Re: [asterisk-users] clarification on gosub, macros and AEL

[Brian J. Murrell]

On Fri, 2019-10-11 at 14:12 -0400, Brian J. Murrell wrote:
> I'm trying to clarify my understand of gosub, macros and AEL.
> 
> My understanding is that macros using the Macro() application, which
> is
> defined in extensions.conf by:
> 
> [macro-foo]
> ...
> 
> and called in extensions.conf with
> 
> exten => _9NXXNXX.,n,Macro(fastbusy)
> 
> is deprecated in favour of Gosub().  True so far?
> 
> But then there are "macro"s defined in extensions.ael:
> 
> macro foo() { ... }
> 
> and called in extensions.ael with:
> 
> (...)
> 
> These are not deprecated because they use Gosub() underneath and not
> Macro()?  Still good so far?
> 
> So if macros in extensions.ael are not deprecated because they just
> use
> Gosub() underneath them, when why, when I try to use Gosub() and
> contexts in extensions.ael directly do I get warnings like:
> 
> [Oct 11 13:50:55] WARNING[10343]: ael/pval.c:2526 check_pval_item:
> Warning: file /etc/asterisk/extensions.ael, line 552-552: application
> call to Gosub affects flow of control, and needs to be re-written
> using AEL if, while, goto, etc. keywords instead!
> 
> but not with these Gosub() implemented "macros"?

Nobody has any information or opinions on any of this?

Cheers,
b.



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[asterisk-users] clarification on gosub, macros and AEL

[Brian J. Murrell]

I'm trying to clarify my understand of gosub, macros and AEL.

My understanding is that macros using the Macro() application, which is
defined in extensions.conf by:

[macro-foo]
...

and called in extensions.conf with

exten => _9NXXNXX.,n,Macro(fastbusy)

is deprecated in favour of Gosub().  True so far?

But then there are "macro"s defined in extensions.ael:

macro foo() { ... }

and called in extensions.ael with:

(...)

These are not deprecated because they use Gosub() underneath and not
Macro()?  Still good so far?

So if macros in extensions.ael are not deprecated because they just use
Gosub() underneath them, when why, when I try to use Gosub() and
contexts in extensions.ael directly do I get warnings like:

[Oct 11 13:50:55] WARNING[10343]: ael/pval.c:2526 check_pval_item: Warning: 
file /etc/asterisk/extensions.ael, line 552-552: application call to Gosub 
affects flow of control, and needs to be re-written using AEL if, while, goto, 
etc. keywords instead!

but not with these Gosub() implemented "macros"?

Cheers,
b.



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[asterisk-users] IPv4 address in SDP o= is (null) when configured for NAT using pjsip

[Brian J. Murrell]

Using Asteirsk 13.28.1:

If I configure my pjsip transport to handle NAT from the Internet:

[transport-tcp]
type=transport
protocol=tcp
bind=10.75.22.8:5060
local_net=10.75.22.0/24
external_media_address=[external address redacted]
external_signaling_address=[external address redacted]

When a call comes from a TCP registered SIP client on the Internet,
Asterisk is setting the IPv4 address in the "o=" and "c=" lines of the
SDP ICE payload to "(null)":

v=0
o=- 3654 548 IN IP4 (null)
s=Asterisk
c=IN IP4 (null)

IPv4 addresses in all of the a= lines are still correct.

This "(null)" of course causes the caller to fail to complete the call.

Is there something more I have to do to have Asterisk/pjsip properly
set the IPv4 address in the SDP ICE payloads?

Cheers,
b.



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Re: [asterisk-users] if statement with true value that contains a colon

[Brian J. Murrell]

On Fri, 2019-09-13 at 14:21 +0200, Administrator TOOTAI wrote:
> 
> Escape it with \

Tried that.  It doesn't work.

Cheers,
b.



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[asterisk-users] if statement with true value that contains a colon

[Brian J. Murrell]

How can I use an IF statement with a true value being a variable that
has a colon in it?  The colon in the true value variable is being taken
as the delimiter for the false value.

The only solution I came up with was some hackery to use STRREPLACE to
replace the : with a % before the IF statement and then use STRREPLACE
again after to change the % back to a :.

i.e.:

Set(dialexts=${STRREPLACE(dialexts,:,%)});
Set(dialexts=${IFTIME(8:00-22:00?${dialexts}&${MBDR}:${dialexts})});
Set(dialexts=${STRREPLACE(dialexts,%,:)});


Is there no better alternative?

Cheers,
b.



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Re: [asterisk-users] Preventing rewrite of To: address in MESSAGE transactions

[Brian J. Murrell]

On Tue, 2019-07-16 at 16:20 -0400, Brian J. Murrell wrote:
> Is there any option to prevent Asterisk from rewriting the To:
> address
> of a SIP MESSAGE that it's received and will forward to another SIP
> client?
> 
> That is, when Asterisk receives a MESSAGE with the To; header saying:
> 
> To: 
> 
> and wants to forward that to foo@10.75.22.100, I'd like the To:
> header
> to remain:
> 
> To: 
> 
> when it sends it.
> 
> Any possibility to do this?

Should I take the lack of response as meaning no?

No comment on any of this thread at all?

Cheers,
b.



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[asterisk-users] Preventing rewrite of To: address in MESSAGE transactions

[Brian J. Murrell]

Is there any option to prevent Asterisk from rewriting the To: address
of a SIP MESSAGE that it's received and will forward to another SIP
client?

That is, when Asterisk receives a MESSAGE with the To; header saying:

To: 

and wants to forward that to foo@10.75.22.100, I'd like the To: header
to remain:

To: 

when it sends it.

Any possibility to do this?

Cheers,
b.



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Re: [asterisk-users] RHS of the To: address in MESSAGE transactions

[Brian J. Murrell]

On Thu, 2019-06-06 at 09:33 -0400, Brian J. Murrell wrote:
> I'm trying to use linphone-android with asterisk but there is an
> aspect
> of the way asterisk and linphone-android interact with MESSAGE
> transactions that is causing problems.
> 
> The linphone-android folks consider both the To: and From: address in
> MESSAGE transactions when deciding which "chat" to put a received
> MESSAGE into.  Every combination of To: and From: address are a
> separate "chat" in their messaging paradigm.
> 
> So when asterisk sends a MESSAGE transaction such as:
> 
> MESSAGE
> sip:my_sip_account@[2001:123:ab:123:51e2:cc83:ae66:8c70]:38915;transp
> ort=udp;app-id=755770037818;pn-type=firebase;pn-timeout=0;pn-
> tok==[redacted];pn-silent=1 SIP/2.0
> Via: SIP/2.0/UDP
> [2001:123:ab:123::2]:5060;rport;branch=z9hG4bKPj138c026c-4437-4b59-
> 982f-f991521d3cdc
> From: "5565551212" ;tag=5b5fe395-
> ff22-44fa-aa6b-7f770f8e0026
> To:  id=755770037818;pn-type=firebase;pn-timeout=0;pn-tok==[redacted];pn-
> silent=1>
> Contact: 
> Call-ID: 5e4fc686-72ce-4c20-bd2f-7f82e232a9db
> CSeq: 29808 MESSAGE
> Max-Forwards: 70
> User-Agent: Asterisk PBX 13.26.0
> Content-Type: text/plain
> Content-Length: 4
> 
> hey!
> 
> it files it into the chat for the combination of:
> 
> From: "5565551212" ;tag=5b5fe395-
> ff22-44fa-aa6b-7f770f8e0026
> To:  id=755770037818;pn-type=firebase;pn-timeout=0;pn-tok==[redacted];pn-
> silent=1>
> 
> and because the To: includes the IP address of the client, every time
> the client moves networks (or even regenerates a new "Privacy
> Extensions" IPv6 address) a new chat is created for the same sender.
> 
> Their position is that the RHS of the To: should be the name of the
> Asterisk machine such as:
> 
> To:  type=firebase;pn-timeout=0;pn-tok==[redacted];pn-silent=1>
> 
> While that seems odd to me, from a common-sense perspective, I don't
> have a deep enough background in the SIP protocol to decide if it's
> wrong or not, and if it's not, how to make Asterisk do it, as in
> something similar to the pjsip from_domain endpoint parameter that
> can
> be used to set the domain of the From: header for MESSAGEs to that
> endpoint.
> 
> Any opinions, ideas or otherwise?

Nobody has any options either way on this one?

Cheers,
b.



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[asterisk-users] RHS of the To: address in MESSAGE transactions

[Brian J. Murrell]

I'm trying to use linphone-android with asterisk but there is an aspect
of the way asterisk and linphone-android interact with MESSAGE
transactions that is causing problems.

The linphone-android folks consider both the To: and From: address in
MESSAGE transactions when deciding which "chat" to put a received
MESSAGE into.  Every combination of To: and From: address are a
separate "chat" in their messaging paradigm.

So when asterisk sends a MESSAGE transaction such as:

MESSAGE 
sip:my_sip_account@[2001:123:ab:123:51e2:cc83:ae66:8c70]:38915;transport=udp;app-id=755770037818;pn-type=firebase;pn-timeout=0;pn-tok==[redacted];pn-silent=1
 SIP/2.0
Via: SIP/2.0/UDP 
[2001:123:ab:123::2]:5060;rport;branch=z9hG4bKPj138c026c-4437-4b59-982f-f991521d3cdc
From: "5565551212" 
;tag=5b5fe395-ff22-44fa-aa6b-7f770f8e0026
To: 

Contact: 
Call-ID: 5e4fc686-72ce-4c20-bd2f-7f82e232a9db
CSeq: 29808 MESSAGE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.26.0
Content-Type: text/plain
Content-Length: 4

hey!

it files it into the chat for the combination of:

From: "5565551212" 
;tag=5b5fe395-ff22-44fa-aa6b-7f770f8e0026
To: 


and because the To: includes the IP address of the client, every time
the client moves networks (or even regenerates a new "Privacy
Extensions" IPv6 address) a new chat is created for the same sender.

Their position is that the RHS of the To: should be the name of the
Asterisk machine such as:

To: 


While that seems odd to me, from a common-sense perspective, I don't
have a deep enough background in the SIP protocol to decide if it's
wrong or not, and if it's not, how to make Asterisk do it, as in
something similar to the pjsip from_domain endpoint parameter that can
be used to set the domain of the From: header for MESSAGEs to that
endpoint.

Any opinions, ideas or otherwise?  Is the position of what should be in
the RHS of the To: address even valid or MUST it always be the address
of the next hop on path to delivering the MESSAGE?

Cheers,
b.



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Re: [asterisk-users] IPv6 transport results in ICE with only IPv6 candidates

[Brian J. Murrell]

On Wed, 2019-04-17 at 13:50 -0400, Joshua C. Colp wrote:
> 
> The same escaping should apply there for extensions.conf as it's a
> config file thing, I don't use AEL and don't know anything in that
> regard. It may work the same way there.

How very odd.  It is working now.  I am sure I did nothing different
than the first time.

Cheers,
b.



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Re: [asterisk-users] IPv6 transport results in ICE with only IPv6 candidates

[Brian J. Murrell]

On Wed, 2019-04-17 at 11:56 -0400, Joshua C. Colp wrote:
> On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote:
> > 
> > I can add it onto the end of the variable in the Dial() command:
> > 
> > Dial(${FRED};transport=tcp,${timeout},TtWw);

[ the part you trimmed ]:

and it has the desired effect, but of course specifying at the point
where the Dial command is sets the transport policy much too broadly.

> You escape the ; character by placing \ in front of it. IE:
> 
> DIal(${FRIED}\;transport=tcp,${timeout},TtWw);

So per the above ("has the desired effect") one doesn't even have to
escape the ; in the Dial() command.

But I don't want the ";transport=tcp" hard-coded into the Dial()
command argument where the Dial() command is executed.  That is making
a transport policy (i.e. not every destination that gets to that Dial()
command should use tcp) decision for far too many destinations.

The ";transport=tcp" needs to be added where the destination is defined
into a variable:

FRED=PJSIP/outgoing/sip:f...@sip.example.com;transport=tcp

in extensions.conf or:

FRED="PJSIP/outgoing/sip:f...@sip.example.com;transport=tcp";

in extensions.ael.

Cheers,
b.



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Re: [asterisk-users] IPv6 transport results in ICE with only IPv6 candidates

[Brian J. Murrell]

On Wed, 2019-04-17 at 10:04 -0400, Joshua C. Colp wrote:
> 
> You specify the transport in the SIP URI. For example:
> 
> sip:t...@example.com;transport=tcp

Hrm.  This is probably going to be pretty basic, but some googling
didn't seem to come up with anything.  How do you do this when you are
assigning to a variable in extensions.conf such as:

FRED=PJSIP/outgoing/sip:f...@sip.example.com;transport=tcp

The ; is taken as a comment starting character, isn't it?

Even in AEL:

FRED="PJSIP/outgoing/sip:f...@sip.example.com;transport=tcp";

The ;transport=tcp doesn't seem to survive.

I can add it onto the end of the variable in the Dial() command:

Dial(${FRED};transport=tcp,${timeout},TtWw);

and it has the desired effect, but of course specifying at the point
where the Dial command is sets the transport policy much too broadly.

Cheers,
b.



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[asterisk-users] IPv6 transport results in ICE with only IPv6 candidates

[Brian J. Murrell]

Hi,

I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport:

[transport-tcp-ipv6]
type=transport
protocol=tcp
bind=[2001:1234:5678:abcd::2]:5060

I also have an IPv4 version of that:

[transport-tcp-ipv4]
type=transport
protocol=tcp
bind=10.75.22.8:5060

I've then configured an endpoint to use it:

[outgoing]
type = endpoint
context = default
dtmf_mode = none
disallow = all
allow = all
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
direct_media = no
language = en
aors = outgoing
t38_udptl = yes
t38_udptl_ec = none
ice_support=yes
transport=transport-tcp-ipv6

But this seems to result in SDP payloads with ICE not offering any IPv4
candidates:

v=0
o=- 508048280 508048280 IN IP6 2001:1234:5678:abcd::2
s=Asterisk
c=IN IP6 2001:1234:5678:abcd::2
t=0 0
m=audio 14384 RTP/AVP 4 0 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 
116 107
a=ice-ufrag:6eea7b135aa990d714a67931034ac864
a=ice-pwd:0494fccd28b93f8a2b544b4741203e4f
a=candidate:H67c355d6 1 UDP 2130706431 fd31:aeb1:48df::2 14384 typ host
a=candidate:Hc1e19c54 1 UDP 2130706431 2607:abcd:9876:5432::247 14384 typ host
a=candidate:Hf5a91683 1 UDP 2130706431 2001:1234:5678:abcd::2 14384 typ host
a=candidate:Hdccea0f2 1 UDP 2130706431 2001:123:ab:123::2 14384 typ host
a=candidate:Hcbb5ed22 1 UDP 2130706431 fe80::21f:c6ff:fec4:926a 14384 typ host
a=candidate:H67c355d6 2 UDP 2130706430 fd31:aeb1:48df::2 14385 typ host
a=candidate:Hc1e19c54 2 UDP 2130706430 2607:abcd:9876:5432::247 14385 typ host
a=candidate:Hf5a91683 2 UDP 2130706430 2001:1234:5678:abcd::2 14385 typ host
a=candidate:Hdccea0f2 2 UDP 2130706430 2001:123:ab:123::2 14385 typ host
a=candidate:Hcbb5ed22 2 UDP 2130706430 fe80::21f:c6ff:fec4:926a 14385 typ host
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=rtpmap:107 opus/48000/2
a=ptime:20
a=maxptime:20
a=sendrecv
m=video 11188 RTP/AVP 31 34 103 99 104 100 108
a=ice-ufrag:28e29ae16a4cc8d54ebf03cf56010f1a
a=ice-pwd:57ec01cf24ac689d2fb459bd0411d2b6
a=candidate:H67c355d6 1 UDP 2130706431 fd31:aeb1:48df::2 11188 typ host
a=candidate:Hc1e19c54 1 UDP 2130706431 2607:abcd:9876:5432::247 11188 typ host
a=candidate:Hf5a91683 1 UDP 2130706431 2001:1234:5678:abcd::2 11188 typ host
a=candidate:Hdccea0f2 1 UDP 2130706431 2001:123:ab:123::2 11188 typ host
a=candidate:Hcbb5ed22 1 UDP 2130706431 fe80::21f:c6ff:fec4:926a 11188 typ host
a=candidate:H67c355d6 2 UDP 2130706430 fd31:aeb1:48df::2 11189 typ host
a=candidate:Hc1e19c54 2 UDP 2130706430 2607:abcd:9876:5432::247 11189 typ host
a=candidate:Hf5a91683 2 UDP 2130706430 2001:1234:5678:abcd::2 11189 typ host
a=candidate:Hdccea0f2 2 UDP 2130706430 2001:123:ab:123::2 11189 typ host
a=candidate:Hcbb5ed22 2 UDP 2130706430 fe80::21f:c6ff:fec4:926a 11189 typ host
a=rtpmap:31 H261/9
a=rtpmap:34 H263/9
a=rtpmap:103 h263-1998/9
a=rtpmap:99 H264/9
a=rtpmap:104 MP4V-ES/9
a=rtpmap:100 VP8/9
a=rtpmap:108 VP9/9
a=sendrecv

But other endpoints where I don't specify a transport, I get both IPv4
and IPv6 candidates in the ICE SDP.

