
target prot opt source   destination
REJECT all  --  anywhere anywhere 
reject-with icmp-host-prohibited


Chain OUTPUT (policy ACCEPT)
target prot opt source   destination


Which means to me that the INPUT chain will ACCEPT all protocols from 
anywhere to anywhere.


But no, iptables -I INPUT 1 -p udp --dport 4569 -j ACCEPT solves the 
problem and asterisk now registers my device.


Now I have to find a way to make it persistent across reboots.

Thanks,

sean


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Re: [asterisk-users] 11.4.0: iax packets lost by amazon ec2

[Sean Darcy]

On 09/06/2013 07:08 PM, Steve Edwards wrote:

On Fri, 6 Sep 2013, Sean Darcy wrote:


I'm not sure asterisk is even listening for the packets:

[root@asterisk ~]# netstat -apnt | grep 4569
[root@asterisk ~]#


'-t' meand TCP. IAX is UDP.



My bad:

netstat -apnu | grep 4569
udp0  0 0.0.0.0:45690.0.0.0:* 
3176/asterisk


But why isn't asterisk seeing/acting upon the registration request? 
Wireshark finds the packet to 4569, so it's not a firewall problem.


Is it an asterisk configuration problem?

Does anybody else have iax working on ec2?

sean

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[asterisk-users] 11.4.0: iax packets lost by amazon ec2

[Sean Darcy]

I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get 
iax to work.


I've opened 4569 in the EC2 Security Group.

I'm using the zoiper client. Using tcpdump I can see the zoiper packets 
coming in on 4569, but nothing shows on the asterisk cli.


Frame 33: 79 bytes on wire (632 bits), 79 bytes captured (632 bits) on 
interface 0


   12 31 3b 12 40 84 fe ff ff ff ff ff 08 00 45 00  .1;.@.E.
0010   00 41 00 00 40 00 2f 11 37 f1 44 c7 80 b8 0a ca  .A..@./.7.D.
0020   43 72 11 d9 11 d9 00 2d fd 22 80 04 80 00 00 00  Cr.-."..
0030   00 02 00 00 06 0d 06 02 67 6e 13 02 00 3c 36 00  gn...<6.
0040   24 0d 5a 6f 69 70 65 72 20 72 31 38 39 37 36 $.Zoiper r18976

iptables is flushed.


I'm not sure asterisk is even listening for the packets:

[root@asterisk ~]# netstat -apnt | grep 5060
tcp0  0 0.0.0.0:50600.0.0.0:* 
LISTEN  1706/asterisk

[root@asterisk ~]# netstat -apnt | grep 4569
[root@asterisk ~]#

But it should be:

ip-10-202-67-114*CLI> iax2 show peer gn


  * Name   : gn
  Description  :
  Secret   : 
  Context  : nexus-in
  Parking lot  :
  Mailbox  :
  Dynamic  : Yes
  Callnum limit: 0
  Calltoken req: No
  Trunk: No
  Encryption   : No
  Callerid : "" <>
  Expire   : -1
  ACL  : No
  Addr->IP : (Unspecified) Port 0
  Defaddr->IP  : 0.0.0.0 Port 4569
  Username :
  Codecs   : (gsm|ulaw|g722)
  Codec Order  : (silk16|ulaw|gsm|g722)
  Status   : UNKNOWN
  Qualify  : every 6ms when OK, every 1ms when UNREACHABLE 
(sample smoothing Off)


iax.conf:

[general]
bandwidth=medium

trunkmtu=1240
disallow=all
allow=silk16
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no

authdebug=yes

tos=ef
cos=5
autokill=yes
codecpriority=caller

[gn]
type=friend
auth=md5
secret=mine
host=dynamic
context=nexus-in
qualify=yes

Thanks for any help.

sean


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Re: [asterisk-users] no silk translation ?

[Sean Darcy]

On 06/10/2013 05:24 PM, Sean Darcy wrote:

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:

[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng- to Motif/+12025551...@voice.google.com-da3c
[Jun 10 16:18:22] WARNING[4090][C-000a]: app_dial.c:3032
dial_exec_full: Had to drop call because I couldn't make SIP/ng-
compatible with Motif/+12025551...@voice.google.com-da3c
   == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-'

core show translations doesn't include any SILK.

SILK is installed:

core show codec 100018
  100018 SILK Custom Format 8khz
  100018 SILK Custom Format 12khz
  100018 SILK Custom Format 16khz
  100018 SILK Custom Format 24khz

sean



Maybe it's because SILK is "unknown"  :

 == Registered translator 'silktolin' from format unknown to slin, 
table cost, 90, computational cost 99
  == Registered translator 'lintosilk' from format slin to unknown, 
table cost, 60, computational cost 99
  == Registered translator 'silktolin12' from format unknown to slin12, 
table cost, 93, computational cost 99
  == Registered translator '12lintosilk' from format slin12 to unknown, 
table cost, 875000, computational cost 99
  == Registered translator 'silktolin16' from format unknown to slin16, 
table cost, 93, computational cost 99
  == Registered translator '16lintosilk' from format slin16 to unknown, 
table cost, 875000, computational cost 99
  == Registered translator 'silktolin24' from format unknown to slin24, 
table cost, 93, computational cost 99
  == Registered translator '24lintosilk' from format slin24 to unknown, 
table cost, 875000, computational cost 99

 codec_silk.so => (Silk Transcoder)

sean


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Re: [asterisk-users] no silk translation ?

[Sean Darcy]

On 06/10/2013 05:24 PM, Sean Darcy wrote:

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:

[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng- to Motif/+12025551...@voice.google.com-da3c
[Jun 10 16:18:22] WARNING[4090][C-000a]: app_dial.c:3032
dial_exec_full: Had to drop call because I couldn't make SIP/ng-
compatible with Motif/+12025551...@voice.google.com-da3c
   == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-'

core show translations doesn't include any SILK.

SILK is installed:

core show codec 100018
  100018 SILK Custom Format 8khz
  100018 SILK Custom Format 12khz
  100018 SILK Custom Format 16khz
  100018 SILK Custom Format 24khz

sean



Maybe the reason SILK is not showing up, is because it's "unknown":

  == Registered translator 'silktolin' from format unknown to slin, 
table cost, 90, computational cost 99
  == Registered translator 'lintosilk' from format slin to unknown, 
table cost, 60, computational cost 99
  == Registered translator 'silktolin12' from format unknown to slin12, 
table cost, 93, computational cost 99
  == Registered translator '12lintosilk' from format slin12 to unknown, 
table cost, 875000, computational cost 99
  == Registered translator 'silktolin16' from format unknown to slin16, 
table cost, 93, computational cost 99
  == Registered translator '16lintosilk' from format slin16 to unknown, 
table cost, 875000, computational cost 99
  == Registered translator 'silktolin24' from format unknown to slin24, 
table cost, 93, computational cost 99
  == Registered translator '24lintosilk' from format slin24 to unknown, 
table cost, 875000, computational cost 99

 codec_silk.so => (Silk Transcoder)

sean


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[asterisk-users] no silk translation ?

[Sean Darcy]

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but 
no success:


[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 
ast_channel_make_compatible_helper: No path to translate from 
SIP/ng- to Motif/+12025551...@voice.google.com-da3c
[Jun 10 16:18:22] WARNING[4090][C-000a]: app_dial.c:3032 
dial_exec_full: Had to drop call because I couldn't make SIP/ng- 
compatible with Motif/+12025551...@voice.google.com-da3c

  == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-'

core show translations doesn't include any SILK.

SILK is installed:

core show codec 100018
 100018 SILK Custom Format 8khz
 100018 SILK Custom Format 12khz
 100018 SILK Custom Format 16khz
 100018 SILK Custom Format 24khz

sean


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[asterisk-users] db.c:329 ast_db_put: Couldn't execute statment: SQL logic error or missing database ??

[Sean Darcy]

I'm showing a lot of these on the console. I'm not using any database.

Where would this be coming from?

sean


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Re: [asterisk-users] how to send dtmf after pause ?

[Sean Darcy]

On 06/07/2013 01:48 PM, Asghar Mohammad wrote:

hi,
you can add more w (ww1234#) for more delay.



On Fri, Jun 7, 2013 at 7:17 PM, Yves A. mailto:yves...@gmx.de>> wrote:

This would be possible with an agi...
the agi can wait for silence or 10 seconds, as u like and then play
the dtmf tones and bridge the call to your extension afterwards.

yves

Am 07.06.2013 17:51, schrieb Sean Darcy:


I'm trying to call a conference service, wait 10 seconds, then
send the passcode.

I've tried ww:

Dial(SIP/18005551212ww12345#@s__ip.com <http://sip.com>,60,r)

The sip channel didn't like that. Added 'p' , still no help.

I tried D:

Dial(SIP/18005551...@sip.com
<mailto:18005551...@sip.com>,__60,rD(12345#)

The dtmf is sent too soon. I tried inserting 'ww' but that was
just sent.

I tried G:

exten => 234.1.Dial(SIP/18005551212@__sip.com
<mailto:18005551...@sip.com>,60,rG(next))
  same=>n(next),Wait(10)
  same=>n,SendDTMF(12345#)

but that didn't work at all,

This is a common use case. There must be some simple answer I'm
missing.

Thanks for any help.

sean





Thanks for the reply, but any 'w' s in the dial string cause 
CHAN_UNAVAILABLE.


I'm not sure I'm up for learning agi just yet. I was hoping for a 
dialplan solution.


sean


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Re: [asterisk-users] how to send dtmf after pause ?

[Sean Darcy]

On 06/07/2013 01:17 PM, Yves A. wrote:

This would be possible with an agi...
the agi can wait for silence or 10 seconds, as u like and then play the
dtmf tones and bridge the call to your extension afterwards.

yves

Am 07.06.2013 17:51, schrieb Sean Darcy:


I'm trying to call a conference service, wait 10 seconds, then send
the passcode.

I've tried ww:

Dial(SIP/18005551212ww12345#@sip.com,60,r)

The sip channel didn't like that. Added 'p' , still no help.

I tried D:

Dial(SIP/18005551...@sip.com,60,rD(12345#)

The dtmf is sent too soon. I tried inserting 'ww' but that was just sent.

I tried G:

exten => 234.1.Dial(SIP/18005551...@sip.com,60,rG(next))
 same=>n(next),Wait(10)
 same=>n,SendDTMF(12345#)

but that didn't work at all,

This is a common use case. There must be some simple answer I'm missing.

Thanks for any help.

sean



Thanks for the response. My agi mojo is not strong. I was hoping to do 
this with dialplan logic.


sean


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[asterisk-users] how to send dtmf after pause ?

[Sean Darcy]

I'm trying to call a conference service, wait 10 seconds, then send the 
passcode.


I've tried ww:

Dial(SIP/18005551212ww12345#@sip.com,60,r)

The sip channel didn't like that. Added 'p' , still no help.

I tried D:

Dial(SIP/18005551...@sip.com,60,rD(12345#)

The dtmf is sent too soon. I tried inserting 'ww' but that was just sent.

I tried G:

exten => 234.1.Dial(SIP/18005551...@sip.com,60,rG(next))
 same=>n(next),Wait(10)
 same=>n,SendDTMF(12345#)

but that didn't work at all,

This is a common use case. There must be some simple answer I'm missing.

Thanks for any help.

sean


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Re: [asterisk-users] 11.4: motif can only handle one channel at a time?