Presumably the problem with the [outgoing] endpoint is the
specification of an IPv6-TCP transport.  But I need to force the use of
TCP, so how can I do that without limiting it to a particular IP
protocol?

I can't seem to define a generic TCP transport:

[transport-tcp]
type=transport
protocol=tcp

in addition to the ones I have already defined above but I do need to
define the IPv4 and IPv6 transports as I need to bind them to specific
addresses.

Any ideas?

Cheers,
b.



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Re: [asterisk-users] Message: Authentication failed on manager interface

[Brian J. Murrell]

On Thu, 2019-04-04 at 15:08 +0200, Antony Stone wrote:
> 
> It's not "Password", it's "Secret" :)

Ha ha.  I knew it would be a head-smack type problem.

Cheers,
b.



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[asterisk-users] Message: Authentication failed on manager interface

[Brian J. Murrell]

I'm not sure how much more simple I can make this but I just cannot
seem to get my Asterisk 13 to accept a connection on the manager
interface:

--- manager.conf ---
[general]
enabled = yes

port = 5038
bindaddr = 127.0.0.1

[myasterisk]
secret=a
permit=0.0.0.0/0.0.0.0

read = all
write = all

So, couldn't be any more wide open and simpler to connect yet:

# echo -e "Action: Login\r\nUsername: myasterisk\r\nPassword: a\r\n\r\n" | ncat 
127.0.0.1 5038
Asterisk Call Manager/2.10.4
Response: Error
Message: Authentication failed

Even more basically if you like:

# telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to 127.0.0.1.
Escape character is '^]'.
Asterisk Call Manager/2.10.4
Action: Login
Username: myasterisk
Password: a

Response: Error
Message: Authentication failed

Connection closed by foreign host.

I have restarted asterisk after editing manager.conf, even though a
manager reload should be all that's needed.

server*CLI> manager show user myasterisk

  username: myasterisk
secret: 
   ACL: yes
 read perm: 
system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,security,message,all
write perm: 
system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,security,message,all
   displayconnects: yes
allowmultiplelogin: yes
 Variables: 

I know this is going to be a "head smack" problem, but seriously.  What
am I missing?

Cheers,
b.



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Re: [asterisk-users] best practices for dialing multiple contacts of multiple extensions

[Brian J. Murrell]

On Thu, 2019-02-21 at 11:17 -0500, Brian J. Murrell wrote:
> In the past, I have created variables that hold multiple extensions
> such as:
> 
> HOUSEPHONES=PJSIP/mom/grandma
> 
> so that I can do a Dial(${HOUSEPHONES},...) with it, to ring multiple
> phones.
> 
> But now some of those phones will be registering multiple times and
> thus have multiple contacts, so I want to use ${PJSIP_DIAL_CONTACTS()
> to expand all of the contacts of an endpoint.
> 
> But of course you can't pass a string such as what's in
> ${HOUSEPHONES}
> to ${PJSIP_DIAL_CONTACTS()}.
> 
> I wonder what others are doing about this?
> 
> I had considered:
> 
> HOUSEPHONES=PJSIP/mom&${PJSIP_DIAL_CONTACTS(grandma)}
> 
> but I suspect that that just adds the contacts for grandma at the
> time
> the dialplan is loaded and is not updated as her contacts actually
> change.  Correct?
> 
> I could imagine a bunch of "dialplan" code to break up a & separated
> string, strip off the PJSIP/ and then run each item through
> ${PJSIP_DIAL_CONTACTS()} and concatenate it all back together with
> 
> 
> Even in AEL (which is probably what I'd do it in) it seems like this
> could be a bit hairy and possibly somebody else must have invented
> this
> while already.
> 
> Or is there some other best known practice to handle all of this that
> I
> am not considering?

Since nobody offered up any BPs, I will provide the AEL macro that I
wrote to do this expansion:

macro expand_exten(ext) {
while ("${ext}" != "") {
Set(e=${SHIFT(ext,&)});
if ("${e:0:5}" = "PJSIP") {
Set(e=${PJSIP_DIAL_CONTACTS(${STRREPLACE(e,PJSIP/,)})});
} 
if ("${e}" != "") {
Set(newext=${newext}${e}&);
}
}
// blech!  global variables because there no way to make a
// function/gosub/etc that returns a value
Set(newext=${newext:0:-1});
Return(${newext});
return;
}

where the first argument passed in is an extention of the format:

PJSIP/

I'm pretty new to AEL, so any constructive criticism would be welcome.

Cheers,
b.



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Re: [asterisk-users] pjsip: don't require authentication from remote i register to

[Brian J. Murrell]

On Fri, 2019-03-01 at 15:54 -0500, Joshua C. Colp wrote:
> 
> That's correct. You'd either need to retrieve the line parameter from
> the outbound registration or forge the source IP address,

Can I eliminate the identify by IP address then, given that my ITSP is
supporting the line parameter?  Or make even better, require them both
to be identified?

> and as you stated the scope of what they can do is limited.

I guess this is just a risk that everyone lives with.  As a limited
scope risk, anyway.

Cheers,
b.



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Re: [asterisk-users] pjsip: don't require authentication from remote i register to

[Brian J. Murrell]

On Fri, 2019-03-01 at 15:41 -0500, Joshua C. Colp wrote:
> 
> I don't understand what you mean. Your ITSP has stated that they
> don't want you to do authentication with them, so you can't.

They are implying, as I am understanding them, that somehow SIP packets
they send me shouldn't need to be authenticated because they are
associated (i.e. "identify"ed in pjsip nomenclature) with my
registration to them.  It all sounds suspect to me but that's what I am
understanding them to be saying.

Ultimately, if I have this endpoint and it's unauthenticated, does it
create a security risk?

I suppose anyone could forge a UDP packet as coming from their IP
address, and as it's "identify"ed by IP on my side and I would accept
it without authentication being necessary.

But then I suppose they are only getting access to being able to
connect into an incoming dialplan context, so ringing extensions here,
but not being able to launch in and outbound (money costing) phone
call, at least without there being dialplan support to make outgoing
calls when calling in (i.e. like a calling card application or
somesuch, which should have it's own authentication anyway).

> If you are referring to the template - it's a template so by itself
> does not create an endpoint.

Yes, completely understood.

b.



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Re: [asterisk-users] pjsip: don't require authentication from remote i register to

[Brian J. Murrell]

On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote:
> you can try line functionality on the outbound registration which
> may or may not work[2] (requires the upstream to adhere to the RFC,
> which not all do).

My provider seems to implement this.

However even with the line=... in the:

SIP to address: sip:551212@:5060;line=dpnlyiu

res_pjsip is still sending a 401 challenge.

Removing the:

auth=itsp-auth

from my endpoint [template]:

[itsp-endpoint](!)

Has stopped pjsip from sending a 401 when my ITSP sends a SIP MESSAGE,
but do I really want to have that endpoint without authentication?

Cheers,
b.



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Re: [asterisk-users] pjsip: don't require authentication from remote i register to

[Brian J. Murrell]

On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote:
> 
> You either configure IP based matching using an identify section[1] 

That's what I did:

[itsp]
type=registration
transport=transport-udp
outbound_auth=itsp-auth
server_uri=sip:pop1.itsp.example.com
client_uri=sip:x...@pop1.itsp.example.com

[itsp-auth]
type=auth
auth_type=userpass
password=XXX
username=XXX

[itsp-endpoint](!)
type=endpoint
transport=transport-udp
context=from-itsp
message_context=messages
disallow=all
allow=ulaw
from_user=XXX
outbound_auth=itsp-auth
auth=itsp-auth
send_pai=yes

[itsp-aor](!)
type=aor
qualify_frequency=15

[itsp-pop1](itsp-endpoint)
aors=itsp-pop1
[itsp-pop1](itsp-aor)
contact=sip:x...@pop1.itsp.example.com:5060

[itsp-pop1]
type=identify
endpoint=itsp-pop1
;match=pop1.itsp.example.com
match=192.168.5.6

but SIP INVITE and SIP MESSAGE packets coming from 192.168.5.6 are
still being challenged with 401 and not even printing any
errors/warnings in the console about not being able to find an
endpoint.

> or you can try line functionality on the outbound registration which
> may or may not work[2] (requires the upstream to adhere to the RFC,
> which not all do).

I'll read up on that and try in the meanwhile.

Cheers,
b.



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[asterisk-users] pjsip: don't require authentication from remote i register to

[Brian J. Murrell]

I'm being told by my ITSP that my Asterisk shouldn't be challenging
their system to authenticate (i.e. a 401 response) when they send me a
SIP MESSAGE (or I suppose a SIP INVITE for that matter).

But I'm not sure what a pjsip.conf configuration for that looks like.

How does one associate an incoming call/message with an existing
authenticated outgoing registration so that Asterisk doesn't return a
401 requiring authentication?

Cheers,
b.



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Re: [asterisk-users] PJSIP: 481 Call/Transaction Does Not Exist (only) for MESSAGE method

[Brian J. Murrell]

On Sun, 2019-02-17 at 17:31 -0500, Brian J. Murrell wrote:
> I have a PJSIP trunk set up which works fine for voice.  I can call
> out
> and I receive calls from it once it registers.
> 
> What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE)
> events.  It was working earlier today but I seem to have done
> something
> as I was enabling voice on the trunk to mess it up.  On receiving of
> a
> MESSAGE, my Asterisk sends a 401 for the ITSP to authenticate it's
> message, which it does, to which my Asterisk responds with a "481
> Call/Transaction Does Not Exist" and displays nothing at all in the
> console.

Nobody has any idea about this?  :-(

Cheers,
b.



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[asterisk-users] best practices for dialing multiple contacts of multiple extensions

[Brian J. Murrell]

In the past, I have created variables that hold multiple extensions
such as:

HOUSEPHONES=PJSIP/mom/grandma

so that I can do a Dial(${HOUSEPHONES},...) with it, to ring multiple
phones.

But now some of those phones will be registering multiple times and
thus have multiple contacts, so I want to use ${PJSIP_DIAL_CONTACTS()
to expand all of the contacts of an endpoint.

But of course you can't pass a string such as what's in ${HOUSEPHONES}
to ${PJSIP_DIAL_CONTACTS()}.

I wonder what others are doing about this?

I had considered:

HOUSEPHONES=PJSIP/mom&${PJSIP_DIAL_CONTACTS(grandma)}

but I suspect that that just adds the contacts for grandma at the time
the dialplan is loaded and is not updated as her contacts actually
change.  Correct?

I could imagine a bunch of "dialplan" code to break up a & separated
string, strip off the PJSIP/ and then run each item through
${PJSIP_DIAL_CONTACTS()} and concatenate it all back together with 

Even in AEL (which is probably what I'd do it in) it seems like this
could be a bit hairy and possibly somebody else must have invented this
while already.

Or is there some other best known practice to handle all of this that I
am not considering?

Cheers,
b.



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Re: [asterisk-users] branching in extensions.conf?

[Brian J. Murrell]

On Wed, 2019-02-20 at 12:38 -0700, John Kiniston wrote:
> I don't see any other messages in this thread other than your initial
> one
> and my response, perhaps the listserv hasn't relayed it to me yet.

I started a new thread:

http://lists.digium.com/pipermail/asterisk-users/2019-February/293668.html

since it was a separate question and only tangentially related to this
one.

> I switch between ExecIf/GotoIf/GosubIf and the IF function as needed
> in my
> dialplan design, if It's just a one liner I'll usually use either the
> IF
> function or the ExecIf,

Yeah, ExecIf was the suggested workaround in the companion thread
mentioned above.

> I've not run into any problem with colons in my IF statements but I
> don't
> think that's something I've had to work with/around before.

Likely that's just it.  I've only just run into it for the first time
now after many years of Asterisk.

> Shoot, I must have erased the colon when I was trying to fix the
> parentheses.

NP.  I understood your proposal.

> It seems like these days everyone wants to use ARI, You could always
> send
> the call into Stasis and handle your logic in a language of your
> choice but
> that's a whole different discussion.

Yeah.  extensions.conf is starting to feel long-in-the-tooth these
days.  :-)

Cheers,
b.



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Re: [asterisk-users] branching in extensions.conf?

[Brian J. Murrell]

On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote:
> Use the IF function to evaluate and change the dial command directly.

Thanks for taking the time, but that doesn't actually answer the
question I asked.  It in fact answers the caveat I specifically
mentioned:

> Granted the particular above example could probably be better
> written to simply modify $ARG2 based on ${SIP} rather than having two
> Dial() branches

But I also stated that I was just using it as example of a need to
branch and how cumbersome using Goto[If] makes it.

Again, appreciate that you took the time, but really looking for an
answer to the "better way to branch" question.

> My braces and parens may be off in this example sorry if it doesn't
> work
> out of the box.
> exten => s,n,Dial(${IF($["${SIP}" = "PJSIP"]?
> ${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})}{ARG2})},20,Tt
> Ww)

Hah.  This won't work either, for the reason I asked about in my
followup question about where the "true" value of an ${IF ...}has a
colon in it and is taken to be the end of the true value and the start
of the false value.

And your ${IF ...} is missing the colon between the true and false
values but I still understood what you were getting at.

Cheers,
b.



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[asterisk-users] if function when the true value has a colon in it?

[Brian J. Murrell]

Following up on my previously asked question if I rewrite the branching
example (not that it negates the more general branching question) I was
using as such:

exten => 
s,n,Set(EXT=${IF($[${SIP}=PJSIP]?${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})}:${ARG2})})
exten => s,n,Dial(${EXT},20,TtWw)
exten => s,n,Goto(s-${DIALSTATUS},1)

But that ${IF expression?tval:fval} doesn't work because tval has a :
in it which the if function is taking as the terminator for the tval.

How can I deal with that?  Double quoting the tval doesn't seem to work
as it introduces an unwanted close-double-quote into EXT which Dial()
barfs at.

Cheers,
b.



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[asterisk-users] branching in extensions.conf?

[Brian J. Murrell]

Is there any less cumbersome way of doing conditionalized/branching in
extensions.conf other than something like:

exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip)
exten => s,n,Dial(${ARG2},20,TtWw)
exten => s,n,Goto(afterdial)
exten => 
s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})},20,TtWw)
exten => s,n(afterdial),Goto(s-${DIALSTATUS},1)

Granted the particular above example could probably be better written
to simply modify $ARG2 based on ${SIP} rather than having two Dial()
branches, but using the above as just an example for wanting to have
branches, is there a less cumbersome way?

Cheers,
b.


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[asterisk-users] PJSIP: 481 Call/Transaction Does Not Exist (only) for MESSAGE method

[Brian J. Murrell]

I have a PJSIP trunk set up which works fine for voice.  I can call out
and I receive calls from it once it registers.

What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE)
events.  It was working earlier today but I seem to have done something
as I was enabling voice on the trunk to mess it up.  On receiving of a
MESSAGE, my Asterisk sends a 401 for the ITSP to authenticate it's
message, which it does, to which my Asterisk responds with a "481
Call/Transaction Does Not Exist" and displays nothing at all in the
console.

The configuration of the ITSP is:

[trunk]
type=registration
transport=transport-udp
outbound_auth=trunk-auth
server_uri=sip:itsp.example.com
client_uri=sip:use...@itsp.example.com

[trunk-auth]
type=auth
auth_type=userpass
password=
username=userid

[trunk-endpoint](!)
type=endpoint
transport=transport-udp
context=from-trunk
message_context=messages
disallow=all
allow=ulaw
from_user=userid
outbound_auth=trunk-auth
auth=trunk-auth
send_pai=yes

[trunk-aor](!)
type=aor
qualify_frequency=15

[trunk-foo](trunk-endpoint)
aors=trunk-foo
[trunk-foo](trunk-aor)
contact=sip:use...@itsp.example.com:5060

[trunk-foo]
type=identify
endpoint=trunk-foo
match=itsp.example.com

The SIP conversation when the ITSP is trying to send the MESSAGE:

<--- Received SIP request (456 bytes) from UDP:10.0.0.1:5060 --->
MESSAGE sip:s@10.75.22.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK51d6d4da;rport
Max-Forwards: 70
From: "551212" ;tag=as6c34cb69
To: 
Contact: 
Call-ID: 3e9735b4313e58f90b4b61c82f392c2e@10.0.0.1:5060
CSeq: 102 MESSAGE
User-Agent: itsp.example.com
X-SMS-To: 5551234567
Content-Type: text/plain;charset=UTF-8
Content-Length: 1

test message
<--- Transmitting SIP response (515 bytes) to UDP:10.0.0.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
10.0.0.1:5060;rport=5060;received=10.0.0.1;branch=z9hG4bK51d6d4da
Call-ID: 3e9735b4313e58f90b4b61c82f392c2e@10.0.0.1:5060
From: "551212" ;tag=as6c34cb69
To: ;tag=z9hG4bK51d6d4da
CSeq: 102 MESSAGE
WWW-Authenticate: Digest 
realm="asterisk",nonce="1550441754/[redacted]",opaque="2c504a1035f74a1d",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.25.0
Content-Length:  0


<--- Received SIP request (707 bytes) from UDP:10.0.0.1:5060 --->
MESSAGE sip:s@10.75.22.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK516e2ba7;rport
Max-Forwards: 70
From: "551212" ;tag=as6c34cb69
To: ;tag=z9hG4bK51d6d4da
Call-ID: 3e9735b4313e58f90b4b61c82f392c2e@10.0.0.1:5060
CSeq: 103 MESSAGE
User-Agent: itsp.example.com
Proxy-Authorization: Digest username="userid", realm="asterisk", algorithm=MD5, 
uri="sip:s@172.1.2.3:5060", nonce="1550441754/[redacted]", 
response="[redacted]", opaque="2c504a1035f74a1d", qop=auth, cnonce="544772bb", 
nc=0002
X-SMS-To: 5551234567
Content-Type: text/plain;charset=UTF-8
Content-Length: 1

c
<--- Transmitting SIP response (388 bytes) to UDP:10.0.0.1:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 
10.0.0.1:5060;rport=5060;received=10.0.0.1;branch=z9hG4bK516e2ba7
Call-ID: 3e9735b4313e58f90b4b61c82f392c2e@10.0.0.1:5060
From: "551212" ;tag=as6c34cb69
To: ;tag=z9hG4bK51d6d4da
CSeq: 103 MESSAGE
Server: Asterisk PBX 13.25.0
Content-Length:  0

I couldn't find any solutions to a 481 in response to a MESSAGE after
much searching.

I know MESSAGE does work, as I had it working earlier today.  I just
seemed to have messed it up adding the additional configuration to the
trunk to support incoming and outgoing calls.

Any ideas?

Cheers,
b.



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Re: [asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway

[Brian J. Murrell]

On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote:
> When you have an "identify" object configured, you should just use
> "ip" as
> the "identify_by",

But isn't ip the highest priory check in the default value of
endpoint_identifier_order and by extension, wouldn't an endpoint
without an "identify_by" implicitly be treated as "identify_by=ip"?

So isn't my existing configuring already implicitly using
"identify_by=ip"?

Cheers,
b.



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Re: [asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway

[Brian J. Murrell]

On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote:
> 
> What version of Asterisk

13.11.1

I know, I could stand to upgrade.

> and what's the value of the "identify_by"
> parameter for the endpoint?

It doesn't have one. I guess you are implying it should have one.

> When you have an "identify" object configured, you should just use
> "ip" as
> the "identify_by",

I see.  OK.  Will give that a try.

Cheers,
b.



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[asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway

[Brian J. Murrell]

I have a trunk set up for the DID from my provider:

[my_provider]
type=registration
outbound_auth=my_provider
server_uri=sip:sip.example.com
client_uri=sip:my_usern...@sip.example.com
retry_interval=60
 
[my_provider]
type=auth
auth_type=userpass
password=123456
username=my_username
 
[my_provider]
type=aor
contact=sip:sip.example.com:5060
 
[my_provider]
type=endpoint
context=from-my_provider
disallow=all
allow=ulaw
outbound_auth=my_provider
aors=my_provider
 
[my_provider]
type=identify
endpoint=my_provider
match=sip.example.com

And it registers fine:

 

==

 mytrunk/sip:sip.example.com my_provider   
Registered  


And when it gets an INVITE from my provider (192.168.0.1):

<--- Received SIP request (997 bytes) from UDP:192.168.0.1:5060 --->
INVITE sip:1235551212@10.75.22.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK06d035fd;rport
Max-Forwards: 70
From: "Fred Flintstone" ;tag=as539f9476
To: 
Contact: 
Call-ID: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060
CSeq: 102 INVITE
User-Agent: foobar
Date: Sat, 26 Jan 2019 17:40:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Fred Flintstone" 
;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 295

[SDP redacted]

It logs an error:

[Jan 26 12:40:00] NOTICE[21775]: res_pjsip/pjsip_distributor.c:525 
log_failed_request: Request 'INVITE' from '"Fred Flintstone" 
' failed for '192.168.0.1:5060' (callid: 
3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060) - No matching endpoint found

But then goes on to complete the call:

<--- Transmitting SIP response (352 bytes) to UDP:192.168.0.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
192.168.0.1:5060;rport=5060;received=192.168.0.1;branch=z9hG4bK06d035fd
Call-ID: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060
From: "Fred Flintstone" ;tag=as539f9476
To: 
CSeq: 102 INVITE
Server: Asterisk PBX 13.11.1
Content-Length:  0

[ launch into dialplan ]

So why the spurious error when it was able to complete the call?

Cheers,
b.



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Re: [asterisk-users] (NAT) direct media to host on local net when registering from external address

[Brian J. Murrell]

On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote:
> 
> The chan_sip module has this implemented under the "nat" option using
> "comedia" as I recall.

Yeah.  The help for which reads:

Send media to the port Asterisk received it from regardless
of where the SDP says to send it.

> It causes media to be sent to where media was originally received
> from.

Right.  But this is the part I don't think I'm understanding.  What is
"originally received from"?  There is no media at the point where the
media session is being set up according the the SDP is there?

> As for whether it would work or not... this all ultimately depends on
> how exactly the intermediary behaves, what is let through, what is
> altered.  There's nothing inherent in either chan_sip or chan_pjsip
> to know/care, as it's just SIP. You'd need to look at the SIP
> signaling and the SDP to understand what is happening.

I have looked.  Asterisk on my LAN is sending an INVITE to the mobile
client and since the INVITE is being sent to an external IP, it is
(correctly for all other cases) writing the external IP into the SDP
payload.  What else could/should I be looking for?

Ultimately I don't think comedia is the problem or solution because
it's not where Asterisk is sending it's media that is the problem.

The problem is that Asterisk is sending an INVITE to the client with
it's external IP address in the SDP because it sees the client as being
external and the client is then correctly trying to set up the media
session with the external IP address of the Asterisk instead of it's
internal address.

What needs to happen is that Asterisk has to know "somehow" that even
though the client registered from an external address, that it really
is internal.  Probably that "somehow" doesn't actually exist at this
point.

Let me ask this, in the REGISTER request, where is the IP address of
the client taken from?  Is it taken from the Via: header?

b.



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Re: [asterisk-users] (NAT) direct media to host on local net when registering from external address

[Brian J. Murrell]

On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
>  How is your endpoint currently configured in asterisk?

It's configured as a chan_sip peer.

> Have you tried
> rtp_symmetric to see if the endpoint sends audio to asterisk if
> asterisk
> can send audio back to the client?

That would require using chan_pjsip wouldn't it?  Not that I am opposed
to trying that.  I need to use chan_pjsip at some point to be able to
authenticate to my SIP provider for SIP SIMPLE anyway.

But will rtp_symmetric really solve the problem?  Isn't the problem the
setting up of the RTP session, so there is no address and port that it
receives from yet?

> Alternatively if your SIP Proxy is also a Media proxy you could set
> the
> media_address on the endpoint to be your proxy and let your proxy
> handle
> proxying the media to your endpoint.

The idea of sending my media out of the LAN (where I have almost zero
latency) and introducing the latency of a round trip to the proxy and
back doesn't seem like a great solution.

Cheers,
b.



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[asterisk-users] (NAT) direct media to host on local net when registering from external address

[Brian J. Murrell]

This is going to be a bit of an odd situation, but perhaps might become
more and more common (as mobile phone SIP clients utilize PUSH proxies
instead of the battery draining direct registering with SIP servers).

I have a SIP client which can be on the same RFC-1918 LAN as my
Asterisk server.  Even though it's on the same LAN as the Asterisk
server, it's registration comes from an IP address external to the LAN.
This is because the client is registering to the local Asterisk server
through a SIP proxy server that is external to the LAN.

Is there any way for Asterisk to determine that this is what is
happening and to direct/setup the media session to the client on it's
LAN address?

Put another way, even though the registration comes from an external
(NATted) IP address, I want the media connection to stay within the
LAN.

One solution of course is to add the external IP address of the SIP
proxy -- the address that the client's registration is coming from --
to localnet but that breaks the use-case of the SIP client (which is
mobile) leaving the LAN and having an external IP address.

Ultimately I am hoping there is something in the registration that will
indicate to Asterisk that even though the registration is coming from
an external IP address, the client has an internal IP address that is
on the same network as it is and sets the IP address in the SDP payload
to a local IP address.

I do realize this is fairly non-standard configuration so it might not
be possible.

Any suggestions would be welcome.

Cheers,
b.



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[asterisk-users] messagesend to SIP peer in sip.conf (or otherwise authenticated)

[Brian J. Murrell]

Hi,

I want to be able to send SIP SIMPLE messages via/to my VOIP provider
but in trying to do so with MessageSend() I am getting 401 errors back
from them, unsurprisingly.  They want such messages from me
authenticated with my account just as they would for SIP voice calls.

For voice calls, of course I use Dial(SIP/remote@peer) where remote is
the phone number I want to call and peer is my SIP provider's entry
from my sip.conf complete with account and password.

So how do I achieve the same thing with MessageSend()?  When I try to
use the same kind of recipient string I get an error:

[Oct  1 09:48:40] WARNING[3286][C-13ab]: message.c:1232 msg_send_exec: No 
message technology 'SIP/sms-number@sip-peer' found.

Using MessageSend() with raw URLs (sip:
sms-num...@sip-provider.example.com) is accepted by Asterisk but it
provides no authentication to the MESSAGE command sent to sip-
provider.example.com so is rejected.

So how can I, preferably, use my existing sip.conf entries to send SIP
SIMPLE messages, or at least provide some kind of authentication
otherwise with MessageSend()?

Cheers,
b.



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[asterisk-users] taskprocessor.c: The 'sorcery/contact-00000015' task processor queue reached 1500 scheduled tasks.

[Brian Capouch]

Running 15.1.2.  I have four devices transitioned to use pjsip.

After about 1-2 days of uptime, psjip stops accepting registrations,
and the messages log contains the entry as per the subject.

At any given time, "pjsip show contacts" only shows the four devices.

Could someone point me to a fix, short of rebooting the server every day?

Thanks.

b.

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[asterisk-users] PJSIP console messages with Zoiper

[Brian Capouch]

I'm running Asterisk 15.1.0 and in the process of converting my
various SIP endpoints to use PJSIP.

My Zoiper client causes the messages quoted below to show up on the
CLI once per minute.  Things seem to work OK, but I am curious because
there seems to be no way to suppress the messages, and there are three
per minute, clogging up the console.

Thanks.

b.

** snip **
-- Added contact
'sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096' to AOR
'8005' with expiration of 60 seconds
  == Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096
has been deleted
  == Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096
has been created
-- Contact 8005/sip:8005@64.184.17.216:51891;rinstance=21e198ab7d94a096
is now Unknown.  RTT: 0.000 msec

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Re: [asterisk-users] Removing mailbox and password prompt for voicemail

[Brian Wilson]

On Sat, Jul 30, 2016 at 7:18 AM, D'Arcy J.M. Cain <da...@vex.net> wrote:

>
> Bad, bad idea.  If you remove the password then anyone can get to the
> mailbox.


Depends on your use case, at home I have several phones and one mailbox. So
I _want_ everyone to get to the mailbox with a minimum of effort.

Brian
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Re: [asterisk-users] Removing mailbox and password prompt for voicemail

[Brian Wilson]

On Fri, Jul 29, 2016 at 10:43 PM, Nabeel <nabeelshik...@gmail.com> wrote:

> Hello,
>
> I am using Asterisk voicemail on a CentOS 7 server. I would like to be
> able to remove the 'mailbox' prompt and 'password' prompt when a user tries
> to access their voicemail. I can remove the 'password' prompt by not
> setting a password for the user, but the 'mailbox' prompt is always heard.
> Please let me know how Asterisk can be configured to remove these prompts.
>
> Nabeel
>

It sounds like you really want it to jump straight into one voicemail box
with no auth, not just removing the prompts. Try something like this

exten => *85,1,VoicemailMain(100,s)

When you dial *85, you will get voicemail for mailbox 100, and no passcode
prompt. Jumps straight into the menu "You have FOUR new messages..."