[sean darcy]

On 05/16/2013 10:07 AM, sean darcy wrote:

On 05/16/2013 09:41 AM, sean darcy wrote:

I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.

motif only handles one 1 channel at a time??

sean



More:

Two different motif sections. Two different xmpp sections.
xmpp shows both connections.

sean



I've made some headway, but not enough.

I have a motif call coming in. That bridges to a motif call going out.



-- Executing [s@incoming:2] Wait("Motif/+1-2927", "1") in new stack
-- Executing [s@incoming:3] Answer("Motif/+1-2927", "") in new stack
-- Executing [s@incoming:4] Set("Motif/+1-2927", 
"crazygooglecid=+1<>@voice.google.com/srvenc-nrqQ46ayZknr1yPFlefF9PJfrsgA2TBA") in new stack
-- Executing [s@incoming:5] Set("Motif/+1-2927", 
"stripcrazysuffix=+1<>") in new stack
-- Executing [s@incoming:6] Set("Motif/+1-2927", "CALLERID(all)=+1<>") 
in new stack
-- Executing [s@incoming:7] Goto("Motif/+1-2927", "relay") in new stack
-- Goto (incoming,s,14)
-- Executing [s@incoming:14] Dial("Motif/+1-2927", 
"motif/51/+1@voice.google.com") in new stack
-- Called motif/51/+1@voice.google.com
-- Motif/+1@voice.google.com-d119 is proceeding passing it to 
Motif/+1-2927
[May 20 12:17:25] NOTICE[4323][C-0167]: chan_motif.c:1636 jingle_indicate: 
Don't know how to indicate condition '15'
  == Spawn extension (incoming, s, 14) exited non-zero on 'Motif/+1<>-2927'


I set xmpp debug on. As far as I can see the incoming call (xmpp port 
<45> ) terminates because of "success" just after the outgoing call ( 
xmpp <51> ) is set up. The called number rings 3 times. If it's 
answered, it's dead.




-- Called motif/<51>/+1@voice.google.com

<--- XMPP received from '<51>' --->
<
<->

<--- XMPP received from '<51>' --->
iq from="+1@voice.google.com" to="<51>@gmail.com/asterisk-xFF200C8A" type="error" id="m">http://www.google.com/session";>http://www.google.com/session/phone";>xmpp:+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==http://www.google.com/session";>xmpp:+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==
<->

<--- XMPP sent to '<51>' --->

<->

<--- XMPP received from '<51>' --->
<
<->

<--- XMPP received from '<51>' --->
iq to="<51>@gmail.com/asterisk-xFF200C8A" 
from="+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==" id="n" type="result"/>
<->

<--- XMPP sent to '<51>' --->


<->
-- Motif/+1@voice.google.com-7abc is proceeding passing it to 
Motif/+1-5e4e
[May 20 12:19:53] NOTICE[4326][C-0168]: chan_motif.c:1636 jingle_indicate: 
Don't know how to indicate condition '15'

<--- XMPP received from '<51>' --->
<
<->

<--- XMPP received from '<51>' --->
iq to="<51>@gmail.com/asterisk-xFF200C8A" 
from="+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==" id="o" type="result"/>
<->

<--- XMPP received from '<51>' --->
<
<->

<--- XMPP received from '<51>' --->
iq from="+1@voice.google.com/srvenc-N1BbqASaz0QMBuBJQvV34g==" to="<51>@gmail.com/asterisk-xFF200C8A" id="jingle:10.13.61.5-23164143:1:AB47DEE4" type="set">http://www.google.com/session";>
<->

<--- XMPP sent to '<51>' --->

<->

<--- XMPP received from '<45>' --->
<
<->

<--- XMPP received from '<45>' --->
iq from="+1@voice.google.com/srvenc-xPt/DDObueIws5jdu1V1AmeBB2R4pgkM" to="<45>@gmail.com/asterisk-xAF820CA3" id="jingle:10.68.166.37-7896298:1:898DBA27" type="set">http://www.google.com/session";>Call endedhttp://www.google.com/session/phone"/>
<->

<--- XMPP sent to '<45>' --->

<->

<--- XMPP sent to '<51>' --->

<->
  == Spawn extension (incoming, s, 14) exited non-zero on 'Motif/+1-5e4e'



The incoming motif also drops if the bridged call is SIP:


-- Called SIP//

<--- XMPP received from '<45>' --->
<
<->

<--- XMPP received from '<45>' --->
iq from="+1@voice.google.com/srvenc-jgCvKGNmqXXjKPKzhCuU3KoKurQ4+Fi1" to="<45>@gmail.com/asterisk-xAF820CA3" id="jingle:10.229.151.16-7608331:1:9E0372A9" type="set">http://www.google.com/session";>Call endedhttp://www.google.com/session/phone"/>
<->

<--- XMPP sent to '<45>' --->

<->
  == Spawn extension (incoming, s, 14) exited non-zero on 'Motif/+1



Very odd.

Any help appreciated.

sean





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Re: [asterisk-users] 11.4: motif can only handle one channel at a time?

[sean darcy]

On 05/16/2013 09:41 AM, sean darcy wrote:

I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.

motif only handles one 1 channel at a time??

sean



More:

Two different motif sections. Two different xmpp sections.
xmpp shows both connections.

sean


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[asterisk-users] 11.4: motif can only handle one channel at a time?

[sean darcy]

I have a call on gv over motif. I try to bridge it to another call over 
motif, but a different gv account, and I get congestion.


motif only handles one 1 channel at a time??

sean




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[asterisk-users] 11.4: no incoming gv/xmpp

[sean darcy]

I've set up google voice to chat with me:

Forwards calls to:

@gmail.com

and xmpp:

[general]
debug=no; Enable debugging (disabled by 
default).
autoprune=yes   ; Auto remove users from buddy 
list. Depending on your
; setup (ie, using your 
personal Gtalk account for a test)
; you might lose your contacts 
list. Default is 'no'.

autoregister=no ; Auto register users from buddy list.
;collection_nodes=yes   ; Enable support for XEP-0248 
for use with
; distributed device state. 
Default is 'no'.
;pubsub_autocreate=yes  ; Whether or not the PubSub 
server supports/is using
; auto-create for nodes.  If it 
is, we have to
; explicitly pre-create nodes 
before publishing them.

; Default is 'no'.
;auth_policy=accept ; Auto accept users' 
subscription requests (default).

; Set to deny for auto denial.
[gv](!)
type=client
serverhost=talk.google.com
secret=
priority=25
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="Not available"
timeout=5

[google1](gv)
username=@gmail.com

and xmpp show connections:

asterisk*CLI> xmpp show connections
Jabber Users and their status:
   [google1] @gmail.com - Connected


But when I call me, nothing rings through to asterisk.
i see the call on gv,

Any help appreciated.

sean


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[asterisk-users] 11.4.-rc1: new segfault in iksemel ??

[sean darcy]

I rebooted our server Fedora 17 today, and now asterisk won't start;

asterisk[1063]: segfault at 0 ip 7f117aee122d sp 7fffbc398990 
error 4 in libiksemel.so.3.1.1[7f117aed8000+d000]


iksemel is required for motif and xmpp.

I downloaded the iksemel source and rebuilt. No luck.

Any help really appreciated.

sean


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[asterisk-users] set google voice callerid as Unknown/Unavailable ?

[sean darcy]

I know you that GV won't respect CALLERID from motif, but is there a way 
have GV show Unknown on outgoing calls. I don't want to have people 
think my GV number is really my number.


sean


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Re: [asterisk-users] 11.3: how to hang up on google voice

[sean darcy]

On 03/07/2013 09:48 AM, Joshua Colp wrote:

sean darcy wrote:

Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've
hung up.

extensions.conf:

same => n,GoToIf($["${CALLERID(num)}"="office"]?email)
.
same => n(email),System(/usr/local/bin/emailme)
same => n,Answer() ; also tried without this
same => n,Hangup()


You need to Answer, Wait, send a DTMF of 1, wait a bit more, and then
hang up.


Brilliant.

 same => n(hangup),Answer()
 same => n,Wait(3)
 same => n,SendDTMF(1)
 same => n,Wait(3)
 same => n,Hangup()

Worked like a charm. It does cause gv to give a circuit busy. But that's ok.

sean


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[asterisk-users] 11.3: how to hang up on google voice

[sean darcy]

Some calls I get from google voice, I just send myself an email about 
the call and want to hangup. But I can't seem to make gv know I've hung up.


extensions.conf:

same => n,GoToIf($["${CALLERID(num)}"="office"]?email)
.
 same => n(email),System(/usr/local/bin/emailme)
 same => n,Answer() ; also tried without this
 same => n,Hangup()

console:

-- Executing [s@incoming-171:13] Answer("Motif/+1...-1b35", "") 
in new stack
-- Executing [s@incoming-171:14] Hangup("Motif/+1..-1b35", 
"") in new stack
  == Spawn extension (incoming-171, s, 14) exited non-zero on 
'Motif/+1..-1b35'


but the calling phone keeps ringing until the google voice attendant 
comes on.


OTOH, if I dial an extension, hanging up works fine.

So how do I get gv to recognize a hangup?

sean


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Re: [asterisk-users] how to join calls - not barge?

[sean darcy]

On 02/13/2013 09:39 AM, Matthew Jordan wrote:

On 02/12/2013 06:48 PM, sean darcy wrote:

On 02/12/2013 05:37 PM, Rusty Newton wrote:

 Original Message -

From: "sean darcy" 




Can I throw A and B into a confbridge and then add C?  Create a new
channel that grabs the A <-> B channel? Or is there a more straight
forward way to do this?



The Asterisk Definitive guide has some good info on what you can do
with ConfBridge. That might work for you. See "Advanced
Conferencing"[1] and "Conferencing with ConfBridge()"[2]

Also, there is the Shared Line Appearance stuff in Asterisk[3]. That's
a bit more confusing, but may help you as well. I'd recommend playing
with both to really see if they work for your needs.


[1] http://ofps.oreilly.com/titles/9781449332426/asterisk-SysAdmin.html
[2]
http://ofps.oreilly.com/titles/9781449332426/asterisk-DP-Deeper.html#confbridgeConferencing

[3]
http://ofps.oreilly.com/titles/9781449332426/asterisk-DeviceStates.html#SLA




Thanks.

confBridge could work here, but how do I throw an existing bridge into
the confbridge? That is, if A <-> B exists, how do I trigger the entry
into confbridge. Once there, it's pretty easy to see how C would join.


Assuming you have something like:

exten => s,1,Dial(SIP/B)

Such that SIP/A and SIP/B are in a bridge formed by the Dial
application, you would first need to break that bridge and move each
channel into the ConfBridge. One possible approach without resorting to
AMI would be something like this:

; Entry point for SIP/C

exten => c,1,ChannelRedirect(SIP/A,default,goto_confbridge,1)
same => n,Goto(default,goto_confbridge,1)

; Entry point for SIP/A

exten => a,1,Dial(SIP/B,,F(default,goto_confbridge,1)

; Extension that drops any channel into the multi-party bridge

exten => goto_confbridge,1,NoOp()
same => n,ConfBridge(1234)

This is obviously an example - you'd have to get the actual channel name
for SIP/A; you'd want to check whether or not the caller was hung up
when the called party gets shunted off into the goto_confbridge
extension (since it will happen unilaterally when the caller is removed
from the bridge), etc.


maybe EVERY call is done with confbridge. Would that cause some other
problem?


That would work, but ConfBridge doesn't have support for some two-party
concepts, such as Hold. It depends on your use cases whether or not that
would be acceptable.


I don't necessarily have DAHDI, so the SLA stuff wouldn't work. Just as
well, since my head hurt reading about it.