--
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Re: [asterisk-users] Remove 'Comedian Mail' Message

[Brian Wilson]

On Fri, Jul 29, 2016 at 11:01 PM, Nabeel <nabeelshik...@gmail.com> wrote:

> Hello,
>
> I would like to remove the 'Comedian Mail' name/message played when a user
> tries to access their voicemail. Please let me know how to do this.
>
> Nabeel
>

It plays a file /var/lib/asterisk/sounds/en/vm-login.wav

The recording says "Comedian Mail, please log in now." We thought we still
needed a voice prompt, so we recorded a new message.

I am not sure what happens if the file is missing, try removing it and test.
Alternatively the file could be empty (0 seconds of audio) or have a tone
or some other sound in it as feedback.

The file has to be in the right format for asterisk, after recording it on
a Windows machine, I used this command to convert it:

sox infile.wav -r 8000 -e signed-integer -b 16 -c 1 outfile.wav

"Sox" figures out the input format.

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[asterisk-users] __sip_xmit Returned -1 Invalid Argument

[Brian Wilson]

Reviving an old thread, still seeing this.

Brian Wilson wrote:
>* I've been getting slammed with these messages on my console lately.
*>>* ed -1: Invalid argument
*>* [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit:
*>* sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0
<http://192.168.1.45:0>
*>* <http://192.168.1.45:0 <http://192.168.1.45:0>> returned -1:
Invalid argument
*>* [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit:
*>* sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0
<http://192.168.1.45:0>
*>* <http://192.168.1.45:0 <http://192.168.1.45:0>> returned -1:
Invalid argument
*
Joshua Colp responded:
> It's an invalid argument because the port appears to be 0,
> which won't work. How it got to be that I don't know... a full log may provide
further enlightenment.

Currently I have 9000+ messages similar to this:

[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 725) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 725) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 727) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 728) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 729) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:06:53] WARNING[1993] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 729) to 192.168.89.172:54490 returned -2: Success
[2016-07-25 23:07:04] WARNING[14771] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 711) to 192.168.89.172:54499 returned -2: Success
[2016-07-25 23:22:33] WARNING[15801] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 711) to 192.168.89.172:54502 returned -2: Success
[2016-07-25 23:22:33] WARNING[15802] chan_sip.c: sip_xmit of
0x7f1ce8687450 (len 711) to 192.168.89.172:54502 returned -2: Success

Port is not zero on these. Error code has changed from -1 to -2

What do you mean by a "full log".

Thanks -- Brian
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[asterisk-users] 1 way audio but audio+video is fine

[Brian Wilson]

Sporadically we get 1 way audio when one party is outside our firewall.

The caller is on NAT, and it works fine most of the time. Caller can hear
the called party, same thing going the other direction. Caller can hear
called party.

Asterisk 13.9 on Debian
chan_sip with  two identical Grandstream GXV3240 SIP phones

In the CLI, I can see calls on our PBX running media on port 5004 when the
call is audio only. When the caller switches to video calls it opens ports
5004 and 5006 and everything works fine.

This does not make sense to me. It seems like audio on port 5004 should
fail either way -- adding another media channel for video should not affect
audio.

I believe I have UDP ports 5000-4 open right now on the firewall.

I also don't understand why it varies from day to day.

Any ideas on how to debug or what might be happening?

-- 
Brian Wilson, GISP
Wildsong: 707-827-0001
Mobile: 707-332-3521
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Re: [asterisk-users] SPA112 flapping

[Brian Wilson]

I gave up on a couple devices here in my home network (an old SPA2000 and a
cheap DP715 grandstream) and plugged their IP addreses into Asterisk and
turned off registration. No more annoying messages.

On Fri, Jun 17, 2016 at 2:56 PM, Mike Diehl <mdiehlena...@gmail.com> wrote:

> Hi all,
>
> I've got a device that seems to become unreachable for about 2 minutes,
> every
> hour.  From what I can tell, it isn't due to network or server issues.  Any
> ideas?
>
> TIA.
>
>
> --
> Mike Diehl
> Diehlnet Communications, LLC.
> Voice: (505) 903-5700
> Fax: (505) 903-5701
>
>
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[asterisk-users] __sip_xmit Returned -1 Invalid Argument

[Brian Wilson]

I've been getting slammed with these messages on my console lately.

ed -1: Invalid argument
[2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit: sip_xmit
of 0x7f05140803f0 (len 559) to 192.168.1.45:0 returned -1: Invalid argument
[2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit: sip_xmit
of 0x7f05140803f0 (len 559) to 192.168.1.45:0 returned -1: Invalid argument
[
etc etc etc comes in bunches of about 10, all with same IP address in that
bunch.
Does not appear to be related to one phone -- I have a mix of devices.
They appear to happen when a phone registers. The debug message "Invalid
argument" is not helping me, I wish it were more descriptive. I peeked at
the code and there is no indication what __sip_xmit() does (like all
Asterisk code I have looked at so far), looks like it might do MANY things.

It does not appear to be tied to one version, I have tried 13.7.2, 13.8.0,
13.9.1

I don't see anything glaring when I "sip set debug on" and "core set debug
10" -- just more messages that I don't understand.

Like this --

---
[2016-05-31 10:10:39] WARNING[16249]: chan_sip.c:3775 __sip_xmit: sip_xmit
of 0x7f0514050430 (len 559) to 192.168.1.45:0 returned -1: Invalid argument
Retransmitting #5 (no NAT) to 192.168.1.45:0:
NOTIFY sip:spa2000@192.168.1.45:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK0d196419
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as522feace
To: <sip:spa2000@192.168.1.45:5060>
Contact: <sip:asterisk@192.168.1.2:5060>
Call-ID: 0d6963aa7350ce3274e7af420c35daa6@192.168.1.2:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 13.9.1
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 91

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.1.2
Voice-Message: 0/0 (0/0)

--
I am guessing this is some apparently unrelated and undocumented side
effect of a setting I have changed recently but have not been successful
chasing it down so far. I have this happening on two different servers now.

-- 
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Re: [asterisk-users] asterisk admin interface

[Brian Wilson]

You are correct. Is there any open source application in that?


> According to WikiPedia, there are open-source implementations of vi
> available:
>

All my instincts say "No no! Use emacs not vi!"
but I think the OP might not know saying "vi" is intended as a joke?

For small systems using a text editor is okay... watch out for typos that
will disable your entire pbx... do "dialplan reload" when you change
extensions.conf and then scroll back looking for ERROR messages. I find the
color coding to be extremely helpful.

Likewise open "asterisk -r" immediately after starting asterisk and watch
for error messages. Be warned that sometimes the errors will lead you far
far astray. Usually they are useful.


Brian
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Re: [asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message

[Brian Wilson]

ight on the frequent
> "res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short" warnings I'm
> seeing, that'd also be appreciated.
>
> Oh - one more thing, I had to disable 2 codecs (lpc10 and ilbc) because
> they used an instruction that doesn't exist on the server (it's an oldish
> HP mini-server). I'm guessing from the above message that VM might be
> afflicted by the same issue. Presumably compiling from source will solve
> this? (I've compiled 13.7, no errors reported, but I've not tried running
> it yet)
>
> Cheers!
> Ade.
>
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Re: [asterisk-users] cannot find -lasteriskssl

[Brian Wilson]

I ran into that (and reported it) and found the easiest hack is to copy the
library manually into place, when you hit that error it's built and waiting
in main/
For example

sudo cp main/libasteriskssl.so.1 /usr/lib
cd /usr/lib
sudo ln -s libasteriskssl.so.1 libasteriskssl.so
sudo ldconfig

Then the build can complete. I can't remember if I had to do the symlink, I
think it will build anyway and then it does that step in 'make install'

I always try to build new asterisks on brand new vagrant virtual machines
that have the minimum required packages installed, and no cruft left over
from previous builds or installations. Things like this are more likely to
show up that way.

Brian


On Thu, May 5, 2016 at 10:21 AM, Michael Ströder <mich...@stroeder.com>
wrote:

> Joshua Colp wrote:
> > Michael Ströder wrote:
> >> Joshua Colp wrote:
> >>> Michael Ströder wrote:
> >>>> HI!
> >>>>
> >>>> I'm trying to compile asterisk 13.8.2+ on openSUSE Linux but it
> fails. It seems
> >>>> file ./main/libasteriskssl.so.1 is present when it fails. Building
> 13.7.2 works
> >>>> without any problem. It fails since 13.8.0.
> >>>>
> >>>> $ ./bootstrap.sh
> >>>> $ ./configure
> >>>> $ make menuselect.makeopts;menuselect/menuselect --enable chan_ooh323
> >>>> $ make
> >>>> ..
> >>>> failure (see message below)
> >>>>
> >>>> Any hint is appreciated. Thanks in advance.
> >>> Does this problem still occur in 13.9.0-rc2?
> >>
> >> No. Build of 13.9.0-rc2 seems to work.
> >>
> >> When will 13.9.0 be released?
> >
> > Probably a week or so I'd guess, maybe sooner.
>
> Ok, then I'll wait for this release. Thanks.
>
> Ciao, Michael.
>
>
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Re: [asterisk-users] PJProject Bundled Update

[Brian Wilson]

Clever. Some packages are missing though. Should I convey this to someone?

On Thu, Mar 31, 2016 at 12:28 PM, George Joseph  wrote:

>
> ​Run ./contrib/scripts/install_prereq.  I think your'e missing the
> python-dev package.  I'll update the Wiki.​
>
>
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Re: [asterisk-users] PJProject Bundled Update

[Brian Wilson]

The way I got this build to succeed last night was by using a separate
pjproject, error I get with bundle is the same after applying your patches.

First patch succeeds.
Second patch fails in 'configure'.

What I did -- I downloaded your diffs, unpacked a fresh copy of the
asterisk tarball, then applied patches.
Then did configure, make menuselect, make

Looks like you were working with an older version of configure? I think
those changes that failed are already in the release file.

Build still fails with

Makefile:117: recipe for target
'source/pjsip-apps/src/python/build/_pjsua.so' failed
make[2]: *** [source/pjsip-apps/src/python/build/_pjsua.so] Error 1
Makefile:20: recipe for target 'pjproject' failed
make[1]: *** [pjproject] Error 2
Makefile:398: recipe for target 'third-party' failed
make: *** [third-party] Error 2

I did ./configure --prefix=/usr --with-imap=system --with-pjproject-bundled

I am building on a Debian 8 virtual machine.


configure.rej
Description: Binary data
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Re: [asterisk-users] PJProject Bundled Update

[Brian Wilson]

I think that I worked around the first issue (libasteriskssl) last night.
Should have gone to bed earlier instead. :-)
Will test your patches this morning. I build on a barebones Debian 8.x
virtual machine - not "older" unless you consider "stable" = "older".

On Thu, Mar 31, 2016 at 8:57 AM, George Joseph <george.jos...@fairview5.com>
wrote:

>
> As you know, the ability to use a bundled version of pjproject was
> introduced with Asterisk 13.8.0.
>
> More info on the Asterisk Wiki
> <https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled>
>  and
> in this email thread
> <http://lists.digium.com/pipermail/asterisk-users/2016-March/288685.html>.
>
> Since then I've fixed a few issues related to older versions of Debian and
> CentOS which you can in these 2 patches.
> https://gerrit.asterisk.org//2516
> https://gerrit.asterisk.org/2449
>
> Any other feedback?  I'd like to get an idea of how many folks have tried
> it.
>
> --
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Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

[Brian ::]

12 calls isn't under any type of load.

Someone with better understanding of Asterisk internals may chime in here.

Could it be vmware timing? Is timing critical when using mixmonitor?

I've seen > 100 concurrent calls being recorded wtihout issue.



On Fri, Jan 29, 2016 at 10:39 AM, Marek Červenka <cerv...@fpf.slu.cz> wrote:

> Dne 28.1.2016 v 13:37 Brian :: napsal(a):
>
> when you say load - how many concurrent calls? Is there transcoding
> happening? sip / PRIs ? what load?
>
>
> 12  concurrent calls
>
> no transcoding
>
> SIP
>
> under 1.5 with 4x  1Ghz vcpus  (its vmware VPS)
>
>
>
> On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka <cerv...@fpf.slu.cz>
> wrote:
>
>> Dne 27.1.2016 v 17:50 A J Stiles napsal(a):
>>
>>> On Wednesday 27 Jan 2016, Marek Červenka wrote:
>>>
>>>> Dne 27.1.2016 v 13:14 A J Stiles napsal(a):
>>>>
>>>>> On Wednesday 27 Jan 2016, Marek Červenka wrote:
>>>>>
>>>>>> hi,
>>>>>>
>>>>>> i have strange problem with asterisk 13 mixmonitor, recording to wav
>>>>>> (centos6)
>>>>>> when the system is under load, there are sometimes missing syllable
>>>>>>
>>>>>> there arent BIG spikes on cpus
>>>>>> recordings are to ramdisk (/dev/shm)
>>>>>>
>>>>>> any hints?
>>>>>>
>>>>> First, try recording to a real disk  (preferrably a separate drive, so
>>>>> nothing else will be seeking the heads about; and connected by SATA,
>>>>> not
>>>>> USB, for full speed).  Does that work any better?
>>>>>
>>>> i tried before. IO is not the problem
>>>>
>>> Are you saying that it records fine when you use a real disk, but not
>>> with a
>>> ramdisk?
>>>
>>> And why are you using a ramdisk for your mixmonitor recordings?
>>>
>>>
>> i have problem in both scenarios
>> im using ramdisk because is faster and IO cannot be problem
>>
>
> --
> ---
> Marek Cervenka
> ===
>
>
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Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

[Brian ::]

when you say load - how many concurrent calls? Is there transcoding
happening? sip / PRIs ? what load?

On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka  wrote:

> Dne 27.1.2016 v 17:50 A J Stiles napsal(a):
>
>> On Wednesday 27 Jan 2016, Marek Červenka wrote:
>>
>>> Dne 27.1.2016 v 13:14 A J Stiles napsal(a):
>>>
 On Wednesday 27 Jan 2016, Marek Červenka wrote:

> hi,
>
> i have strange problem with asterisk 13 mixmonitor, recording to wav
> (centos6)
> when the system is under load, there are sometimes missing syllable
>
> there arent BIG spikes on cpus
> recordings are to ramdisk (/dev/shm)
>
> any hints?
>
 First, try recording to a real disk  (preferrably a separate drive, so
 nothing else will be seeking the heads about; and connected by SATA, not
 USB, for full speed).  Does that work any better?

>>> i tried before. IO is not the problem
>>>
>> Are you saying that it records fine when you use a real disk, but not
>> with a
>> ramdisk?
>>
>> And why are you using a ramdisk for your mixmonitor recordings?
>>
>>
> i have problem in both scenarios
> im using ramdisk because is faster and IO cannot be problem
>
> --
> ---
> Marek Cervenka
> ===
>
>
> --
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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Brian ::]

sip trace?

On Mon, Dec 21, 2015 at 6:56 PM, Luca Bertoncello 
wrote:

> Karsten Wemheuer  schrieb:
>
> Hi Karsten!
>
> > the timeout value of 15 minutes directs me to an issue with session
> > timer. Try to refuse them by putting the line
> > session-timers = refuse
> > into the general context of sip.conf. Reload the sip stack with "sip
> > reload".
>
> Sorry, I forgot to mention that...
> I already have this setting:
>
> session-refresher=uac
> session-timers=refuse
>
> > (I assume You are using chan_sip. I don't know how to disable session
> > timer in pj sip).
>
> I use chan_sip.
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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Re: [asterisk-users] GSM call routing issue

[Brian ::]

There has been some real stupid stuff going on in the inter carrier market
recently.

One of them was to attach a massive premium €2.50 per minute to calls to
Switzerland sunrise mobile for example.. This was only if your CLI was of
certain countries or invalid.

I don't know anything about voip.ms but perhaps this is whats happening
here, they could be just flagging swiss mobile as a problem due to the
potential of landing charges of 2.50 euro in excess of what they should be
paying and perhaps they're caller ID into the swiss pstn is less than
reliable hence they could fall foul of this.



On Tue, Dec 8, 2015 at 6:39 PM, Julien Sansonnens 
wrote:

> Hello everybody,
>
> I'm experiencing quite a strange issue with some voip carriers when I
> try to call my own GSM cell phone, using international route (I'm in
> switzerland).
>
> With some budget routes (value route from voip.ms for example), I get
> a fast busy tone, congestion message, or nothing at all when I try to
> call my cell phone number +4178NNN. I can call swiss landlines
> numbers, or even other cell phones without any problem.
>
> voip.ms support wrote me that my phone number is "showing as premium
> number". To avoid fraud, there are no routes to this kind of number,
> so the call cannot be made.
>
> I'm using a regular cell phone account, and I never ordered premium
> services. My number lookis like every other cell phone number out
> there. I never experienced any routing problem with my cellphone,
> before I tried to call it from voip providers.
>
> Can someone explain me what's this "premium number" thing about my
> very regular and usual cell phone number ? How can I change this ? As
> voipms carrier got the information from some database, who should I
> ask to remove my number from the list of premium numbers ? I guess I
> should ask my telephone operator, but of course nobody's is
> understanding what I'm talking about when I call the support...
>
> Thanks a lot, best regards, Julien
>
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Re: [asterisk-users] Dialing a call back out on same SIP trunk as it came in

[Brian ::]

add a pause in the dialplan for a second then proceed..



On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield 
wrote:

> In article <20151125133008.6369360.14455.17...@gmail.com>,
> Israel Gottlieb  wrote:
> > Try putting progress instead of answer
>
> Yes, I tried Progress already, and it didn't help. But thanks for
> the suggestion!
>
> Tony
>
> > I have a puzzling situation, and would be grateful for any insight.
> >
> > I have a dialplan that forwards an incoming call out to another
> > number via the same SIP trunk as it came in on. e.g.
> >
> > [from-siptrunk]
> > exten => 0123456789,1,NoOp
> > exten => 0123456789,n,Dial(SIP/siptrunk/0987654321)
> >
> > Now, if I use a different SIP trunk for the outbound call, than the
> > inbound call came on, the call is set up fine - the Answer signal from
> the
> > called party gets propagated back to the caller, and they can hear each
> > other.
> >
> > But if the outbound SIP trunk is the same as the one the call came in on,
> > the caller doesn't hear any progress, and has no notification of when the
> > call was answered. Neither can the parties hear each other.
> >
> > I have tried this on two different machines using two different SIP
> > providers.
> >
> > However, if I change the above NoOp to be Answer(100), i.e. answer the
> > inbound call before placing the outbound Dial, the caller hears progress
> > and when the called party answers, they hear each other fine.
> >
> > Of course, if the called party is busy, the caller just hears in-band
> > busy tone, as the caller's inbound call was already answered.
> >
> > Can anyone explain why I need the Answer? It feels wrong that I should.
> >
> > The siptrunk entry contains canreinvite=no and directmedia=no.
> >
> > The version of Asterisk on these boxes is 10.5.1, if that's relevant.
> >
> > Thanks for any insight!
> >
> > Cheers
> > Tony
> >
> > --
> > Tony Mountifield
> > Work: t...@softins.co.uk - http://www.softins.co.uk
> > Play: t...@mountifield.org - http://tony.mountifield.org
> >
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> >
>
>
> --
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>
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Re: [asterisk-users] SIP calls dropping at 15 minutes

[Brian ::]

probably opensips isn't forwarding the re-invite to the endpoint.. set
re-invites up and run sip tracing on your opensips and asterisk box and see
what happens when the reinvites arrive.

On Sat, Nov 21, 2015 at 8:10 PM, Steve Edwards 
wrote:

> On 11/20/15 11:13 AM, Steve Edwards wrote:
>>
>
> I have a problem where SIP calls from some providers are dropping at 15
>>> minutes.
>>>
>>> The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS
>>> sends calls to an Asterisk server.
>>>
>>
> 1) Is a 'ds_select_dst()' followed by a 'forward()' the right way to route
>>> calls in OpenSIPS? It works most of the time.
>>>
>>> 2) Can (or should) I configure Asterisk to not send the INVITE at 15
>>> minutes?
>>>
>>
> On Sat, 21 Nov 2015, Andres wrote:
>
> Looks like session timers are kicking in and a Re-Invite is being sent. I
>> would disable them in sip.conf and try again:
>>
>> session-timers=refuse
>>
>> http://doxygen.asterisk.org/trunk/sip_session_timers.html
>>
>
> 3) Should OpenSIPS be responding differently to the INVITE at 15 minutes?
>>>
>>
> This appears to work, but it feels wrong. Shouldn't I be configuring
> Asterisk or OpenSIPS  to respond or receive the re-invite correctly?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>
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Re: [asterisk-users] [OT] switches

[Brian Franklin]

If your phones support PoE,

I have had huge success with Zyxel:
http://www.amazon.com/ZyXEL-ES1100-16P-16-Port-Ethernet-Unmanaged/dp/B00
5GRETMM/ref=sr_1_3?ie=UTF8qid=1426296572sr=8-3keywords=zyxel+poe

If you want to go even cheaper, I have successfully used these as well:
http://www.amazon.com/TRENDnet-8-Port-100Mbps-Switch-TPE-S44/dp/B000QYEN
1W/ref=sr_1_10?ie=UTF8qid=1426296706sr=8-10keywords=poe+8-port


Brian Franklin
NTG, Inc. - Problem Solved

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of thufir
Sent: Friday, February 20, 2015 1:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [OT] switches

Pardon, this might be off-topic.  I'm reading:

http://en.wikipedia.org/wiki/Network_switch

For a setup of ~5 agents, would I be wrong in thinking that a generic 16
port unmanaged switch would fit the bill?

The first model to come up for me in an Amazon search is:

http://support.netgear.com/product/fs116



Is this a reasonable choice?  Would I be wrong in thinking that most any
Fast Ethernet switch would be fine for Asterisk?



thanks,

Thufir


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Re: [asterisk-users] allo.com gsm card with AsteriskNOW

[Brian]

On Sun, 19 Oct 2014 08:20:40 +
Nicolas Pham Van Huyen nico...@phamvanhuyen.com wrote:

 
 Dear all,
 
 I'm searching someone who already installed allo.com gsm card 
 with AsteriskNOW.
 
 I installed the hardware in my new server, but when running
 dahdi_genconf always have the no span message.
 I I tried to install the driver according to allo.com doc, but it
 seems not up to date, and maybe done more specifically for an
 appliance..
 
 [root@localhost ~]# dahdi_genconf
 Empty configuration -- no spans
 Empty configuration -- no spans
 Empty configuration -- no spans
 
 Thanks for help
 Nico
 
  
 David Duffett wrote :
 Step 1 would be an 'lspci' on the Linux command line to see if the
 Linux box recognises the card Step 2 would be to ensure that your
 DAHDI version is new enough to work with the card
 
 Yes , the card is present :
 04:00.0 Bridge: PLX Technology, Inc. Device d44e (rev 01)
 
 Jg wrote :
 Don't they have a kernel module that communicates with the card on
 one and with DAHDI on the other side? The first steps are probably
 to check with lspci whether the card is detected and then make sure
 the allo module is loaded.

It's also important to check for any variables needed to pass to the
module at load time: 'modinfo module_name' should return any
available options.

 
 They have a package 
 http://www.allo.com/firmware/gsm-card/chan_allogsm-1.1.2_P2.tar.gz
 
 with an installer inside (install.sh), but it is turned as full
 installer :
 - download and install  dahdi-linux-complete-2.5.0+2.5.0.tar.gz
 - seems makes some compilation for the driver 
 - finishes by proposing to install asterisk 
 
 As I installed AsteriskNOW before, it makes no sense to install
 Asterisk again, I just need to install the driver
 
 In the package I can found src folders, but I don't know how to use
 them to build the module manually. The installer seems building
 allog4c.ko file but it doesn't works installed by this script
 
 I guess it is just a basic operation to know how to compile the
 module and load it every time the system boots Then how to configure
 the channels  chan_allogsm ..


You'll need to install the basics like kernel headers and build
essentials to compile any modules for your running kernel. Once
properly compiled and loaded you should see lspci list which kernel
modules are used by others.
 
 Thanks for help :-)
 Nico
 

Good luck HTH! ;)

Brian

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Re: [asterisk-users] if statement recording - after hours

[Brian LaVallee]

There are multiple ways to do time-of-day routing.
ExecIf w/ IFTIME, GotoIfTime, and ExecIfTime.
I put some examples below.


Sincerely,
Brian LaVallee


On 9/12/14, 10:05, Eric Wieling wrote:

See ExecIf in the output of core show applications.  The IF function might be useful, 
see core show functions.   I assume the Asterisk Book also covers this.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Thursday, September 11, 2014 5:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] if statement recording - after hours

In my dial plan I have these two lines:

exten = 
_NXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten = _NXX,n,MixMonitor(${recordfilename},b)

[main]
exten = _NXX,1,NoOP(Check Time)
 same = GotoIfTime(9:00-17:00,mon-fri,*,*?open:closed)
[open]
exten = _NXX,1,NoOP(Normal Call)
[closed]
exten = _NXX,1,NoOP(Take a Message)
; end

[main]
exten = _NXX,1,NoOP(Check Time)
 same = n,ExecIf(${IFTIME(9:00-17:00,mon-fri,*,*?1:0)}?Goto(closed))
 same = n,NoOp(Process Normal Call)
[closed]
exten = _NXX,1,NoOP(Take a Message)
; end

[main]
exten = _NXX,1,NoOP(Check Time)
 same = n,ExecIfTime(9:00-17:00,mon-fri,*,*?open)
 same = n,NoOp(Take a Message)
[open]
exten = _NXX,1,NoOP(Normal Call)
; end

How to add if statement to execute these line only after let say 5pm.  To 
record conversation only after 5pm.






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[asterisk-users] FYI: Block Comments

[Brian LaVallee]

Hello,

Here's a fun issue that recently caused me some serious heartache.
Hope this helps others from making the same mistake.

Did you know that the configuration parser supports block-comments.
Like an idiot, I've been highlighting text between dashes.

[example]
; -- Most of my comments were single-line like this --
exten = _1X.,1,NoOp(This would load fine)
;
;-- I did not even use
block-comments  like
this one in my config
file --;
;
exten = _2X.,1,NoOp(Loads OK)
;
;-- But look! -- I missed a space between the semi-colon and two dashes --
exten = _3X.,1,NoOp(This would NOT load)
; -- The parser stopped loading anything past the above mistake --
; -- Missing that space started a block-comment - Arghhh! --
exten = _4X.,1,NoOp(This would NOT load either)
; -end

Guess I have to change my highlight syntax, avoiding dashes in the future.



Sincerely,
Brian LaVallee






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Re: [asterisk-users] Error: 'LENGTH' is not a recognized built-in function name

[Brian LaVallee]

On 8/11/14, 11:31, Matthew Jordan wrote:

On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
deepaksingh.ra...@gmail.com wrote:



On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:

Hi,

I modified the query in res/res_config_odbc.c.
Original:  SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'
Modified: SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s'

I rebuilt the code and installed Asterisk again. Now static realtime is
working. Should I file a bug report?


You're more than welcome to; at the same time, the number of people
using MS SQL Server with Asterisk is not tremendously high - at least
when compared with the alternatives. Unfortunately, this is one place
where making things compatible is problematic: MySQL and PostgreSQL
(which are far more likely to be used with Asterisk) along with Oracle
use LENGTH, not LEN.


Another option, just add LENGTH as a user-defined function to MS SQL.

||-- Something like this...|
CREATE FUNCTION LENGTH (@input)
||BEGIN
   RETURN ( LEN(@input) )
END|




Your solution, as it is currently, wouldn't be acceptable, as it would
cause far more problems than it would solve. About the only way I
could see solving this would be to make it configurable some place.

Given the relatively few number of people who use MS SQL Server, I
wouldn't expect this issue to receive a lot of attention without a
patch.






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Re: [asterisk-users] The plain old PBX functionality

[Brian LaVallee]

On 8/8/14, 14:05, Gergo Csibra wrote:

Hi,

back in the old analog telephony days there was digital PBX-es and
digital system phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blinks) or busy (constant
light), and the user can transfer the call with one touch (pressing
one of this button).


Because of the peer-to-peer nature of SIP, many of the digital PBX 
features can be difficult to reproduce.


If you consider where the 'brains' of the system reside, you can see the 
reason.  In the traditional digital PBX, all functionality was 
controlled by the PBX itself.  It was a Master/Slave communication 
model.  Phones were basically dumb terminals, how a button functioned 
was determined by the digital PBX.


With SIP, phones and servers are peers.  Master/Slave roles don't exist 
with SIP.  Control is determined by the device that initiated the 
session.  I will not go into the pro's and con's.  But by dialing a URL, 
it's possible to entirely exclude 'the server' from a call.




I search this functionality in Asterisk. What versions, and what
extension functions (or other settings), and what VoIP phones can do
this?
The key thing in the SIP architecture to understand, the server DOES NOT 
control the phone.  How a button functions depends on how each 
individual phone is configured.  How a phone reacts to an instruction, 
depends on how the phone is configured.


While it's possible to host a phone configuration template on the 
Asterisk server for all phones to use, it's actually independent from 
the Asterisk software.


Depending on the make/model of the phone, most of the basic features 
(hold, transfer, redial) are available by default.  To duplicate the 
digital PBX features you're looking for, will involve two groups of 
settings.  Configuration on the server -and- configuration on the phone.


SIP phones are NOT dumb terminals, you have to configure them to operate 
how you want.



Sincerely,
Brian LaVallee



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Re: [asterisk-users] OPTIONS Request without username - Forbidden

[Brian LaVallee]

Hi Rafael,

It's nothing to worry about -and- you might not be able to fix it.  But
it's nothing to worry about.

--

Asterisk is using OPTIONS like a ping, qualify=yes.  Since 403 is a
*valid* SIP reply, the remote SIP service is considered reachable.

My carrier replies with 405 Method Not Allowed, but it still indicates
the SIP connection is up and working.

--

Some carriers do not support OPTIONS.  This is normally due to a proxy
or other security mechanisms.

Remember, OPTIONS is a request for what commands will be accepted.
Sometime, you just don't want to advertise that kind of information.

--

Check an INBOUND call (INVITE) and it will typically show what the
carrier allows.  If OPTIONS is not listed, there's nothing you can do.


IP CARRIER_IP.sip  LOCAL_IP.sip: UDP, length 870
E.@.9.9:=...j.p.n$BINVITE sip:212555@LOCAL_IP:5060 SIP/2.0
Via: SIP/2.0/UDP
CARRIER_IP:5060;branch=z9hG4bKdac2492a2a1a086867cfb73fb2b5c8ac
Via: SIP/2.0/UDP PROXY_IP:5060;branch=z9hG4bK09B55db052ffec696bd
From: sip:212555@PROXY_IP:5060;tag=gK094dc1e4
To: sip:212555@CARRIER_IP:5060;tag=as2953dd14
Call-ID: 1980326667_35899190@PROXY_IP
CSeq: 7852 INVITE
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,PRACK,UPDATE
snip
Accept: application/sdp


Sincerely,
Brian LaVallee



On 6/25/14, 11:30 PM, Rafael Visser wrote:
 Hi gurus!!!
 