As an aside, we are working in Asterisk 12 to make this kind of scenario
much easier. The work being done now should allow you to seamlessly
transition from two-party to multi-party bridges (and back) without
having to do dialplan shenanigans.

Matt


Thanks. Very helpful.

sean


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[asterisk-users] motif - gv not working today?

[sean darcy]

I had motif working two days ago but now:

Executing [1171@internal:1] Dial("DAHDI/1-1", "Motif/1171") in new stack
[Feb 12 20:56:18] ERROR[7794][C-0001]: chan_motif.c:1762 
jingle_request: Unable to determine endpoint name and target.


motif.conf:

[11XX](!)
transport=google-v1
disallow=all
allow=ulaw

[1171](11XX)
context=incoming-171
connection=gmail1171

xmpp.conf:

[gmail11XX](!)
type=client
serverhost=talk.google.com
priority=1
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="Not available"
timeout=5

[gmail1171](gmail11XX)
username=gmail1...@gmail.com
secret=gmailsecret

cli:
xmpp show connections
Jabber Users and their status:
[gmail1171] gmail1...@gmail.com - Connected

Have I messed up, or is google voice just not working today?

sean


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Re: [asterisk-users] how to join calls - not barge?

[sean darcy]

On 02/12/2013 05:37 PM, Rusty Newton wrote:

 Original Message -

From: "sean darcy" 




Can I throw A and B into a confbridge and then add C?  Create a new
channel that grabs the A <-> B channel? Or is there a more straight
forward way to do this?



The Asterisk Definitive guide has some good info on what you can do with ConfBridge. That might 
work for you. See "Advanced Conferencing"[1] and "Conferencing with 
ConfBridge()"[2]

Also, there is the Shared Line Appearance stuff in Asterisk[3]. That's a bit 
more confusing, but may help you as well. I'd recommend playing with both to 
really see if they work for your needs.


[1] http://ofps.oreilly.com/titles/9781449332426/asterisk-SysAdmin.html
[2] 
http://ofps.oreilly.com/titles/9781449332426/asterisk-DP-Deeper.html#confbridgeConferencing
[3] http://ofps.oreilly.com/titles/9781449332426/asterisk-DeviceStates.html#SLA



Thanks.

confBridge could work here, but how do I throw an existing bridge into 
the confbridge? That is, if A <-> B exists, how do I trigger the entry 
into confbridge. Once there, it's pretty easy to see how C would join.


maybe EVERY call is done with confbridge. Would that cause some other 
problem?


I don't necessarily have DAHDI, so the SLA stuff wouldn't work. Just as 
well, since my head hurt reading about it.


sean


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[asterisk-users] how to join calls - not barge?

[sean darcy]

I'd like to have an extension "join" a call. That is, C can join A and 
B, just as if it were an analog extension phone.


ChanSpy works, sort of. The problem is that once A or B hangs up, the 
channel is gone. With an analog extension, C would remain connected with 
B if A hung up.


Can I throw A and B into a confbridge and then add C?  Create a new 
channel that grabs the A <-> B channel? Or is there a more straight 
forward way to do this?


sean


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Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

[sean darcy]

On 12/11/2012 10:12 PM, Mitul Limbani wrote:

snom m9 dect ip


But it's 2-3 x the price!

sean


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Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

[sean darcy]

On 12/11/2012 04:37 PM, Roy Abshire wrote:

That is true about the A580.

I don't like the interface much to check messages.

Besides that every time I go to dial a number...it always uses the first
digit pressed to go into phone mode..so I have to press the first digit
twice...

I would test other phones but it's for home and I can't fork over $$ to
try them all out

I have tested some Nokia cell phones, the N97, N900, and E71 and the E71
and N900 worked well.  I didn't like the N97.

Co-op Vacation Rentals
www.coopvr.com
15218 Summit Ave
Suite #300-354
Fontana, CA 92336
Phone/Fax (855) 760-COOP (2667)

On 12/11/2012 12:52 PM, Pete Mundy wrote:

One thing I dislike about the A580H is that the handset always says 'You have 
new messages' if I've missed a call. It wouldn't bug me if it said 'missed 
call' but it tells me I have new messages and even lights up a red LED under a 
button with a picture of an envelope on it.

I'm about to test an A510IP and an A610IP to compare against the A580. Fingers 
crossed neither of them has that issue, because the Gigaset phone is a pretty 
good phone other than that, and the difficulty doing a (blind) transfer, as 
referred to by the OP.

Pete


On 12/12/2012, at 8:57 AM, Roy Abshire  wrote:


I've been using the Gigaset A580 Base and A58H Phone for about 3 years now.  
Never gave me problems. The call Quality is excellent!
I only have 1 handset connected to the Base but I want more. I bought a Linksys 
WIP330 as a 2nd phone to try out and that works just as good without a base 
unit.

The A580 Base supports up to 6 handsets.

I have 6 Incoming VOIP Numbers using separate SIP accounts pointed to 1 Handset 
but you can point each SIP to separate handsets.

The call goes to the first phone that picks up.  When on a call, picking up 
another phone makes a separate call and does not conference.  I don't use 
conference yet but I know you have to put the call on hold or something.

The thing I don't like about the A580 and might be the same on all of them is 
that you can only specify 1 Sip Account for making outgoing calls.  In other 
words, all 6 phones would use the same caller id out, but I wanted to be able 
to choose that because I have a business number and number for each person in 
our household.  In order to use a different Caller ID (SIP Account) for making 
outgoing calls I added a extension to my Dial Plan and before making outgoing 
calls I press *1-6 before the number.

I'm going to try adding more handsets that are compatible.  I want the SL78H 
but they are so expensive for just home everyday use.

Make sure you check the compatibility page here before buying handsets.

http://gigaset.com/us/en/cms/PageCustomerServicesCompatibility.html



Some stupid questions:

I understand the A510 allows 2 sip calls. Let's say you've registered 
the base with asterisk. A uses handset 1 to call out over sip. B picks 
up handset 2. Does B hear a dial tone? Can B dial out over the asterisk 
server?


Or do you need two registrations with asterisk? In which case, is 
handset 2 always tied to the second registration?


sean


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[asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

[sean darcy]

I have an asterisk server at home. I'm looking to replace my internal 
phones with sip cordless (DECT) phones. I'm now looking at the Siemens 
A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base 
($80) and DP710 handset ($45).


The Siemens has a feature were I can also use a PSTN landline, but I not 
sure that's a great benefit.


Has anybody tried these phones? I assume they both integrate with 
asterisk since they both are sip.


I'm leaning towards Grandview. Seems to be easier to transfer calls.

A questions, I'm on a call with one handset, can another person pick up 
a second handset to make another call? can that person join the first 
call? In other words, do I need two base stations to make two calls?


Any suggestions? Thoughts?

sean


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[asterisk-users] 11.0: how to get remote commands to show on cli?

[sean darcy]

I'd like to see on cli what happens on executing remote commands. For 
instance:


asterisk -rx "originate 
Motif/gvoice/12026668...@voice.google.com,,rL(5000)) extension  s@default"


Now I get on cli, verbose 10:

-- Remote UNIX connection
-- Remote UNIX connection disconnected

Any way to see call progress for remote commands? Something is going 
wrong with this command, although it works fine from the dialplan.


sean


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Re: [asterisk-users] 11.0.1: more sip registry woes

[sean darcy]

On 11/06/2012 09:45 PM, Michael L. Young wrote:

- Original Message -

From: "sean darcy" 
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 6, 2012 7:51:04 PM
Subject: [asterisk-users] 11.0.1: more sip registry woes

Upgrade to 11. This worked on 10.X.X

sip.conf:

register=>:@nyc.teliax.net

telnet  nyc.teliax.net 5060
Trying 8.14.120.23...
Connected to nyc.teliax.net.
Escape character is '^]'.

sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
nyc.teliax.net:5060 N  
  120
Unregistered
1 SIP registrations.


Nothing on the cli to show any problems.

teliax says no problems on their end.

In 10 if I wasn't registered I got lots on registration failed
messages.

Added this to sip.conf:

registertimeout=20 ; retry registration calls every 20
seconds (default)
registerattempts=0   ; if 0 try forever

which is supposed to be the default anyhow.


I am registered without any problems to nyc.teliax.net.

How is your peer definition set in sip.conf?  Try turning verbosity up on the console and 
also "set sip debug" on for your peer in order to see the communication between 
your server and Teliax.  Hopefully, that will provide some clues as to why you are not 
registering.

Michael
(elguero)



Are you running 11.0.1?

sean


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[asterisk-users] 11.0.1: more sip registry woes

[sean darcy]

Upgrade to 11. This worked on 10.X.X

sip.conf:

register=>:@nyc.teliax.net

telnet  nyc.teliax.net 5060
Trying 8.14.120.23...
Connected to nyc.teliax.net.
Escape character is '^]'.

sip show registry
Hostdnsmgr Username   Refresh 
StateReg.Time
nyc.teliax.net:5060 N120 
Unregistered

1 SIP registrations.


Nothing on the cli to show any problems.

teliax says no problems on their end.

In 10 if I wasn't registered I got lots on registration failed messages.

Added this to sip.conf:

registertimeout=20 ; retry registration calls every 20 
seconds (default)

registerattempts=0   ; if 0 try forever

which is supposed to be the default anyhow.

sean


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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?

[Sean Darcy]

On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D  wrote:
> Hello,
>
> How do I use the asterisk application 'Transfer' to transfer a SIP
> call from one asterisk to another?
>
> I have the following scenario. I have two asterisk servers S1 and S2.
> There is a third asterisk server C1 which registers as a peer to S1.
> From C1, I dial into S1 using 'Dial' command. What I want to do is,
> use the Transfer command in S1 and transfer the call to S2.
>
> Dialplan on S1
> [test_context]
> exten => _X.,1,Transfer(SIP/${EXTEN}@IP_of_S2)
> exten => _X.,n,NoOp(${TRANSFERSTATUS})
> exten => _X.,n,Hangup
>
> Dialplan on S2
> [default]
> exten => _X.,1,Playback(somemsg)
> exten => _X.,n,Hangup
>
> [test_context]
> exten => _X.,1,Answer
> exten => _X.,n,Playback(msg)
> exten => _X.,n,Hangup
>
> The context for the SIP peer C1 is defined as 'test_context' in S1 and S2.
>
> In C1, I have set 'promiscredir = yes' in sip.conf.
>
> When I dial from C1, the call is successfully transferred to S1 (I get
> TRANSFERSTATUS as SUCCESS and I can see C1 trying to send the call to
> S2). But the call does not get authenticated on S2 and goes into
> default context instead of 'test_context'. How can I transfer the call
> such that S2 authenticates the call and sends it to the required
> context?
>
> Thanks
>

What happens when you dial into S2 from outside?

Did you set a context in sip.conf on S2?

sean

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[asterisk-users] how does GOTO_ON_BLINDXFR work?

[sean darcy]

10.9.0. I'm trying to have a setup where hitting # sends the called 
party to the confbridge. I've set  GOTO_ON_BLINDXFR:


CLI> dialplan show globals
.
GOTO_ON_BLINDXFR=tel-incoming^confbridge^1

(Also tried tel-incoming,confbridge,1 and using | )

but it doesn't work:

Dial("DAHDI/1-1", "DAHDI/4/xxxyyy,,tT") in new stack
-- Called DAHDI/4/xxxyyy
...
-- DAHDI/4-1 answered DAHDI/1-1
-- Started music on hold, class 'default', on DAHDI/4-1
--  Playing 'pbx-transfer.ulaw' (language 'en')
[Oct  9 18:10:33] WARNING[28164]: features.c:2367 builtin_blindtransfer: 
No digits dialed.