 I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn
 Every minute asterisk sends an OPTION Request, i beleived that it's related
 to qualify functions.
 The every minute annoyng answer of the pstn is 403 Forbidden.
 Some people told that asterisk is not sending the username in the OPTION,
 required by the pstn.
 
 
 Taking a look of the example of rfc3261.txt (pg 67), we found carol, so
 it makingme see that i am missing some config.

  OPTIONS sip:ca...@chicago.com SIP/2.0
   Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
   Max-Forwards: 70
   To: sip:ca...@chicago.com
 
 
 
 Is it wright?
 How can i instruct FREEPBX to send the username in the option request?
 
 Sorry for this silly question but a found no answer googling.
 
 
 
 Thans in advance.
 rv
 
 
 
 This is the debug of the case
 
 
 Reliably Transmitting (NAT) to 201.217.31.XX:5060:
 OPTIONS sip:201.217.31.10 SIP/2.0
 Via: SIP/2.0/UDP 18x.16.204.XXX:6060;branch=z9hG4bK1d8715df;rport
 Max-Forwards: 70
 From: Unknown sip:59x212376...@186.16.204.xxx:6060;tag=as4491c6af
 To: sip:201.217.31.10
 Contact: sip:59x212376...@18x.16.204.xxx:6060
 Call-ID: 4f02699e2632410c359e1ee43a021...@186.16.204.xxx:6060
 CSeq: 102 OPTIONS
 User-Agent: FPBX-2.11.0(1.8.25.0)
 Date: Wed, 25 Jun 2014 13:47:19 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH
 Supported: replaces, timer
 Content-Length: 0
 
 
 --- SIP read from UDP:201.217.31.XX:5060 ---
 SIP/2.0 403 Forbidden
 Via: SIP/2.0/UDP
 18x.16.204.XXX:6060;received=18x.16.204.XXX;branch=z9hG4bK1d8715df;rport=5060
 From: Unknown sip:59x212376...@18x.16.204.xxx:6060;tag=as4491c6af
 To: sip:201.217.31.XX;tag=aprqngfrt-nm50ea1c6
 Call-ID: 4f02699e2632410c359e1ee43a021...@18x.16.204.xxx:6060
 
 CSeq: 102 OPTIONS
 
 
 This is the peer.
 
 
   * Name   : desde-XopaXo-2376XXX
   Secret   : Set
   MD5Secret: Not set
   Remote Secret: Not set
   Context  : from-trunk
   Subscr.Cont. : Not set
   Language :
   AMA flags: Unknown
   Transfer mode: open
   CallingPres  : Presentation Allowed, Not Screened
   Callgroup:
   Pickupgroup  :
   MOH Suggest  :
   Mailbox  :
   VM Extension : *97
   LastMsgsSent : 32767/65535
   Call limit   : 0
   Max forwards : 0
   Dynamic  : No
   Callerid :  
   MaxCallBR: 384 kbps
   Expire   : -1
   Insecure : port,invite
   Force rport  : Yes
   ACL  : No
   DirectMedACL : No
   T.38 support : No
   T.38 EC mode : Unknown
   T.38 MaxDtgrm: -1
   DirectMedia  : No
   PromiscRedir : No
   User=Phone   : No
   Video Support: No
   Text Support : No
   Ign SDP ver  : No
   Trust RPID   : No
   Send RPID: No
   Subscriptions: Yes
   Overlap dial : Yes
   DTMFmode : rfc2833
   Timer T1 : 500
   Timer B  : 32000
   ToHost   : 201.217.31.10
   Addr-IP : 201.217.31.10:5060
   Defaddr-IP  : (null)
   Prim.Transp. : UDP
   Allowed.Trsp : UDP
   Def. Username: 595212376458
   SIP Options  : timer
   Codecs   : 0xe (gsm|ulaw|alaw)
   Codec Order  : (ulaw:20,alaw:20,gsm:20)
   Auto-Framing :  No
   Status   : OK (36 ms)
   Useragent:
   Reg. Contact :
   Qualify Freq : 6 ms
   Sess-Timers  : Accept
   Sess-Refresh : uas
   Sess-Expires : 1800 secs
   Min-Sess : 90 secs
   RTP Engine   : asterisk
   Parkinglot   :
   Use Reason   : No
   * Name   : desde-XopaXo-2376XXX
   Secret   : Set
   MD5Secret: Not set
   Remote Secret: Not set
   Context  : from-trunk
   Subscr.Cont. : Not set
   Language :
   AMA flags: Unknown
   Transfer mode: open
   CallingPres

Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info

[Brian LaVallee]

There seem to be a number of places number presentation could go wrong.
 Since the PRI 'used to' show toll-free numbers correctly, you need to
look at the gateway.

Can you debug the ISDN message on the gateway?  See how the toll-free is
being sent to the carrier.

Since you are looking to manipulate the ISDN message via SIP, it all
comes down to how the gateway handles the desired functions.


Sincerely,
Brian LaVallee

On 6/26/14, 11:24 PM, Positively Optimistic wrote:
 We're using a Earthlink PRI converted to SIP via a MediaGateway.   I assume
 the mediagateway will convert the headers to something that PRI can
 understand.
 
 
 On Thu, Jun 26, 2014 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote:
 
 It depends on your carrier.With some carriers, such as Verizon SIP,
 you do this using P-Asserted-Identity.  With Verizon SIP, if they can’t
 figure out how to bill the call, it will be rejected.

 With Level 3 SIP, you can use From: or PAID but if the number you present
 to them is not on your account, they will bill the call against the
 “failsafe” number on your account.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Positively
 Optimistic
 *Sent:* Thursday, June 26, 2014 10:11 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] CLID Presentation  Billing Number |
 Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info



 We would like to present a toll free CallerID when making outbound toll
 calls.  In the past, when our PRIs were directly connected to a Nortel
 CS1000 we could do this, without issue.  Now that the PRIs are front ended
 by a mediagateway facing asterisk, we can no longer do this.



 Is it possible to set the billing number via a SIP header and set what
 should be presented as callerid as another header for presentation?



 We can't possibly be the only people in the world that has faced this
 challenge.   Searching the internet has provided limited results.

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[asterisk-users] Multiple Servers: Multiple Peers: call-limit

[Brian LaVallee]

I would like to enforce call-limit across multiple servers.  Is there
any way to pass a call-limit variable between servers 01  02, as shown
below?  Use a global call-limit between multiple servers and peer
connections.

A -- 01 -- Z
A -- 02 -- Z

A is using round-robin to reach Z, but in the event that 01 or 02 fail,
I want the full call-limit available to A.  The call-limit is only
applied between A and the middle servers.  For the sake of discussion,
let's say call-limit=10 for both, and the total limit should also be 10.

Since my round-robin configuration will fall-back to the other server,
calls can reach a maximum of 20.  Not a state I want to allow.

#server_a_extensions.conf
[SERVER01]
exten = _X.,1,NoOp(Use: First Server)
 same = n,Dial(SIP/A-to-01-to-Z/${EXTEN})
 same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER02,${EXTEN},1)
 same = n,NoOp(yes, it's incomplete)

[SERVER02]
exten = _X.,1,NoOp(Use: Second Server)
 same = n,Dial(SIP/A-to-02-to-Z/${EXTEN})
 same = n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?SERVER01,${EXTEN},1)
 same = n,NoOp(yes, it's incomplete)

I've though about passing the variable between the middle servers in a
SIP message, side communication channel.  But, hoping there might be a
simpler solution.


Sincerely,
Brian LaVallee



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Re: [asterisk-users] T1 Card RED ALARM

[Brian LaVallee]

Since there are a number of setting that could be causing the alarm,
AMI/B8ZS, SF/ESF, etc...

Start with a loop-test, make sure the card can communicate with itself
(using the current settings).

Connect the following pins:

01 (RX-) -- 04 (TX+)
02 (RX+) -- 05 (TX-)


Sincerely,
Brian LaVallee



On 6/25/14, 1:16 PM, arun kumar wrote:
 Cables are workig fine in my other box.
 On 25 Jun 2014 00:46, Steve Totaro stot...@totarotechnologies.com wrote:
 
 Remember to always check your cables first.

 Thanks,
 Steve T


 On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com
 wrote:


 Thank you Josh for your valuable reply. I will do try changing the server
 and let you know what happening.


 ~Arun


 On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger joshdmetz...@gmail.com
 wrote:



 On Tue, Jun 24, 2014 at 5:25 AM, arun kumar arunvsadni...@gmail.com
 wrote:

 Hello All,

 I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do
 connect T1 lines it goes in RED. When I do connect the same line on a
 different Server (Same Model T1 Card) it works fine. How do I
 examine/diagnose my T1 Card for any hardware failures. I heard about
 loopback test , how helpful it is?

 Here are my configuration
 /etc/zaptel.conf
 span=1,1,0,esf,b8zs
 bchan=1-23
 dchan=24

 Zaptel Configuration
 ==
 SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
 24 channels configured

 Thanks
 ~Arun


 It could still be some sort of system config issue, even if you think
 everything is configured the same.  Have you tried moving the T1 card from
 the Bad system to the good system?  That will at least help narrow down
 if it's a bad card / port, or a config issue.

 -Josh

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Re: [asterisk-users] Play announcement only once in a Call Queue after 10 seconds

[Brian LaVallee]

Hi Jonas,

While I don't work with queues, but you could playback announce-holdtime
before putting the caller into the queue.

exten = _X.,1,NoOp(Post Queue Announcement)
 same = n,Answer()
 same = n,Wait(10)
 same = n,Playback(announce-holdtime)
 same = n,Queue(real_queue)


Brian



On 6/25/14, 10:11 PM, Jonas Kellens wrote:
 Hello,
 
 how can I create the following scenario :
 
 I have a Call Queue and I want to play an announcement, but only once
 after about 10 seconds.
 
 The current option |periodic| |-| |announce| |-| |frequency| keeps on
 playing the announcement indefinitely. (it should have an option 'once'
 like the option |announce-holdtime|)
 
 
 
 Kind regards,
 
 Jonas.
 
 
 



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Re: [asterisk-users] Maximum number of users

[Brian LaVallee]

Hi Bilal,

Assuming you have the latest hardware, sufficient memory, cpu, etc...
The key to determine the maximum number of users comes down to the 
office type, RTP path, network interface, and primary codec used.


First we need to determine the over-subscription rate, how many people 
will be using the phones at any given time.


For a call center, the ratio is 1:1.
For a normal office, the industry standard is 4:1.
{This ratio is also used to determine the number of PSTN channels you 
will need too}


Will the PSTN connections be Digium card(s) in your server or external 
gateway(s)?

Assuming Diguim card(s), the RTP will be going through your server.

Determine the network interface.  10/100/1000baseT
Then we need to consider the largest codec used, and divide the 
available bandwidth by the typical packet size.


µ-law/A-law is roughly 80 kbps, so we can support 128/1280/13107 audio 
streams.
Divide that by 2 (just to be safe) and allow RTP in both directions. 
64/640/6553


Now multiple the result by the over-subscription ratio.  4:1 = 
256/2560/26212


So we see that the maximum number of users is 2560 for a normal office 
when there is a 100baseT NIC in your Asterisk server.
You would also need to have 640 channels (28 T1 PRI's) connecting to the 
PSTN.
/Using SIP trunks to connect to the PSTN through the same 100baseT NIC 
will reduce the maximum number of users you can support./


The real challenge is not supporting thousands of users (IP Phones), 
it's connecting a sufficient number of PSTN connections to support those 
users.



Sincerely,
Brian LaVallee




On 12/18/13, 11:45 PM, bilal ghayyad wrote:

Hello;

Can someone advise me what is the maximum number of users (IP Phones) 
that can be supported by asterisk 1.8 or later?


Regards
Bilal




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Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

[Brian LaVallee]

Are you looking for something like this?

Note: This will continuously go between the two trunks until the caller 
hangs up, can be fixed by adding loop counter.


;
; extensions.conf
;
[LOADBALANCE]
exten = _X.,1,NoOp(Connect to least used trunk)
; - show active count
exten = _X.,n,NoOp(Calls: 
${GROUP_COUNT(TRUNK01CNT)}/${GROUP_COUNT(TRUNK02CNT)} of 
${MATH(${GROUP_COUNT(TRUNK01CNT)}+${GROUP_COUNT(TRUNK02CNT)},int)})

; - goto least used trunk
exten = _X.,n,GotoIf($[${GROUP_COUNT(TRUNK01CNT)}  
${GROUP_COUNT(TRUNK02CNT)}]?TRUNK02,${EXTEN},1:TRUNK01,${EXTEN},1)

;
;
[TRUNK01]
exten = _X.,1,NoOp(Using Trunk 01)
; - set trunk used counter
exten = _X.,n,Set(GROUP()=TRUNK01CNT)
; - dial trunk
exten = _X.,n,Dial(IAX/T01/${EXTEN})
; - add loop counter to stop infinite loop
exten = _X.,n,NoOp(Use next TRUNK02is congestion or chanunavial)
; - next trunk if CONGESTION
exten = _X.,n,GotoIf($[${DIALSTATUS}=CONGESTION]?TRUNK02,${EXTEN},1)
; - next trunk if CHANUNAVAIL
exten = _X.,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?TRUNK02,${EXTEN},1)
exten = _X.,n,Hangup()
;
;
[TRUNK02]
; - same as above
exten = _X.,1,NoOp(Using Trunk 02)
exten = _X.,n,Set(GROUP()=TRUNK02CNT)
exten = _X.,n,Dial(IAX/T02/${EXTEN})
exten = _X.,n,NoOp(Use next TRUNK01is congestion or chanunavial)
exten = _X.,n,GotoIf($[${DIALSTATUS}=CONGESTION]?TRUNK01,${EXTEN},1)
exten = _X.,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?TRUNK01,${EXTEN},1)
exten = _X.,n,Hangup()
;
; - end

On 12/14/13, 4:41 PM, Muhammad Usman wrote:

Friends let me define the scenario please;
Scenario:
2 asterisk servers (A  B) are connected using 05 IAX2 trunks between 
them. The machine A is running asterisk  Openvpn server in TUN mode 
(5 instances with difference IP addresses for clients). The machine B 
is running asterisk with 05 OpenVPN clients using 05 bandwidths. The 
IAX trunks are established between each pair of P-2-P ip address of 
machine A (The OPENVPN Server)  machine B (The Openvpn client).

Requirement:
Required dial plan configuration at machine A for incoming calls from 
VoIP Switch/VOS which can forward the calls to IAX2 trunks in round 
robin fashion like Load Balancing. If any trunk goes down it starts 
forwarding the traffic to other available trunks  when it gets UP the 
dialplan should perform as desired. Like L.B  Fail-over scenarios.



On Fri, Dec 13, 2013 at 8:52 PM, Hans Witvliet aster...@a-domani.nl 
mailto:aster...@a-domani.nl wrote:


On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
 On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
  Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks.
I want
  to load balance incoming calls over IAX2 trunks. If any trunk goes
  down the calls traffic will be shared with other available trunks.
  When it gets Up the script is supposed to perform as desired
i.e in
  load balance mode.

  Thanks in advance.
 

 Hans said:


 Perhaps it is possible to do the L.B. at the O.S. or network
level, and let
 all trunks appear to asterisk to one single trunk.

 Don asks:

 What's the value of load balancing multiple IAX trunks between
the same
 system pair? What resources are being balanced?

++

Perhaps the O.P. can explain about his intentions...