--  Playing 'pbx-invalid.ulaw' (language 'en')

sean


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Re: [asterisk-users] Asterisk 11.0.0-rc1 Now Available!

[sean darcy]

On 10/08/2012 05:15 PM, Asterisk Development Team wrote:

The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 11.0.0.  This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the
Asterisk 11 testing process.  Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira.  It is also very useful to see
successful test reports.  Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing channel
on IRC to work together in testing the many parts of Asterisk.

Asterisk 11 is the next major release series of Asterisk.  It will be a Long
Term Support (LTS) release, similar to Asterisk 1.8.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 11, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

A short list of new features includes:

* A new channel driver named chan_motif has been added which provides support
   for Google Talk and Jingle in a single channel driver.  This new channel
   driver includes support for both audio and video, RFC2833 DTMF, all codecs
   supported by Asterisk, hold, unhold, and ringing notification. It is also
   compliant with the current Jingle specification, current Google Jingle
   specification, and the original Google Talk protocol.

* Support for the WebSocket transport for chan_sip.

* SIP peers can now be configured to support negotiation of ICE candidates.

* The app_page application now no longer depends on DAHDI or app_meetme. It
   has been re-architected to use app_confbridge internally.

* Hangup handlers can be attached to channels using the CHANNEL() function.
   Hangup handlers will run when the channel is hung up similar to the h
   extension; however, unlike an h extension, a hangup handler is associated 
with
   the actual channel and will execute anytime that channel is hung up,
   regardless of where it is in the dialplan.

* Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
   allows you to execute a dialplan subroutine on a channel before a call is
   placed but after the application performing a dial action is invoked. This
   means that the handlers are executed after the creation of the callee
   channels, but before any actions have been taken to actually dial the callee
   channels.

* Log messages can now be easily associated with a certain call by looking at
   a new unique identifier, "Call Id".  Call ids are attached to log messages 
for
   just about any case where it can be determined that the message is related
   to a particular call.

* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
   Asterisk. Unlike traditional ACLs defined in specific module configuration
   files, Named ACLs can be shared across multiple modules.

* The Hangup Cause family of functions and dialplan applications allow for
   inspection of the hangup cause codes for each channel involved in a call.
   This allows a dialplan writer to determine, for each channel, who hung up and
   for what reason(s).

* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
   lets you set some of the configuration options from the general section
   of features.conf on a per-channel basis. FEATUREMAP() lets you customize
   the key sequence used to activate built-in features, such as blindxfer,
   and automon.

* Support for DTLS-SRTP in chan_sip.

* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
   and callgroups to be defined for several channel drivers.

* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

For a full list of changes in the current release, please see the ChangeLog.

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1

Thank you for your continued support of Asterisk!




Thanks for all the great work.

We've started using the silk codec a lot for phone app voip. We've found 
it the most effective low bit rate (16K) codec. Could we get a release 
11 version of the silk codec in 
http://downloads.digium.com/pub/telephony/codec_silk/  ?


That way we could start messing with RC 1.

sean


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Re: [asterisk-users] Fax for Asterisk success rates?

[sean darcy]

On 10/09/2012 07:40 AM, Steve Underwood wrote:

On 10/09/2012 12:28 AM, Brett Lehrer wrote:

How many fax and voice calls (which codecs for tha latter ones ?) are on
average using your DSL line ?
1. Previously, I experienced failures during the process of converting
incoming PDF documents into ready-to-send fax image files while the
reverse
process (from a fax file into a PDF or whatever document) never failed.
I would be curious to check if a greater failure rate for outbound
faxing
(greater than inbound faxing failure rate) could simply comes from image
processing, before any transmission.
2. Though your DSL line may have enough bandwidth from your location
to its
DSLAM, chances are packets are dropped or delivered too late for T.38
faxing.
An interesting test would be to use an Asterisk PBX hosted somewhere at
"close range" from netVortex fax gateways : that would remove most
networking issues out of the equation.

I'll have to look more closely into what codecs we traditionally use,
but g.722 up and ulaw down is common.  Generally don't have more than
2-3 calls active at once.  At most, 5, and that's a rarity.  Record
for fax is 4 simultaneous send/receive, but typically just 1, maybe
2.  I imagine that's encroaching on the upper limits of the 768 kbps
upspeed.  I've wondered about how lag might impact the problem but I
just don't know how I'd go about testing it properly without spending
a bunch of money on hosting.

I do my PDF -> TIFF conversion on another machine with ghostscript.
Here's the line:

gs -q -dNOPAUSE -dBATCH -dSAFER -sDEVICE=tiffg4
-sOutputFile= -f 

I changed from tiffg3 to tiffg4 because the filesize got cut in half
assuming that the less time spent transmitting, the less chance there
was to run into a problem that might stop the fax.  However, most
failures that I've looked at seem to occur immediately or fail to
connect at all, rather than get cut off due to a hiccup in the
connection.

Brett Lehrer


A FAX can only be sent in ECM mode when using tiffg4 format. It will
have to be recoded into tiffg3 format if ECM is inhibited, which it far
too often is. On the other hand, if you are using ECM any decent FAX
system (e.g. spandsp) will recode into tiffg4, and really good ones
(e.g. the very latest spandsp) may recoed into T.85/JBIG, for faster
transmission times. Digium don't seem to specify what FFA does in this
area.

Steve



A little puzzled. Do you mean:

1. tiffg4 encoded fax will(might?) fail if ECM is inhibited at either 
send or receive.


2. tiffg3 will work if ECM is inhibited.

3. If ECM is not inhibited, any decent fax system, will reencode tiffg3 
to tiffg4.


Therefore we should encode to tiffg3 and let spandsp determine if it 
should be rencoded to tiffg4 (or T.85/JBIG)?


sean


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Re: [asterisk-users] 10.9.0-rc1 : Help with GoSubIf Parsing

[sean darcy]

So here's what I used:
$['x${CALLERID(num)}'='x2024324321']

And that worked!
On 10/05/2012 08:28 AM, Richard Kenner wrote:

  I'm getting a parsing error with the folllowing:

  same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
  {thisexten}):)

  WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():  syntax
  error: syntax error, unexpected '=', expecting $end; Input:
   = 2024324321

  I've tried with and without spaces the = sign. Same  Result. I've
  counted my parens and braces.


If there *is* a caller-ID, it should work without spaces.  But not if
there isn't.  The proper test is:

   $[x${CALLERID(num)}=x2024324321]

And this only works if you're *sure* that it'll be just numbers or blank.
Otherwise, use quotes on both sides.



So here's what I used:
$['x${CALLERID(num)}'='x2024324321']

and that worked.

Thanks for the help!

sean


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[asterisk-users] 10.9.0-rc1 : Help with GoSubIf Parsing

[sean darcy]

I'm getting a parsing error with the folllowing:

same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1(${thisexten}):)

WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():  syntax 
error: syntax error, unexpected '=', expecting $end; Input:

 = 2024324321
 ^
[Oct  4 21:53:35] WARNING[11356]: ast_expr2.fl:472 ast_yyerror: If you 
have questions, please refer to 
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [2124531143@from-teliax-sip:3] GosubIf("DAHDI/1-1", 
"?other,1(2124531143):") in new stack


I've tried with and without spaces the = sign. Same  Result. I've 
counted my parens and braces.


Any help really appreciated!

sean


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Re: [asterisk-users] is silk included in asterisk 11?

[sean darcy]

On 09/25/2012 11:49 AM, Jonathan Rose wrote:

Jonathan Rose wrote:


Sean Darcy wrote:


I'm building asterisk 11 beta 2. I've been using silk a lot. I
don't
see
silk listed in menuselect as a codec. But I also don't see an
asterisk
11 silk codec on
http://downloads.digium.com/pub/telephony/codec_silk.

Do we use the asterisk 10 codec_silk.so ?

sean


https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats

See the bottom blurb in particular --

* The SILK licensing, like the licensing for Polycom's Siren 7
G.722.1 and Siren 14 G.722.1C codecs, requires that the distribution
of binary codec modules that can be used by Asterisk. To download
the SILK codec module for Asterisk, browse to
http://downloads.digium.com/pub/telephony/codec_silk/unsupported/asterisk-10.0/
and drop the untar'd .so file into /usr/lib/asterisk/modules and
issue an Asterisk restart, or simply load the codec module from the
Asterisk CLI

Actually, it seems that link provided is a little incorrect and the
actual codec modules are available here:
http://downloads.digium.com/pub/telephony/codec_silk/asterisk-10.0/

These modules are actually compiled for Asterisk 10 rather than 11,
but the architecture for codec translators remains largely
unchanged, so I would guess it'll probably work. Probably.


I've been informed that this probably won't actually work due to other
changes that were probably made in resources that codecs tend to rely
on, so take the above with a grain of salt. However, I'm also hearing
that a SILK codec for 11 may be released once 11 is out of beta.



OK. I'll wait on 11 for the silk codec.

Thanks for the quick response.

sean


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[asterisk-users] is silk included in asterisk 11?

[sean darcy]

I'm building asterisk 11 beta 2. I've been using silk a lot. I don't see 
silk listed in menuselect as a codec. But I also don't see an asterisk 
11 silk codec on http://downloads.digium.com/pub/telephony/codec_silk.


Do we use the asterisk 10 codec_silk.so ?

sean


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Re: [asterisk-users] 10.6.0-rc2: tmp full of core.PBX

[sean darcy]

On 07/10/2012 11:44 AM, Matthew Jordan wrote:

- Original Message -

From: "sean darcy" 
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 10, 2012 10:42:20 AM
Subject: [asterisk-users] 10.6.0-rc2: tmp full of core.PBX

I've installed 10.6.0-rc2 on two machines. On one of the machines
(but
not the other) /tmp gets filled with:

...
-rw---. 1 asterisk asterisk 53661696 Jul  7 23:46
core.PBX-2012-07-07T23:46:10-0400
-rw---. 1 asterisk asterisk 53891072 Jul  7 23:48
core.PBX-2012-07-07T23:48:55-0400
-rw---. 1 asterisk asterisk 53469184 Jul  7 23:53
core.PBX-2012-07-07T23:53:00-0400
-rw---. 1 asterisk asterisk 53739520 Jul  7 23:58
core.PBX-2012-07-07T23:58:25-0400
...

and finally fills up all the space.

grep -v ';'  /etc/asterisk/logger.conf

[general]
[logfiles]
console => notice,warning,error
messages => notice,warning,error

Any clue what to look for?

sean



You'll need to provide a backtrace using the instructions below:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

As soon as you have the information, please open an issue in JIRA.

Thanks!

--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



Nothing in /tmp since I erased all the core* files this morning. If it 
starts again, I'll do a backtrace.


sean



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[asterisk-users] 10.6.0-rc2: tmp full of core.PBX

[sean darcy]

I've installed 10.6.0-rc2 on two machines. On one of the machines (but 
not the other) /tmp gets filled with:


...
-rw---. 1 asterisk asterisk 53661696 Jul  7 23:46 
core.PBX-2012-07-07T23:46:10-0400
-rw---. 1 asterisk asterisk 53891072 Jul  7 23:48 
core.PBX-2012-07-07T23:48:55-0400
-rw---. 1 asterisk asterisk 53469184 Jul  7 23:53 
core.PBX-2012-07-07T23:53:00-0400
-rw---. 1 asterisk asterisk 53739520 Jul  7 23:58 
core.PBX-2012-07-07T23:58:25-0400

...

and finally fills up all the space.

grep -v ';'  /etc/asterisk/logger.conf

[general]
[logfiles]
console => notice,warning,error
messages => notice,warning,error

Any clue what to look for?

sean


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Re: [asterisk-users] 10.5.0: channel name inserted as callerid number ??

[sean darcy]

Tried that. That gave me the dreaded username mismatch:

chan_sip.c:14807 check_auth: username mismatch, have , 
digest has 
NOTICE[1738]: chan_sip.c:23250 handle_request_invite: Failed to 
authenticate device


sean

On 06/20/2012 04:26 PM, Warren Selby wrote:

On Wed, Jun 20, 2012 at 3:21 PM, sean darcy mailto:seandar...@gmail.com>> wrote:

[home_outgoing]
type=friend
transport=tcp
secret=<>
fromuser=office_incoming
host=dynamic
disallow=all
allow=ulaw



It's because you're using "fromuser" as your username setting.  This
will overwrite your CallerID settings.  Instead try using "defaultuser".


--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>



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[asterisk-users] 10.5.0: channel name inserted as callerid number ??

[sean darcy]

I'm trying to set the callerid on a SIP call:

  same=n,Set(CALLERID(all)="test"<2023214321>)
  same=n,Dial(SIP/home_outgoing/150)

-- Executing [202454@from-test-sip:3] Set("SIP/sip-test-0019", 
"CALLERID(all)="test"<2023214321>") in new stack
-- Executing [202454@from-test-sip:4] Dial("SIP/sip-test-0019", 
"SIP/home_outgoing/150") in new stack


[home_outgoing]
type=friend
transport=tcp
secret=<>
fromuser=office_incoming
host=dynamic
disallow=all
allow=ulaw

But the answering box inserts the channel name as the callerid number, 
though the callerid name is correct:


[from_home]
exten => 150,1,NoOp(${CALLERID(all)})

-- Executing [150@from_home:1] NoOp("SIP/office_incoming-0043", 
""test" ") in new stack


[office_incoming]
type=user
transport=tcp
context=from_home
dtmfmode=rfc2833
disallow=all
allow=ulaw

Puzzled. Any help appreciated.

sean


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[asterisk-users] local vs remote bridging: no audio

[sean darcy]

Calling into 10.5.0-rc2 from a pstn did provider, I get no audio:

-- Executing [111@from-teliax:1] Dial("SIP/teliax-0010", 
"SIP/office2/+1") in new stack

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/office2/+1
-- SIP/office2-0011 answered SIP/teliax-0010
-- Locally bridging SIP/teliax-0010 and SIP/office2-0011

But if I call in over sip from outside with the same number and channel 
all works fine:


-- Executing [111@from_11hidden:1] 
Dial("SIP/office_incoming-0012", "SIP/office2/+1") in 
new stack

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/office2/+1
-- SIP/office2-0013 answered SIP/office_incoming-0012
-- Remotely bridging SIP/office_incoming-0012 and 
SIP/teliax2-0013


The only difference I can see is Locally vs. Remotely bridging.

sip.conf

nat=yes
directmedia=nonat

Any suggestions appreciated.

sean


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Re: [asterisk-users] DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available

[sean darcy]

On 04/21/2012 12:00 PM, Shaun Ruffell wrote:

On Sat, Apr 21, 2012 at 10:26:30AM -0400, sean darcy wrote:

On 04/20/2012 02:05 PM, Asterisk Development Team wrote:

The Asterisk Development Team has announced the releases of:
   DAHDI-Linux 2.6.1
   DAHDI-Linux 2.5.1
   DAHDI-Tools 2.6.1
   DAHDI-Tools 2.5.1
   DAHDI-Linux-Complete 2.6.1+2.6.1
   DAHDI-Linux-Complete 2.5.1+2.5.1





How do we build dahdi with oslec? At least on Fedora 16, the kernel
sources no longer include the oslec source in staging/echo:

ls -l /usr/src/kernels/3.3.2-1.fc16.x86_64/drivers/staging/echo
total 8
-rw-r--r--. 1 root root 251 Apr 13 20:49 Kconfig
-rw-r--r--. 1 root root  29 Apr 13 20:49 Makefile


It's still in the staging tree in the mainline kernel [1] so must be
something Fedora 16 specific.

[1] 
http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=tree;f=drivers/staging/echo



When I did copy over oslec files from a previous build, I got this warning:

...
   Building modules, stage 2.
   MODPOST 32 modules
WARNING: "oslec_create" 
[/home/asterisk/build/dahdi/dahdi-linux-2.6.1/drivers/dahdi/dahdi_echocan_oslec.ko]
undefined!
WARNING: "oslec_free" 
[/home/asterisk/build/dahdi/dahdi-linux-2.6.1/drivers/dahdi/dahdi_echocan_oslec.ko]
undefined!
WARNING: "oslec_update" 
[/home/asterisk/build/dahdi/dahdi-linux-2.6.1/drivers/dahdi/dahdi_echocan_oslec.ko]
undefined!
...


I don't have time to figure out what changed in the kernel build
system which breaks this but it looks like the "Makefile" is no
longer processed by default.

In drivers/dahdi/Kbuild change:
obj-m += ../staging/echo/

  to:

obj-m += ../staging/echo/echo.o

And you should be back in business.

Cheers,
Shaun


Worked For Me.

Thanks for the quick response.

sean



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Re: [asterisk-users] DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available

[sean darcy]

On 04/20/2012 02:05 PM, Asterisk Development Team wrote:

The Asterisk Development Team has announced the releases of:
   DAHDI-Linux 2.6.1
   DAHDI-Linux 2.5.1
   DAHDI-Tools 2.6.1
   DAHDI-Tools 2.5.1
   DAHDI-Linux-Complete 2.6.1+2.6.1
   DAHDI-Linux-Complete 2.5.1+2.5.1

These releases are available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

2.6.1 and 2.5.1 are bugfix releases of which the most noteable changes are:

- Fix for Digium dual and quadspan cards in E1 mode when used with a hardware
   echocanceler that was introduced in 2.6.0.
- Fix for intermittent failure to decode FSK caller ID on Digium voicebus
   analog cards introduced in 2.6.0.
- Support for Linux kernel versions up to 3.4.

Issues closed in these releases:
 DAHLIN-275: E1 spans have noise on some alternative channels when VPM is 
active
 DAHLIN-274: dahdi_dummy failes to compile
 DAHLIN-283: Disable Active State Power Management on PCIe links for DAHDI 
devices.
 DAHLIN-280: dahdi_dynamic_eth(ethmf,loc)
 DAHLIN-286: DAHDI driver wctdm24xxp does not compile with GCC 3.4.4
 DAHLIN-279: dahdi will not compile with CONFIG_DAHDI_ECHOCAN_PROCESS_TX
 DAHLIN-278: dahdi will not compile with CONFIG_DAHDI_NET
 DAHLIN-185: Dahdi dummy includes time.h, should be timer.h for low-res 
timer
 DAHLIN-288: compilation error when CONFIG_DAHDI_WATCHDOG is defined
   And in the 2.5.1 release only:
 DAHLIN-272: No PCM on a TDM410 FXS module since r10167

The DAHDI-Linux shortlog of changes since 2.6.0:

   Mike Sinkovsky (1):
 dahdi: Fix compilation when CONFIG_DAHDI_WATCHDOG is defined.

   Oron Peled (9):
 xpp: bugfix: fix bad refcount
 xpp: Don't deactivate XPDs on unregistration
 xpp: handle failures during dahdi_register_device()
 xpp: reset Astribank SPI busses
 xpp: FXS: better power-down to lower noise
 A parent-less device should not crash dahdi
 remove a duplicate dev_set_name()
 xpp: FXS: atomic vbat_h power handling
 xpp: FXS: added a 'lower_ringing_noise' parameter

   Shaun Ruffell (30):
 wctdm24xxp: FXS on-hook transmission timer incorrect.
 wct4xxp: VPM module creates noise on alternate channels on E1 spans.
 wctdm24xxp: Shorten RINGOFF debounce interval from 512ms to 128ms.
 xpp: Use 'bool' type for boolean module parameters on kernel 
versions>= 2.6.31.
 xpp: '%d' ->  '%lu' when displaying module_refcount on kernel 
versions>= 3.3
 dahdi_dummy: Fix compilation since dahdi-linux 2.6.0.
 dahdi: Add dahdi_pci_disable_link_state for kernel<  2.6.25.
 wct4xxp: __t4_frame_in and __t4_framer_out slowdowns.
 wct4xxp: Add compile-time option to disable ASPM for PCIe devices.
 wcte12xp, wctdm24xxp: Add compile-time option to disable ASPM for PCIe 
devices.
 dahdi: Update dev_set_name / dev_name for RHEL 5.6+.
 dahdi_dynamic_eth: Move tx packet flushing to process context.
 dahdi_dynamic: Since dynamic devices are 'parentless' we must name 
them.
 dahdi_dynamic_eth: Prevent crash is packet arrives before span is 
fully configured.
 dahdi_dynamic_eth: Fix compilation on kernels<  2.6.22.
 wct4xxp: Disable all interrupts explicitly in interrupt handler.
 wct4xxp: Trivial formatting changes around request_irq.
 wctdm24xxp: Remove forward declaration of inline for GCC 3.4.4
 wctdm24xxp, wcte12xp: Allow VPMOCT032 firmware to be compiled into 
driver.
 dahdi_dynamic: Do not call into dahdi_dynamic without holding 
reference.
 dahdi_dynamic: Remove calls to __module_get().
 dahdi_dynamic: Close race on unload if red alarm timer was running 
when unloaded.
 dahdi_dynamic_eth: Make ztdeth_exit() symetrical with ztdeth_init() 
and fix race on unload.
 dahdi_dynamic_loc: Change and check the dyn->pvt pointer under lock.
 dahdi: Fix compilation when CONFIG_DAHDI_ECHOCAN_PROCESS_TX is defined.
 dahdi: Fix compilation when CONFIG_DAHDI_NET is defined.
 dahdi_dummy: Include timer.h instead of time.h
 wcb4xxp: Remove asm/system.h include.
 wcte12xp, wctdm24xxp, wct4xxp: Print warning about potential GPL 
violation w/HOTPLUG_FIRMWARE=no.
 xpp: Fix compilation when CONFIG_DAHDI_WATCHDOG is defined.

   Tzafrir Cohen (8):
 Build OSLEC EC if in the tree
 Astribank I firmwares rev. 7107
 USB_RECOV.hex: recovering from xpp hardware issues
 xpp: USB_FW rev 10401: minor 6FXS/2FXO caps issue
 xpp: firmwares to support E-Main 4
 xpp: firmwares: useless 0x1A at EOF
 FPGA_1161.201.hex rev 10532: fix reset of XR1000
 FPGA_1161.201.hex rev 10545: fix reset of XR1000

The DAHDI-Linux diffstat fro

Re: [asterisk-users] 10.3 : sip loses registration ?

[sean darcy]

On 04/16/2012 04:09 PM, Larry Moore wrote:

I have experienced this issue with a provider with Asterisk 1.2, 1.6 & 1.8.

I never got to the root cause of the problem however it used to occur
quite frequently, now it appear to occur once every month or two -
haven't seen it occur for a while now but then I have been incrementally
updating my version of asterisk, currently 1.8.11.0

The preceding events I observed was that there would be a timeout
communicating with the peer followed by retry attempts and finally a
message reporting "Wrong password", this is the point at which
registration attempts stopped despite the value in sip.conf being set to 0.