In some situations it makes sense though:
If you have to connect two servers, and use different kind of
infrastructure / multiple providers...

hw


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Re: [asterisk-users] Asterisk on Windows

[Brian]

On Tue, 10 Dec 2013 23:02:45 +0200
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:

 On Wed, Dec 04, 2013 at 02:12:41PM -0500, Ruddy Gbaguidi wrote:
  I never tought this is become a Linux vs Windows fight.
  We have been using asterisk on linux from a long time now and happy
  with it.
  But some of our customers who has windows in their environment want
  to use our call center software we developed on top of asterisk.
  So, the question was :
  Did anybody ever tried to isolate the asterisk SIP server/module and
  make it run under Windows ?
  Since, asterisk 12 is using pjsip (which is cross platform already),
  I tought it may be possible and wanted advices.
  
  I would love that every single customer switch to Linux and Ubuntu
  tomorrow morning but at the moment, that's not the case.
 
 There was an old half-working port of Asterisk to Cygwin which does
 run on Windows. It has not worked since at least 1.6.0 .

That's just a unix-like interface which won't address the issues the OP
has/had with running/configuring asterisk. IMHO it would probably be
even more challenging. And IIRC the OP was looking for a non emulated
solution anyway.

 Feel free to try to fix it. I suspect it won't be easy. Patches would
 be welcomed, I guess (look at what odd fixes that were accepted to
 make Asterisk build and work on OS/X).


That advice was already given by multiple posters. OS X is unix-like as
well so I fail to see what help that could be in an endeavour to port
asterisk.

 And for others: the name is [MS-]Windows. Not 'wind-blows or whatever
 name you find for it. Please respect this list. If you don't have
 anything useful to add to the thread, please refrain from replying.
 

I have to agree with the name calling part but the OP did imply that
Windows was superior and that a Windows port would be profitable. You
can't really expect to get away with that on a list devoted to an open
source application without making a complete fool out of yourself.

If it was a post regarding one of the many proprietary closed source
applications/games without a native port to Linux/BSD/OS X then it
would be a valid complaint. Having access to the source as well as
liberal licensing terms which allow porting isn't a valid complaint and
never will be.

B

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[asterisk-users] ARA: realtime: sip.conf: context

[Brian LaVallee]

The answer is probably no, but I would like someone else to confirm it.

When using The Asterisk RealTime Architecture, can context work in 
sip.conf like it does for extensions.conf ?


switch = Realtime/[context]@[family][/options]

More specific, is there a way to use a single table and limit the query 
results to a specific subset for sip.conf ?

I know this can be solved using VIEW in MySQL, but I want to avoid that.

WORKS: SELECT * FROM extensions.conf WHERE context = [context];
DOES NOT WORK: SELECT * from sip.conf WHERE context = [context];

Below, you can see how it works for extensions and does NOT work on the 
context field of the sippeers table, is there any field that can be used?



Sincerely,
Brian LaVallee


---===
;# extconfig.conf
;
[settings]
;
sippeers = mysql,database,sippeers
moresippeers = mysql,database,moresippeers
extensions = mysql,database,extensions
other = mysql,database,other
;
; end===---

---===
;# extensions.conf
;
[unique85]
;
switch = Realtime
; Equivalent to: SELECT * FROM database.extensions WHERE context = 
unique85;

; - uses current [this] context.
;
switch = Realtime/limited@extensions
; Equivalent to: SELECT * FROM database.extensions WHERE context = 
limited;

;
switch = Realtime/specific@other
; Equivalent to: SELECT * FROM database.other WHERE context = specific;
;
; end ===---

---===
;# sip.conf
;
[general]
;
switch = Realtime
; Equivalent to: SELECT * FROM database.sippeers;
;
switch = Realtime/anything@moresippeers
; Equivalent to: SELECT * FROM database.moresippeers;
; NOT EQUIVALENT TO: SELECT * FROM database.moresippeers WHERE context = 
anything;

;
; end ===---

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Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?

[Brian]

On Fri, 22 Nov 2013 13:41:45 -0500
Ron Wheeler rwhee...@artifact-software.com wrote:

 If you have no analog lines, Amazon/Rackspace/...  will probably beat 
 your local ISP on bandwidth to your SIP/IAX carrier.
 
 If your users are not in the same building as your in-house hosted 
 Asterisk, Amazon might have a lot better connectivity with your users.
 
 You certainly have a lot more flexibility in adding power to your
 setup at an Amazon.
 
 I guess that one can decide what are the critical points that need to
 be tested (call volume, call quality, user connectivity) and devise a
 test setup.
 
 Ron
 


I've setup Asterisk in the past on VMs (Linode, VMware, Xen, etc.) and
IIRC the biggest issue we had was with RTC. As in Real Time Clock
since Asterisk requires an accurate timing source. It's been a very
long time since I've dealt with Asterisk on a VM so perhaps it's not
uncommon to have the zaptel kernel modules (ztdummy among others?)
available on most VMs these days.

It's certainly an option for some use cases but not all. I'd recommend
running MTR or a similar tool to determine any latency issues along the
way. In any case good luck with your project. If anyone else has more
recent experience regarding RTC please feel free to correct me. I'm
inclined to fiddle around with a VM based Asterisk install again if
it's gotten simpler to implement.

Brian

 On 22/11/2013 1:18 PM, Todd R. wrote:
  I would have said the same thing a while back but, I can't ignore
  the fact that there have been what seems to be many
  Virtualization success stories.
 
  The idea that Asterisk just likes to be on it's own dedicated
  hardware has always caused me to prefer dedicated hardware.
 
  But, is the possibility of a single piece of hardware failing
  better than something that will likely never just flat out die?
 
  I know there are high availability solutions out there and it's not 
  that I don't have backups and disaster recovery plans in place.
 
  I just want to make things far better regarding redundancy,
  recovery and scalability and virtualization is hard to beat when
  you start talking about these things.
 
  There are definitely people/companies using virtualized Asterisk 
  solutions successfully, so I feel like it can be done.
 
  Asterisk has come a long way since I first starting messing with 
  Asterisk and so has Asterisk itself.
 
  So, I am trying to determine what is bad, what to look out for in 
  terms of virtualizing. If it's still as bad of an idea as it was
  say 5 years ago, then I need to understand why and if there is a
  work around.
 
  At this point, the benefits of virtualizing my Asterisk boxes are
  too many to count. So, if I can't find any concrete reasons to NOT
  do this beyond That's a bad idea then I am going to give it a go.
  If I do, I am looking for any advice good or bad from those that
  have gone down this road successfully or with miserable failure.
 
  My opinion all along has been Asterisk + Virtualization + Real Live 
  Production Use = BAD IDEA!
 
  Now, I am trying to figure out if that's just the opinion of an old 
  man (sort of old) who just doesn't want to accept that
  virtualization if a better way (in terms of Asterisk).
 
  So, I am hoping for people to tell me why Amazon AWS specifically
  is a good or bad idea with as much detail as possible.
 
  Thanks!
 
   To: tjrl...@live.com; asterisk-users@lists.digium.com
   Subject: Re: [asterisk-users] Amazon, Asterisk and reliability 
  beyond a hobby system?
   Date: Fri, 22 Nov 2013 13:04:44 -0500
   From: cov...@ccs.covici.com
  
   I would thinktwice about Amazon -- and virtual in general is not
   a good idea for this sort of thing. I have seen messages about
   bad results with amazon specifically.
  
   Todd R. tjrl...@live.com wrote:
  
Just checking one more time to see if anyone has an opinion on 
  this. I am primarily interested in using a cloud type setup such as 
  Amazon AWS for the redundancy, easy backup and recovery options.
  It's not about price but the idea that it will be very hard for a
  single piece of hardware to ruin my day.
   
From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Mon, 18 Nov 2013 18:33:38 -0600
Subject: [asterisk-users] Amazon, Asterisk and reliability
beyond 
  a hobby system?
   
   
   
   
Took me a while but I have finally embraced cloud computing and 
  all the benefits.
The only thing I have yet to feel comfortable about putting in
the 
  cloud is real live Asterisk boxes to be used in production. I know 
  it's being done because as far as I know Twilio is using Amazon for 
  their Asterisk boxes.
I have read all the fun articles on building hobby type systems 
  and that's all great.
What I really need to hear is from those that have deployed 
  Asterisk in Amazon or Digital Ocean and how many simultaneous calls 
  they are pushing through it and what the call quality and
  reliability has been

Re: [asterisk-users] Auto Redial Unconditional

[Brian]

On Thu, 24 Oct 2013 18:20:09 +0400
Rizwan Hisham rizwanhas...@gmail.com wrote:

 Hi All,
 I need a softphone (PC/Mobile) which does auto redial in any case
 (noanswer,  answer, busy, congestion etc) after a given time
 interval. So if the time interval was 5 secs, it would dial last
 number dialled after every hangup (or every failure to dial).
 
 Does anyone know such feature in a softphone?
 

You might be able to script something using Linphone[1]. That page
states:

  This feature is still being developed, stay connected to svn if you
 use it !
 
 What could be the applications of this tool ? For example:
 
 run VoIP calls from scripts
 from web cgi pages
 from javascript in a browser window...
 

HTH and good luck.

Brian

[1]
http://www.linphone.org/eng/documentation/guide/linphonecsh-control.html

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[asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3

[Brian LaVallee]

Hi Nick,

Going from DS1 to OC-n is a multi-step process.  Typically requiring a
hardware device to handle each MUX step.  But you can find hardware that
handles multiple MUX steps together.

VT1.5 is just a raw OC-n channel containing a single DS1.
An M13 device converts between DS3 and DS1.

A DACS (DCS or DXC) provides M13 conversion, sometimes even capable of
extracting the raw VT1.5 signal directly to DS1.

The ILEC transport option you choose really depends on the terminating
interface.  Do you want to connect with a DS3 or OC-n?

No matter what hardware you choose, you will need to convert to single
copper pairs (DS1/T1) to connect to your Asterisk boxes.  So an M13 or DCS
will be necessary to reach the DS1 level.

The device you choose depends on budget and growth expectations.  Typically
a DCS is an expensive investment, handling hundreds of DS3's. An M13 device
is typically a small unit that handles one or two DS3's.

The advantage comes when you add the 29th DS1.  With VT1.5 it's just adding
a single channel, DS3 will require another whole DS3 to get an additional
DS1.


Sincerely,
Brian LaVallee



 From: Nick Khamis sym...@gmail.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Wed, 12 Jun 2013 16:19:06 -0400
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: [asterisk-users] ILEC Interconnect
 
 Hello Everyone,
 
 We are looking to interconnect with a local ILEC over an OC-n transport layer.
 They basically gave us two options in terms of mapping the SONET to the DS3:
 
 * VT1.5s mapping
 * DS1s mapping
 
 The second option is quite clear. We would MUX the connection, and plug
 the lines into qaud t1 cads etc... The tech mentioned that with the second
 option we would also need a DACS to convert back to M13 mapping. I was
 scared to tell him that I could not follow can someone explain that to
 me kindly :).
 
 I don't know much about VT1.5 mapping. Can someone kindly explain what
 the benefits
 or lack of are in choosing that option. Also what type of additional
 equipment we
 would need?
 
 In case I have overlooked something, can you gents please tell me what
 I will need in
 terms of hardware in both cases (minus routers and switches). What we
 are looking at is:
 
 CO
 |
 |
 | OC-n
 |
 v
 DS3 MUX
 |
 |
 v
 21 Asterisk boxes with quad T1s
 
 
 Kind Regards,
 
 Nick.
 
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[asterisk-users] Initial REGISTER Request: Contains Credentials before 401

[Brian LaVallee]

My SIP provider is not happy that credentials (in the Authorization header
field) are provided in the initial REGISTER request.

The SIP provider ONLY wants the credentials AFTER rejecting the message with
a 401.

I know it's dumb, because the RFC says that the the initial REGISTER message
MAY include credentials.  If it fails, the proper authentication method is
included in the 401.  I know there is nothing wrong, it is how SIP is
supposed to work.

However I would like to keep my SIP provider from complaining.  Asterisk is
NOT SUPPORTED by the SIP provider.

Does anyone in the Asterisk community know how to avoid sending the
credentials until AFTER receiving a 401?

Any suggestions would be appreciated!


Sincerely,
Brian LaVallee

  
# ===
# sip.conf
# Asterisk 1.8.15-cert1
# ---
; 
[general]
;
; - trucated
; 
register=accountnum...@server.carrier.tld:secret:acco...@proxy.carrier.tld/
DID
;
; - end

# ===
# SIP REGISTER Dialog
# ---

IP 4.4.4.4.sip  8.8.8.8.sip: UDP, length 602
REGISTER sip:server.carrier.tld SIP/2.0
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aAaAaAaAaAaAaAa
Max-Forwards: 70
From: sip:accountnum...@server.carrier.tld;tag=as6c2d23d4
To: sip:accountnum...@server.carrier.tld
Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4
CSeq: 190 REGISTER
Authorization: Digest username=account, realm=carrier.tld,
algorithm=MD5, uri=sip:sip:8.8.8.8, nonce=1368595443265327,
response=0b833bff6d83337f9f88f6fb53bbcef6
Expires: 1800
Contact: sip:DID@4.4.4.4:5060
Content-Length: 0


IP 8.8.8.8.sip  4.4.4.4.sip: UDP, length 469
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aAaAaAaAaAaAaAa
From: sip:accountnum...@server.carrier.tld;tag=as6c2d23d4
To: sip:accountnum...@server.carrier.tld;tag=3577586129
Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4
CSeq: 190 REGISTER
Content-Length: 0
Date: Wed, 15 May 2013 05:55:29 GMT
WWW-Authenticate: Digest realm=carrier.tld, domain=sip:8.8.8.8,
nonce=1368597329273572, opaque=, stale=TRUE, algorithm=MD5


IP 4.4.4.4.sip  8.8.8.8.sip: UDP, length 602
REGISTER sip:server.carrier.tld SIP/2.0
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=BbBbBbBbBbBbBbB
Max-Forwards: 70
From: sip:accountnum...@server.carrier.tld;tag=as333ffab1
To: sip:accountnum...@server.carrier.tld
Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4
CSeq: 191 REGISTER
Authorization: Digest username=account, realm=carrier.tld,
algorithm=MD5, uri=sip:sip:8.8.8.8, nonce=1368597329273572,
response=097ee5b915cd39c1407c785fb3c06caf
Expires: 1800
Contact: sip:DID@4.4.4.4:5060
Content-Length: 0


IP 8.8.8.8.sip  4.4.4.4.sip: UDP, length 373
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=BbBbBbBbBbBbBbB
From: sip:accountnum...@server.carrier.tld;tag=as333ffab1
To: sip:accountnum...@server.carrier.tld
Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4
CSeq: 191 REGISTER
Contact: sip:DID@4.4.4.4:5060;q=0;expires=1901
Content-Length: 0
Date: Wed, 15 May 2013 05:55:29 GMT

# ===





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[asterisk-users] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan

[Brian LaVallee]

Hi Matthew,

Thanks for the response.

 From: Matthew J. Roth mr...@imminc.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Date: Wed, 15 May 2013 12:28:11 -0500 (CDT)
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Initial REGISTER Request: Contains Credentials
 before 401
 
 Brian LaVallee wrote:
 
 My SIP provider is not happy that credentials (in the Authorization header
 field) are provided in the initial REGISTER request.
 
 The SIP provider ONLY wants the credentials AFTER rejecting the message with
 a 401.
 
 I know it's dumb, because the RFC says that the the initial REGISTER message
 MAY include credentials.  If it fails, the proper authentication method is
 included in the 401.  I know there is nothing wrong, it is how SIP is
 supposed to work.
 