As per your observations a 'sip reload' gets things going again.

When the problem was occurring within a 24-hour period I set up an
SPA-942 phone to register to the service and captured packets between
them, I don't recall seeing any issues over a period of a few days with
the SPA phone hence was baffled by this phenomenon and have been since.

I was considering writing a script to check for the "No Authentications"
status and to then issue a 'sip reload' but as the problem is rarely
seen now I haven't had to do this.

My suspicion to the cause of the problem is that the authentication
database at the VSP may have been offline momentarily hence why the
response of a wrong password, I wasn't convinced of this as the packet
capture of the SPA-942 did not reveal any authentication errors.

Cheers,

Larry.



On 16/04/2012 10:26 PM, sean darcy wrote:

We found this morning we had no SIP connection to another site. sip
show registry on the main site gave "no authentication". sip show
peers on the other site showed the peer unspecified.

The odd part about this: doing sip reload on the main site made it all
work. Nothing else was changed.

Main site:

sip show registry

SFO:5060 N sip_outgoin 105 No Authentication Sat, 14 Apr 2012 14:48:15
4 SIP registrations.
..

PBX*CLI> sip reload
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': == Found
== Parsing '/etc/asterisk/users.conf': == Found
== Using SIP TOS bits 96
== Using SIP CoS mark 3
== Parsing '/etc/asterisk/sip_notify.conf': == Found
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

On other site:

sip show peers
.
sip_outgoing/s (Unspecified) D N 0 Unmonitored
..
13 sip peers [Monitored: 3 online, 7 offline Unmonitored: 2 online, 1
offline]

result of sip reload on main site:
-- Registered SIP 'sip_outgoing' at xxx.yyy.zzz.aaa:9345


How do I/can I set the main site to retry a registration? I've now
changed sip.conf to add:

registertimeout=20
registerattempts=0 ;Default is 0 tries, continue forever

But these are the defaults anyhow!

Thanks,

sean



Thanks for confirming this occurring.

If it happens again I'll file a bug, but it's really hard to track down 
something so sporadic.


sean


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[asterisk-users] 10.3 : sip loses registration ?

[sean darcy]

We found this morning we had no SIP connection to another site. sip show 
registry on the main site gave "no authentication". sip show peers on 
the other site showed the peer unspecified.


The odd part about this:  doing sip reload on the main site made it all 
work. Nothing else was changed.


Main site:

 sip show registry

SFO:5060  N  sip_outgoin   105 No 
AuthenticationSat, 14 Apr 2012 14:48:15

4 SIP registrations.
..

PBX*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf':   == Found
  == Parsing '/etc/asterisk/users.conf':   == Found
  == Using SIP TOS bits 96
  == Using SIP CoS mark 3
  == Parsing '/etc/asterisk/sip_notify.conf':   == Found
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

On other site:

sip show peers
.
sip_outgoing/s   (Unspecified)D   N 
0Unmonitored

..
13 sip peers [Monitored: 3 online, 7 offline Unmonitored: 2 online, 1 
offline]


result of sip reload on main site:
-- Registered SIP 'sip_outgoing' at xxx.yyy.zzz.aaa:9345


How do I/can I set the main site to retry a registration? I've now 
changed sip.conf to add:


registertimeout=20
registerattempts=0;Default is 0 tries, continue forever

But these are the defaults anyhow!

Thanks,

sean


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Re: [asterisk-users] syntax error from digium fax manual ??

[sean darcy]

On 04/09/2012 08:51 PM, Barry Miller wrote:

On Mon, Apr 09, 2012 at 06:21:40PM -0400, sean darcy wrote:

I've cut and pasted from the digium fax admin manual:

exten =>  send,1,NoOp( SENDING FAX )
exten =>  send,n,Wait(6)
exten =>  send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten =>  send,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})

 -- Executing [send@sendPDFasFax:1] NoOp("DAHDI/4-1", " SENDING
FAX ") in new stack
 -- Executing [send@sendPDFasFax:2] Wait("DAHDI/4-1", "6") in new stack
 -- Channel 4 detected a CED tone from the network.
[Apr  9 15:29:02] WARNING[2912]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror():  syntax error: syntax error, unexpected '+', expecting
$end; Input:
   + 1
   ^
[Apr  9 15:29:02] WARNING[2912]: ast_expr2.fl:472 ast_yyerror: If you
have questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
 -- Executing [send@sendPDFasFax:3] Set("DAHDI/4-1",
"GLOBAL(FAXCOUNT)=") in new stack
   == Setting global variable 'FAXCOUNT' to ''

The error seems to be saying that I need a closing "}" or "]", but it
looks like it has closing brackets.

Any suggestions?


This is exactly the error you'd get if FAXCOUNT is null or not set.
(Because then the expression would be the invalid '$[ + 1]'.)


 Yup. Inserted a set to zero if null before test. Thanks.

sean


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[asterisk-users] syntax error from digium fax manual ??

[sean darcy]

I've cut and pasted from the digium fax admin manual:

exten => send,1,NoOp( SENDING FAX )
exten => send,n,Wait(6)
exten => send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten => send,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})

-- Executing [send@sendPDFasFax:1] NoOp("DAHDI/4-1", " SENDING 
FAX ") in new stack

-- Executing [send@sendPDFasFax:2] Wait("DAHDI/4-1", "6") in new stack
-- Channel 4 detected a CED tone from the network.
[Apr  9 15:29:02] WARNING[2912]: ast_expr2.fl:468 ast_yyerror: 
ast_yyerror():  syntax error: syntax error, unexpected '+', expecting 
$end; Input:

  + 1
  ^
[Apr  9 15:29:02] WARNING[2912]: ast_expr2.fl:472 ast_yyerror: If you 
have questions, please refer to 
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [send@sendPDFasFax:3] Set("DAHDI/4-1", 
"GLOBAL(FAXCOUNT)=") in new stack

  == Setting global variable 'FAXCOUNT' to ''

The error seems to be saying that I need a closing "}" or "]", but it 
looks like it has closing brackets.


Any suggestions?

sean


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[asterisk-users] another non-root problem: unable to set utime ??

[sean darcy]

I'm trying to run asterisk as "asterisk". Which is harder than I thought.

10.3.0. When I put a callfile into /var/spool/asterisk/outgoing, I get 
this warning:


Unable to set utime on /var/spool/asterisk/outgoing/callfile.call: 
Operation not permitted


ls -l /var/spool
.
drwxr-x---.  9 asterisk asterisk 4096 Apr  7 21:41 asterisk

ls -l /var/spool/asterisk
...
drwxrwx---. 2 asterisk asterisk 4096 Apr  7 21:14 outgoing

any help appreciated.

sean


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Re: [asterisk-users] Unable to access the running directory (Permission denied).

[sean darcy]

On 04/07/2012 04:20 PM, Noah Engelberth wrote:

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Saturday, April 07, 2012 4:10 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Unable to access the running directory (Permission 
denied).

On 10.3, running as user asterisk.

ps aux | grep bin/asterisk
asterisk  1860  0.2  1.8 1246948 33116 ?   Ssl  16:02   0:00
/usr/sbin/asterisk -C /etc/asterisk/asterisk.conf

When I login into asterisk as user asterisk I get:

"Unable to access the running directory (Permission denied).  Changing to '/' for 
compatibility."

All the folders are set with asterisk.asterisk.

/var/lib/asterisk
/var/log/asterisk
/var/run/asterisk
/var/spool/asterisk

Any suggestions appreciated.

sean


--

In order to reconnect to asterisk (asterisk -r), you need root permissions.  So 
either you have to do it as root (bad), or use sudo to do it as user asterisk 
(recommended).

Noah



Hmm. Surprising. I would have thought you could reconnect to asterisk if 
you were the user under which asterisk was running. But ok, will use sudo.


Thanks,

sean


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[asterisk-users] Unable to access the running directory (Permission denied).

[sean darcy]

On 10.3, running as user asterisk.

ps aux | grep bin/asterisk
asterisk  1860  0.2  1.8 1246948 33116 ?   Ssl  16:02   0:00 
/usr/sbin/asterisk -C /etc/asterisk/asterisk.conf


When I login into asterisk as user asterisk I get:

"Unable to access the running directory (Permission denied).  Changing 
to '/' for compatibility."


All the folders are set with asterisk.asterisk.

/var/lib/asterisk
/var/log/asterisk
/var/run/asterisk
/var/spool/asterisk

Any suggestions appreciated.

sean


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[asterisk-users] 10.3.0: gtalk_request: No XMPP client to talk to, us (partial JID)

[sean darcy]

Trying to use gtalk:

-- Executing [andy@ipkall:2] Dial("SIP/ipkall-", 
"gtalk/andy-gtalk/+1xxxyyyz...@voice.google.com") in new stack
[Apr  1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP 
client to talk to, us (partial JID) : andy-gtalk


gtalk.conf

[general]
context=google-in   ; Context to dump call into
allowguest=yes
stunaddr = numb.viagenie.ca
bindaddr=0.0.0.0
externip=aa.bb.cc.dd

disallow=all
allow=ulaw

[andy-gtalk]
username=@gmail.com
context=google-in
connection=andy-jabber

gtalk show settings

Global Settings:

  UDP Bindaddress:   0.0.0.0
  Stun Address:  66.228.45.110
  External IP:   aa.bb.cc.dd
  Context:   google-in
  Codecs:(ulaw)
  Parking Lot:   default
  Allow Guest:   Yes

jabber.conf:

[andy-jabber]
type=client
serverhost=talk.google.com
username=@gmail.com/Talk
secret=<>
port=5222
usetls=yes
usesasl=yes
statusmessage=No one here
status=xaway

jabber show connections
Jabber Users and their status:
   [andy-jabber] @gmail.com/Talk - Connected

jabber test andy-jabber
User: @gmail.com
Resource: gmail.675A4337
   client: http://mail.google.com/xmpp/client/caps
   version: 1.1
   Jingle Capable: 1
Priority: 0
Status: 3
Message:

Oooh a working message stack!



So jabber seems to be working.

Once while trying this I got this gtalk error:

WARNING[2571]: chan_gtalk.c:1923 gtalk_request: Could not find recipient.

Thanks for any help.

sean



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[asterisk-users] DAHDI works, but returns CHANUNAVAIL ??

[sean darcy]

 DAHDI 2.6.0, dahdi show status
Description  Alarms  IRQbpviol CRC 
Fra Codi Options  LBO
Wildcard TDM400P REV I Board 5   OK  0  0  0 
CAS Unk   0 db (CSU)/0-133 feet (DSX-1)


Dahdi 1 is an internal extension, dahdi 4 is pstn.

This call completes. But DAHDI comes back with CHANUNAVAIL. This a 
problem since we then test for CHANUNAVAIL to use an alternative provider.


   -- Executing [s@DialOut:17] Dial("DAHDI/1-1", "DAHDI/4/1XXXYYY") 
in new stack

-- Called DAHDI/4/1XXXYYY
-- Hanging up on 'DAHDI/4-1'
-- Hungup 'DAHDI/4-1'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@DialOut:18] NoOp("DAHDI/1-1", ""Dialstatus is 
"CHANUNAVAIL"") in new stack


Any ideas?

sean


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[asterisk-users] 10.2.1 res_fax : "Unexpected command after page received..."

[sean darcy]

I'm setting up res_fax to use with an iax provider. I'm calling over 
PSTN to the provider. When I stand at our fax machine (Brother), I can 
see the call come in, and it appears to set up correctly. What is odd, 
however, is that asterisk drops off while the fax machine is still 
sending. I've lowered the baud rate to 9600, it's a single page fax. 
After less than 10 seconds asterisk stops receiving the fax, but the 
machine continues sending for another 20 seconds or so.


ReceiveFAX("IAX2/FaxIAX-4579", 
"/var/spool/asterisk/fax/20120317_1820.tif,df") in new stack
-- Channel 'IAX2/FaxIAX-4579' receiving FAX 
'/var/spool/asterisk/fax/20120317_1820.tif'

-- Auto fallthrough, channel 'IAX2/FaxIAX-4579' status is 'UNKNOWN'
-- Executing [h@incoming-fax:1] NoOp("IAX2/FaxIAX-4579", 
"FAXSTATUS: FAILED FAXERROR: Unexpected command after page received 
FAXPAGES: 0 @ bitrate 9600") in new stack


FWIW, ping times from the asterisk box and the iax provider are < 15ms, 
usually around 10ms.


Any thoughts?

sean


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Re: [asterisk-users] dreaded one-way audio with nat=yes

[sean darcy]

On 03/10/2012 12:13 AM, Vladimir Mikhelson wrote:

Sean,

I do not have experience with the Amazon service.  Cannot advise how to
implement it in their environment.

You need to have a route from your public IP(s) to your Asterisk
instance for all incoming connections on RTP ports.

Absence of this routing explains why SIP connection to your home
(egress) worked whereas incoming SIP connection from your SIP provider
(ingress) has a packed drop issue.  The egress connection is initiated
from the LAN and firewall happily NATs in this case. On the ingress
connection firewall drops all RTP traffic originated by your provider
while happily NATing the traffic originated by your Asterisk.

It is also a good idea to have "qualify=yes" in your SIP peers' settings
to keep these NAT tables on the firewall updated for incoming SIP traffic.

-Vladimir




On 3/9/2012 9:15 PM, sean darcy wrote:

On 03/09/2012 09:42 PM, Arstan Jusupov wrote:

Udp port 5060, udp port range 1-2 open? Those are for sip.

For iax2 udp port 4569

Make sure they are open.

Also can you register two ext from the same instance and see if you
can hear both ways

What kind of trunk do you have to the other side you calling?

Arstan
Sent from my iPhone

On Mar 10, 2012, at 10:20 AM, sean darcy   wrote:


On 03/09/2012 07:20 PM, Arstan Jusupov wrote:

It may sound silly but did you configure/open firewall ports on
amazon ec2? The instance itself as we as from the amazon ec2 panel?

Sent from my iPhone

On Mar 10, 2012, at 7:16 AM, sean darcywrote:


On 03/09/2012 04:16 PM, sean darcy wrote:

I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.

Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.

The dialplan is real easy:

[from-teliax-sip]
exten =>_j.,1,NoOp("From teliax sip with exten "${EXTEN}")
exten =>_j.,n,Set(3digitexten=${EXTEN:12:3}
exten =>_j.,n,NoOp("Callerid is " ${CALLERID(all)} )
exten =>_j.,n,GoTo(from-outside,${3digitexten},1)

[from-outside]
exten =>123,1,NoOp()
exten =>123,n,Answer()
exten =>123,n,Dial(SIP/jnctn/1212xxx)
exten =>123,n,HangUp()

sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat

sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No

sip show peer teliax:
Insecure : port,invite
Force rport : Yes

DirectMedia : No



And the cli doesn't show any problems:

NoOp("SIP/teliax-0022", ""From teliax sip with exten
"(123)"") in new stack
Set("SIP/teliax-0022", "3digitexten=123") in new stack
NoOp("SIP/teliax-0022", ""Callerid is " "") in new stack
Goto("SIP/teliax-0022", "from-outside,123,1") in new stack
-- Goto (from-outside,123,1)
NoOp("SIP/teliax-0022", "") in new stack
Answer("SIP/teliax-0022", "") in new stack
Dial("SIP/teliax-0022", "SIP/jnctn/1212aaa") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to
SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'

The called party can hear the calling party, but not the reverse!

Any help really appreciated!

sean



So I tried having teliax connect to the asterisk box with iax. But
now I get no audio both ways!

Answer("IAX2/iaxtest-1945", "") in new stack
GotoIf("IAX2/iaxtest-1945", "1?123,1") in new stack

 -- Goto (from-outside,123,1)
 -- Executing [123@from-outside:1] NoOp("IAX2/iaxtest-1945",
"") in new stack
 -- Executing [123@from-outside:2] Dial("IAX2/iaxtest-1945",
"SIP/jnctn/1aaabbb") in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Called SIP/jnctn/1aaabbb
 -- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
 -- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
 -- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
 -- SIP/jnctn- is ringing
 -- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
 -- IAX2/iaxtest-1945 requested special control 20, passing it
to SIP/jnctn-
 -- SIP/jnctn- answered IAX2/iaxtest-1945

Really puzzled.

sean


Well that's interesting. I hadn't realized tha

Re: [asterisk-users] dreaded one-way audio with nat=yes

[sean darcy]

On 03/09/2012 09:42 PM, Arstan Jusupov wrote:

Udp port 5060, udp port range 1-2 open? Those are for sip.

For iax2 udp port 4569

Make sure they are open.

Also can you register two ext from the same instance and see if you can hear 
both ways

What kind of trunk do you have to the other side you calling?

Arstan
Sent from my iPhone

On Mar 10, 2012, at 10:20 AM, sean darcy  wrote:


On 03/09/2012 07:20 PM, Arstan Jusupov wrote:

It may sound silly but did you configure/open firewall ports on amazon ec2? The 
instance itself as we as from the amazon ec2 panel?

Sent from my iPhone

On Mar 10, 2012, at 7:16 AM, sean darcy   wrote:


On 03/09/2012 04:16 PM, sean darcy wrote:

I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.

Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.

The dialplan is real easy:

[from-teliax-sip]
exten =>   _j.,1,NoOp("From teliax sip with exten "${EXTEN}")
exten =>   _j.,n,Set(3digitexten=${EXTEN:12:3}
exten =>   _j.,n,NoOp("Callerid is " ${CALLERID(all)} )
exten =>   _j.,n,GoTo(from-outside,${3digitexten},1)

[from-outside]
exten =>   123,1,NoOp()
exten =>   123,n,Answer()
exten =>   123,n,Dial(SIP/jnctn/1212xxx)
exten =>   123,n,HangUp()

sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat

sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No

sip show peer teliax:
Insecure : port,invite
Force rport : Yes

DirectMedia : No



And the cli doesn't show any problems:

NoOp("SIP/teliax-0022", ""From teliax sip with exten
"(123)"") in new stack
Set("SIP/teliax-0022", "3digitexten=123") in new stack
NoOp("SIP/teliax-0022", ""Callerid is " "") in new stack
Goto("SIP/teliax-0022", "from-outside,123,1") in new stack
-- Goto (from-outside,123,1)
NoOp("SIP/teliax-0022", "") in new stack
Answer("SIP/teliax-0022", "") in new stack
Dial("SIP/teliax-0022", "SIP/jnctn/1212aaa") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'

The called party can hear the calling party, but not the reverse!

Any help really appreciated!

sean



So I tried having teliax connect to the asterisk box with iax. But now I get no 
audio both ways!

   Answer("IAX2/iaxtest-1945", "") in new stack
   GotoIf("IAX2/iaxtest-1945", "1?123,1") in new stack

-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp("IAX2/iaxtest-1945", "") in new stack
-- Executing [123@from-outside:2] Dial("IAX2/iaxtest-1945", 
"SIP/jnctn/1aaabbb") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945

Really puzzled.

sean


Well that's interesting. I hadn't realized that iptables was set up on the 
instance, as well as the firewall from the security group on the control panel.

Flushed the instance iptables, which fixed a problem I was having with a phone 
registering.

But I still have my one-way audio. The calling party hears nothing from the 
called party.

sean



The instance firewall is flushed. The security group allows udp 
1-2 , 5060 and 4569.


Well it gets stranger:

I set up a sip link to my home. Dialed the teliax number from my cell. 
Asterisk used the sip link to my home - and that worked!


Dial("IAX2/iaxtest-584", "SIP/sip-to-home")

Which seems to mean that the teliax <-> asterisk link is fine.

But if I use a SIP/PSTN provider , I get one-way audio:

Dial("IAX2/iaxtest-515", "SIP/jnctn/")

Completely baffled.

sean


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Re: [asterisk-users] dreaded one-way audio with nat=yes

[sean darcy]

On 03/09/2012 07:20 PM, Arstan Jusupov wrote:

It may sound silly but did you configure/open firewall ports on amazon ec2? The 
instance itself as we as from the amazon ec2 panel?

Sent from my iPhone

On Mar 10, 2012, at 7:16 AM, sean darcy  wrote:


On 03/09/2012 04:16 PM, sean darcy wrote:

I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.

Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.

The dialplan is real easy:

[from-teliax-sip]
exten =>  _j.,1,NoOp("From teliax sip with exten "${EXTEN}")
exten =>  _j.,n,Set(3digitexten=${EXTEN:12:3}
exten =>  _j.,n,NoOp("Callerid is " ${CALLERID(all)} )
exten =>  _j.,n,GoTo(from-outside,${3digitexten},1)

[from-outside]
exten =>  123,1,NoOp()
exten =>  123,n,Answer()
exten =>  123,n,Dial(SIP/jnctn/1212xxx)
exten =>  123,n,HangUp()

sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat

sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No

sip show peer teliax:
Insecure : port,invite
Force rport : Yes

DirectMedia : No



And the cli doesn't show any problems:

NoOp("SIP/teliax-0022", ""From teliax sip with exten
"(123)"") in new stack
Set("SIP/teliax-0022", "3digitexten=123") in new stack
NoOp("SIP/teliax-0022", ""Callerid is " "") in new stack
Goto("SIP/teliax-0022", "from-outside,123,1") in new stack
-- Goto (from-outside,123,1)
NoOp("SIP/teliax-0022", "") in new stack
Answer("SIP/teliax-0022", "") in new stack
Dial("SIP/teliax-0022", "SIP/jnctn/1212aaa") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'

The called party can hear the calling party, but not the reverse!

Any help really appreciated!

sean



So I tried having teliax connect to the asterisk box with iax. But now I get no 
audio both ways!

   Answer("IAX2/iaxtest-1945", "") in new stack
   GotoIf("IAX2/iaxtest-1945", "1?123,1") in new stack

-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp("IAX2/iaxtest-1945", "") in new stack
-- Executing [123@from-outside:2] Dial("IAX2/iaxtest-1945", 
"SIP/jnctn/1aaabbb") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- SIP/jnctn- answered IAX2/iaxtest-1945

Really puzzled.

sean


Well that's interesting. I hadn't realized that iptables was set up on 
the instance, as well as the firewall from the security group on the 
control panel.


Flushed the instance iptables, which fixed a problem I was having with a 
phone registering.


But I still have my one-way audio. The calling party hears nothing from 
the called party.


sean


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Re: [asterisk-users] dreaded one-way audio with nat=yes

[sean darcy]

On 03/09/2012 04:16 PM, sean darcy wrote:

I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.

Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.

The dialplan is real easy:

[from-teliax-sip]
exten => _j.,1,NoOp("From teliax sip with exten "${EXTEN}")
exten => _j.,n,Set(3digitexten=${EXTEN:12:3}
exten => _j.,n,NoOp("Callerid is " ${CALLERID(all)} )
exten => _j.,n,GoTo(from-outside,${3digitexten},1)

[from-outside]
exten => 123,1,NoOp()
exten => 123,n,Answer()
exten => 123,n,Dial(SIP/jnctn/1212xxx)
exten => 123,n,HangUp()

sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no ; tried nonat

sip show peer jnctn:
Insecure : invite
Force rport : Yes
.
DirectMedia : No

sip show peer teliax:
Insecure : port,invite
Force rport : Yes

DirectMedia : No



And the cli doesn't show any problems:

NoOp("SIP/teliax-0022", ""From teliax sip with exten
"(123)"") in new stack
Set("SIP/teliax-0022", "3digitexten=123") in new stack
NoOp("SIP/teliax-0022", ""Callerid is " "") in new stack
Goto("SIP/teliax-0022", "from-outside,123,1") in new stack
-- Goto (from-outside,123,1)
NoOp("SIP/teliax-0022", "") in new stack
Answer("SIP/teliax-0022", "") in new stack
Dial("SIP/teliax-0022", "SIP/jnctn/1212aaa") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to SIP/teliax-0022
-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
== Spawn extension (from-outside, 123, 3) exited non-zero on
'SIP/teliax-0022'

The called party can hear the calling party, but not the reverse!

Any help really appreciated!

sean



So I tried having teliax connect to the asterisk box with iax. But now I 
get no audio both ways!


   Answer("IAX2/iaxtest-1945", "") in new stack
   GotoIf("IAX2/iaxtest-1945", "1?123,1") in new stack

-- Goto (from-outside,123,1)
-- Executing [123@from-outside:1] NoOp("IAX2/iaxtest-1945", "") in 
new stack
-- Executing [123@from-outside:2] Dial("IAX2/iaxtest-1945", 
"SIP/jnctn/1aaabbb") in new stack

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1aaabbb
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-

-- SIP/jnctn- is ringing
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-
-- IAX2/iaxtest-1945 requested special control 20, passing it to 
SIP/jnctn-

-- SIP/jnctn- answered IAX2/iaxtest-1945

Really puzzled.

sean


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[asterisk-users] dreaded one-way audio with nat=yes

[sean darcy]

I'm trying to move the asterisk server to an Amazon Web instance. We 
have teliax for our sip provider. I'd like for our DID lines to be 
connected to a users cell phone.


Seems simple enough, but I'm getting the dreaded one-way audio, even 
with nat=yes everyplace I can think of.


The dialplan is real easy:

[from-teliax-sip]
exten => _j.,1,NoOp("From teliax sip with exten "${EXTEN}")
exten => _j.,n,Set(3digitexten=${EXTEN:12:3}
exten => _j.,n,NoOp("Callerid is " ${CALLERID(all)} )
exten => _j.,n,GoTo(from-outside,${3digitexten},1)

[from-outside]
exten => 123,1,NoOp()
exten => 123,n,Answer()
exten => 123,n,Dial(SIP/jnctn/1212xxx)
exten => 123,n,HangUp()

sip.conf:
[general]
externaddr=xx.yyy.zz.aa
nat=yes
directmedia=no  ; tried nonat

sip show peer jnctn:
  Insecure : invite
  Force rport  : Yes
  .
  DirectMedia  : No

sip show peer teliax:
  Insecure : port,invite
  Force rport  : Yes
  
  DirectMedia  : No



And the cli doesn't show any problems:

NoOp("SIP/teliax-0022", ""From teliax sip with exten 
"(123)"") in new stack

Set("SIP/teliax-0022", "3digitexten=123") in new stack
NoOp("SIP/teliax-0022", ""Callerid is " "") in new stack
Goto("SIP/teliax-0022", "from-outside,123,1") in new stack
-- Goto (from-outside,123,1)
NoOp("SIP/teliax-0022", "") in new stack
Answer("SIP/teliax-0022", "") in new stack
Dial("SIP/teliax-0022", "SIP/jnctn/1212aaa") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Called SIP/jnctn/1212aaa
-- SIP/jnctn-0023 is making progress passing it to 
SIP/teliax-0022

-- SIP/jnctn-0023 answered SIP/teliax-0022
-- Locally bridging SIP/teliax-0022 and SIP/jnctn-0023
  == Spawn extension (from-outside, 123, 3) exited non-zero on 
'SIP/teliax-0022'


The called party can hear the calling party, but not the reverse!

Any help really appreciated!

sean


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Re: [asterisk-users] 10.2.0-rc2: permitted contact can't register.

[sean darcy]

On 02/29/2012 02:30 AM, Zohair Raza wrote:

You want to allow single IP or whole subnet ?



Regards,
Zohair Raza



On Wed, Feb 29, 2012 at 4:44 AM, sean darcy mailto:seandar...@gmail.com>> wrote:

An outside device can't register:

WARNING: getnameinfo(): ai_family not supported
WARNING: chan_sip.c:14456 parse_register_contact: Domain
'69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP )

sip.conf:
[general]
...
alwaysreject=yes
dynamic_exclude_static = yes
allowguest=no
contactdeny=0.0.0.0/0.0.0.0 <http://0.0.0.0/0.0.0.0>
contactpermit=69.0.0.0/255.0.__0.0 <http://69.0.0.0/255.0.0.0>

I've also tried without any "contactdeny". Same result.

I'm completely puzzled. Any help appreciated.

sean



I don't care. Single ip, subnet, or nothing at all, nothing allows 
registration.


sean


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[asterisk-users] 10.2.0-rc2: permitted contact can't register.

[sean darcy]

An outside device can't register:

WARNING: getnameinfo(): ai_family not supported
WARNING: chan_sip.c:14456 parse_register_contact: Domain 
'69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP )


sip.conf:
[general]
...
alwaysreject=yes
dynamic_exclude_static = yes
allowguest=no
contactdeny=0.0.0.0/0.0.0.0
contactpermit=69.0.0.0/255.0.0.0

I've also tried without any "contactdeny". Same result.

I'm completely puzzled. Any help appreciated.

sean


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Re: [asterisk-users] Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network?

[sean darcy]

On 02/17/2012 03:28 AM, Frank Church wrote:

Should a Linksys Sipura 2102 be configured with nat=yes even if it is on
the local network?

I have been having some troubles with a Linksys Sipura 2100 series,
which suffers from NO AUDIO after a few calls.. Because it is on the
same subnet as Asterisk it is configured with nat=no. When you think of
it because the Sipura 2100 is a broadband router, the voice part may be
considered as being behind NAT, as are other devices plugged into its
yellow socket defintely are.

In theory is it likely to be better that way?



That was my question exactly. Look for the thread:

Should you "ever" use nat=no?

And the answer is almost never.

sean


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Re: [asterisk-users] Should you "ever" use nat=no?

[sean darcy]

On 02/16/2012 12:30 PM, Kevin P. Fleming wrote:

On 02/11/2012 06:59 PM, Bruce B wrote:

If your server is open to the internet and in SIP general section you
have nat=no and in peers you have nat=yes or vice versa then it's
possible to enumerate your extension. Because Asterisk responds with
different messages if the extension exists or not based on that
difference in the nat setting then it's possible to tell if an extension
100 exists or not. Over the past few years, Digium has come to
realization to respond to all unauthenticated calls the same way in
order to thwart any attack attempts or guesses on the extension but it's
still not perfect yet as these improvements are done at a really slow
pace. Regardless, they are being made and there truely is a security
risk.


"really slow pace"? Please point out any one of these issues that took
an unnecessarily long time to resolve once it was identified and brought
to the development team's attention.



I always use nat=yes. I don't even know why nat=no exists as there is
nothing that can't be done with nat=yes. Plus nat=yes will take care of
some of the surprise one-way audio scenarios as well so why use nat=no
at all?! I vote to totally get rid of the nat setting all together and
hard code it and set it to yes but again there are others who may not
agree.


As was already pointed out in the discussions that lead up to the 'nat'
default being changed, there are SIP endpoints out there that do not
work properly with 'nat=yes' (or 'nat=force_rport'). They behave
improperly when Asterisk adds an 'rport' parameter to the top-level Via
header in its responses. Setting 'nat=no' is the only way to keep this
from happening.



So in my case, these 40 internal sip devices (primarily aastra), which 
are not nat'd with respect to asterisk, should all be nat=yes unless 
they are unable to deal with the rport parameter? And if they are 
unable, setting nat=yes would immediately break them? If not, what are 
the symptoms of being unable  to behave properly with the rport parameter?


sean


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[asterisk-users] Should you "ever" use nat=no?

[sean darcy]

I've been lurking on the dev discussion on creating nat=auto. It all 
leads me to think there's no reason to use nat=no.


We have about 60 internal sip extensions connected to an multihomed 
asterisk box where the external ip is not nat'ed. Each of the internal 
sip contexts has nat=no. On startup I get a slew of warnings about 
intruders being able to distinguish real extensions. But that isn't 
right, is it? Or if it is, wouldn't the intruder have to be on the 
"inside" 10.0.0.0 net?


But so what? Does nat=no buy you anything? faster? slicker? richer?

sean



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Re: [asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings

[sean darcy]

Nope. No templates. And sterisk is running as root.

sean
On 01/26/2012 10:50 PM, Jim DeVito wrote:

Are you by chance using templates (!) In your sip.con? Ive had access
denied errors befor when running as non root.

- Original message -

 I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood
 of:

 WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL logic
 error or missing database

 AFAIK, I'm not doing any database puts (or gets). There were no such
 warnings in 1.8.8.0.

 What do I need to do to silence these warnings?

 sean



Nope. No templates. And asterisk is running as root.

sean


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[asterisk-users] upgraded 1.8.8.0 > 10.1.0-rc2: now db warnings

[sean darcy]

I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood of:

WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL logic 
error or missing database


AFAIK, I'm not doing any database puts (or gets). There were no such 
warnings in 1.8.8.0.


What do I need to do to silence these warnings?

sean


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[asterisk-users] 10.1.0-rc1 : WARNING: abstract_jb.c:384 jb_get_and_deliver: AST_JB_IMPL_NOFRAME

[sean darcy]

– adaptive jitterbuffer created on channel DAHDI/1-1
[Jan 20 12:25:53] WARNING[2932]: abstract_jb.c:384 jb_get_and_deliver: 
AST_JB_IMPL_NOFRAME is returned from the adaptive jb when now=4496 >= 
next=4496, jbnext=4496!
[Jan 20 12:25:53] WARNING[2932]: abstract_jb.c:384 jb_get_and_deliver: 
AST_JB_IMPL_NOFRAME is returned from the adaptive jb when now=4498 >= 
next=4496, jbnext=4496!
[Jan 20 12:25:53] WARNING[2932]: abstract_jb.c:384 jb_get_and_deliver: 
AST_JB_IMPL_NOFRAME is returned from the adaptive jb when now=4499 >= 
next=4496, jbnext=4496!

...

This issue seems to have been around for 2 years or more.

https://issues.asterisk.org/jira/browse/ASTERISK-15848

Any one else see this? Is it harmless?

sean


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Re: [asterisk-users] best softphone for 2012?

[Sean Darcy]

On Sat, Jan 7, 2012 at 9:34 AM, Gilles  wrote:
> On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy 
> wrote:
>>But what really made us choose linphone was you use it on android/iphone.
>>
>>That has been a huge plus. As a bonus, you can use any degegistered
>>smartphone - that is, one not hooked up to the cellular network,only
>>wireless - as a softphone.
>
> I guess you meant "de-registered smartphone" : what does it mean?
>
>

Yes, I did mean de-registered.  I meant a phone that no longer has the
ability to use the cellular network - only wifi. For instance, we have
a couple of Droids that used to be on Verizon. They work just fine as
sip-phones over wifi.

sean

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