 Who is your SIP provider?  They need to be called out so that other Asterisk
 users can avoid them.  This tendency to flip the customer/vendor relationship
 on
 its head must be discouraged.

The SIP provider is KDDI Japan.

 However I would like to keep my SIP provider from complaining.
 
 The only thing they should complain about is if you don't pay your bill on
 time.
 
 Asterisk is NOT SUPPORTED by the SIP provider.
 
 The REGISTER request was successful so, at least from a practical standpoint,
 the provider does support Asterisk.  It would be ideal if all providers
 officially supported Asterisk, but this is just one example of how it's not
 worth trying to please everyone.

I know that the SIP provider is being overly diligent to prevent toll-fraud,
but some of their complaints about normal SIP communications have been
outrageous.  

I don't want to go into it, but KDDI has actually complained that the REPLY
to the 401 was too fast.
 
 Does anyone in the Asterisk community know how to avoid sending the
 credentials until AFTER receiving a 401?
 
 Edit the source.  I'm sorry to be blunt, but I really can't see the developers
 adding another option to sip.conf just to satisfy such a pointless request.
 
 Any suggestions would be appreciated!
 
 Ask the provider what platforms are supported.  Pick one of them and use it
 to
 configure the useragent and sdpsession options in sip.conf.  Or look for
 another provider that doesn't waste your time complaining about RFC-compliant
 behavior.  

KDDI does provide a list of supported equipment and vendors.  Specific
hardware or license based software products that quickly become cost
prohibitive.

I doubt that Asterisk will find it's way on the list any time soon.  Because
KDDI follows the traditional big telco method of interoperability, which
normally means licensing products for use on their network.

 Regards,
 
 Matthew Roth
 InterMedia Marketing Solutions
 Software Engineer and Systems Developer
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Re: [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database

[Brian LaVallee]

It sounds like phpMyAdmin is NOT on the same server as the Asterisk DB.

You will run into a couple possible issues when allowing remote MySQL access
on the Asterisk server,

You will need to set the MySQL user privileges to a specific host or a
wildcard (%).
Most common issue is the firewall, likely iptables on the Asterisk server.
Allow: tcp/3306

To test the privileges, stop iptables and try this from the local machine:
mysql --host=asterisk_server --user=myuser --password=mypass

If you can access the MySQL on the Asterisk server, adjust and restart
iptables.  Then try again with the firewall in place.

To add the Asterisk server to phpMyAdmin, see the following:
http://goo.gl/J1Py5


Brian



From: Lobna Hegazy lobna.heg...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tue, 14 May 2013 22:57:34 +0200
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Using PHPMyAdmin to remotely access Asterisk
MySQL Database

I'm successfully configured with mysql. After research, accessing Astersik
MySQL database requires 2 major steps which are: 

(1) Allow remote access. 
(2) Add the new server to my local php configuration files. 

I'm just asking for the right steps to do these two major steps.

Thank you in advance 


On Tue, May 14, 2013 at 10:22 PM, Gertjan Baarda gertjan.baa...@gmail.com
wrote:
 
 
 On Tuesday, May 14, 2013, Lobna Hegazy  wrote:
 I'm running XAMMP on Mac OS which includes  Apache, MySQL, PHP, and Perl.
 phpMyAdmin now connected to the localhost server. What I'm willing to do is
 adding asterisk database server in order to access cdr database. 
 
 Thank you for any help
 
 
 On Tue, May 14, 2013 at 10:00 PM, Gertjan Baarda gertjan.baa...@gmail.com
 wrote:
 
 
 On Tuesday, May 14, 2013, Lobna Hegazy  wrote:
 Dear All, 
 
        I'm trying to connect to Asterisk CDR database using PHPMyAdmin but
 unfortunately all my trials and searches failed. So I'd be more than
 grateful if someone helped me with right steps to do this. Kindly note that
 I'm working on a remore server that I can connect to as a root using ssh.
 
 Asterisk Version: 11.3.0
 MySQL Version: mysql-server.x86_64 0:5.1.69-1.el6_4  
 
 Please ask me for any specifications you need, thank you in advance.
 
 
 -- 
 Best Regards,
 Lobna Hegazy
 
 Can you be (a lot) more specific?
 Are you running Apache? Distro? Firewall and such.  
 
 
 -- 
 Sent from Gmail Mobile
 
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    http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 -- 
 Best Regards,
 Lobna Hegazy
 Public Relations Committee
 Bdaya l we love charity
 German University in Cairo
 
 Asterisk as such does not have a database server, it can use mysql as a
 backend for cdr. Is your cdr succesfully configured with mysql? Ergo: does is
 write the cdr entries into mysql?
 
 
 -- 
 Sent from Gmail Mobile
 
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-- 
Best Regards,
Lobna Hegazy
Public Relations Committee
Bdaya l we love charity
German University in Cairo


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[asterisk-users] qualify=yes: OPTIONS: How to Change?: `From: asterisk`

[Brian LaVallee]

My Google-Fu skills have failed me, I have not been able to find a solution
to the problem I am facing.

asterisk + from + asterisk + options +  qualify != what I am looking for

--

When qualify is enabled on a trunk, the From line shows asterisk.  See the
SIP message below.

I would like to keep qualify enabled without sending the other end any
reference to asterisk.
Can anyone point me to a setting that will change or remove `²asterisk²`
from `FROM:` in the OPTIONS message?


Thanks,
Brian LaVallee

--

/etc/asterisk/sip.conf (Asterisk 1.8.15-cert1)
[general]
; - Truncated
[TRUNK]
; - Truncated
qualify=yes
;
; end

--

IP 4.4.4.4.sip  3.3.3.3.sip: UDP, length 573
OPTIONS sip:server.carrier.tld SIP/2.0
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aBcDeFgHiJkLmNo;rport
Max-Forwards: 70
From: asterisk sip:accountid@4.4.4.4;tag=as1832334c
To: sip:server.carrier.tld
Contact: sip:accountid@4.4.4.4:5060
Call-ID: f80a4ad87fee7c9fdc19b7769495fdb5@4.4.4.4:5060
CSeq: 102 OPTIONS
Date: Thu, 09 May 2013 07:22:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0



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[asterisk-users] Thanks! qualify=yes: OPTIONS: How to Change?: `From: asterisk`

[Brian LaVallee]

Thanks Jeremy!

 
 On 5/9/13 8:21 PM, Brian LaVallee wrote:
 When qualify is enabled on a trunk, the From line shows asterisk.  See the
 SIP message below.
 
 I had the same annoyance/issue.  fixed it in
 https://issues.asterisk.org/jira/browse/ASTERISK-17616

That's looks like the problem I was seeing.

 the patch was included in 1.8.9 rc1.

I've been trying to stick to the current AsteriskNow as my base standard,
hopefully the fix was applied to the latest version.

Thanks again!


Brian LaVallee
 
 -- 
 
 Jeremy Kister
 http://jeremy.kister.net./
 
 
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[asterisk-users] Disable transcoding

[Brian]

Hello
I use asterisk realtime, and I can set the order of codec preference on my 
realtime allow column.

If I could disable transcoding, then I can always ensure a passthrough of the 
common codec from origin to destination without transcoding (expensive on CPU) 
- 
and more or less, force the codec to use by setting the codec preference

So, can I disable transcoding? 


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Re: [asterisk-users] Disable transcoding

[Brian]

yes, but thats no good, as the codec order will be ignored. I need to be able 
to 
allow Asterisk to choose the code from the order, without forcing a single 
codec,


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[asterisk-users] Getting hold status via AMI ?

[Brian Camp]

Hi,

Is there any way to tell via the AMI or console if a given SIP channel is
hold?   ChanIsAvail in the dialplan appears to have a 'hold' status, but
AMI and CLI commands tend to return 'in use', which is the same state as a
regular active call.

Thanks

-Brian
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Re: [asterisk-users] Polycom firmware 4.0.1 and paging

[Brian ipt]

On Fri, Feb 10, 2012 at 10:30 PM, Mike l...@net-wall.com wrote:

 Hi,

 ** **

 I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer
 simply stopped functioning. I can downgrade and make it work, upgrading
 kills it again. There obviously is a difference in how the newer firmware
 is treating this auto answer sip header.

 ** **

 Can anybody tell me if they have Polycom firmware 4.x.x working with
 auto-answer/paging? Just so I know it’s worth my time to investigate, as
 opposed to knowing it`s a Polycom firmware bug? If so, did you have to make
 any changes to the SIP header sent to make Polycom phones auto answer? ***
 *

 ** **

 Regards,

 ** **

 Mike

 **



Hi Mike,

Is there a compelling reason to put version 4.0.1b on these phones?

Brian

 **

 ** **

 ** **

 ** **

 ** **

 ** **

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[asterisk-users] FILTER function and multiple ranges?

[Brian J. Murrell]

I am trying to use the FILTER() function to strip out / from a CID
name.  I have the following in my extensions.conf where I want to
perform the filtering:

exten = s,n,Set(NAME=${FILTER(\x20-\x2e\x30-\7d,${DIAL_NAME})})

However, when ${DIAL_NAME} is, say, J  J DOE the string resulting
from the FILTER() is.  It would appear that only the first range
is being considered.  Do I have the syntax for FILTER() correct?  If so,
is the result that I am seeing the intended behavior or a bug?

Cheers,
b.



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Re: [asterisk-users] Occasional call from asterisk

[Brian Henning]

Bruce B said:



We experience exact same thing on DAHDI with Sangoma USB FXO device on short
circuited lines. Phantom calls are actually due to a short in the lines that
happen occasionally.

 

-Bruce

Also, Warren Selby said:

 

I've seen this on cases where a phantom call comes in on a DAHDI channel -
these calls were the results of faulty wiring on the part of the telco.
Check your logs for any errors on your DAHDI channels around the time of the
ghost calls.  

It could also be a case of someone calls in and then hangs up before the
call is actually passed to asterisk, and the telco is just slow to hangup
the call.  
 

H.  I do see this in the /var/log/asterisk/messages log:

 

[Apr  5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)...

[Apr  5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)...

[Apr  5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 17 (Polarity
Reversal)...

[Apr  5 00:05:41] WARNING[2400] chan_dahdi.c: Detected alarm on channel 3:
Red Alarm

[Apr  5 00:05:42] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 3

[Apr  5 00:05:51] WARNING[2400] chan_dahdi.c: Detected alarm on channel 2:
Red Alarm

[Apr  5 00:05:53] NOTICE[2400] chan_dahdi.c: Alarm cleared on channel 2

 

.and it appears to coincide directly with an 'asterisk' entry in my SIP
phone's missed call log.

 

Our wiring is sketchy; this is known at our facility.  Some years ago a
backhoe severed the entire trunk and the repair work was of questionable
quality.  Also our service entry point / punch-down area is a rat's nest
(one building and service is shared by three companies).  I guess I can
chalk this behavior up to the wiring.

 

Thanks for the input!

 

Cheers,

~Brian

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[asterisk-users] Occasional call from asterisk

[Brian Henning]

Hi,

Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup).  Can anyone offer some insight?  Here's
relevant snippets from my extensions.conf and Master.csv log:

This line shows up in Master.csv:

,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
21:37:05,2011-04-07 21:37:16,2011-04-07
21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,

Here's [inbound] from extensions.conf:
[inbound]
exten = s,1,Answer
exten = s,n,Ringing
exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
exten = s,n,Goto(1-${DIALSTATUS},1)
exten = 1-ANSWER,1,Hangup
exten =
_1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
exten = _1-.,n,Goto(2-${DIALSTATUS},1)
exten = 2-ANSWER,1,Hangup
exten = _2-.,1,Voicemail(499@default,u)
exten = _2-.,2,Hangup

The idea is that first 504 and 506 ring, then if neither of them answer,
everyone rings.  Works great most of the time.

I have a hunch that maybe this happens if the inbound caller hangs up while
the first Dial() is ringing, but I would've expected to see the first Dial
(to 504 and 506) show up in the Master.csv log, and it's not there.  (The
preceding line of the log is a call from almost an hour earlier).  In that
case though I'd expect to see 1-CANCEL in the log instead.  Perhaps if the
caller happens to hang up right between the two Dial() commands?..

As an aside, the Set(CALLERID...) bit doesn't work.  The idea was to prepend
a 9 so that a SIP user could use the redial feature of the phone's call
log to return a missed call (automatically including the 9 for outside
line).  Unfortunately the 9 does not get prepended.

Thanks in advance for any and all advice!
~Brian

-- 
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/\Pine Research Instrumentation 
   //\\   5908 Triangle Drive 
  ///\\\  Raleigh, NC 27617 
  USA 
|| 
||phone: 919.782.8320 
  fax:   919.782.8323 
  email: bhenn...@pineinst.com 
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[asterisk-users] Hang using Festival application

[Brian Henning]

Hello,

Suppose a dialplan such as:

exten = 6004,1,Answer
exten = 6004,n,Wait(1)
exten = 6004,n,SayDigits(1)
exten = 6004,n,Festival(This is a test of Festival)
exten = 6004,n,Hangup

When watching in the CLI, I see this:
  == Using SIP RTP CoS mark 5
-- Executing [6004@internal:1] Answer(SIP/505-0004, ) in new
stack
-- Executing [6004@internal:2] Wait(SIP/505-0004, 1) in new
stack
-- Executing [6004@internal:3] SayDigits(SIP/505-0004, 1) in new
stack
-- SIP/505-0004 Playing 'digits/1.gsm' (language 'en')
-- Executing [6004@internal:4] Festival(SIP/505-0004, This is a
test of Festival) in new stack
  == Parsing '/etc/asterisk/festival.conf':   == Found
ps-pbx*CLI

... and nothing more.  Nothing happens after  == Parsing ..., and the SIP
channel gets stuck open even after I physically hang up the extension (will
not respond to a hangup request, can only be eliminated by restarting
asterisk).  I hear one in the phone and then silence.

Versions:
festival: Festival Speech Synthesis System: 2.0.95:beta April 2010
asterisk: Asterisk 1.6.2.9-2+squeeze1
OS: Debian Squeeze 64 bit
~# uname -a
Linux ps-pbx 2.6.32-5-amd64 #1 SMP Wed Jan 12 03:40:32 UTC 2011 x86_64
GNU/Linux

These are all unmodified packages obtained via aptitude.

What am I getting wrong?

Many thanks,
~Brian


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  USA 
|| 
||phone: 919.782.8320 
  fax:   919.782.8323 
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[asterisk-users] Discover held channel?

[Brian Henning]

Hi,

Here is a scenario:

1) A call comes in on an outside line on a DAHDI device
2) The call is answered by a SIP extension (Linksys SPA942 to be exact)
3) The SIP extension places the outside call on hold 
4) The same SIP extension dials another extension.

Is it possible for the dialplan in step 4 to discover the channel of the
call placed on hold in step 3?  In other words, since it is the same SIP
extension that is dialing, is there a way to ask what other channels are in
use by this SIP extension?...  And if not through the regular
extensions.conf dialplan, would this information be available to a custom
application (AGI, etc)?

I am trying to create one-button access (via speed dial, e.g.) to more
complex Asterisk functions such as call park for our less technically savvy
reception employees.

Many thanks,
~Brian

-- 
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  ///\\\  Raleigh, NC 27617 
  USA 
|| 
||phone: 919.782.8320 
  fax:   919.782.8323 
  email: bhenn...@pineinst.com 
-- 



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Re: [asterisk-users] TDM410P dahdi driver == no lights?

[Brian Henning]

Hi Shaun,

Thanks so much for your response!

Unfortunately this is a production server now, so I'm a little wary of
testing out non-release builds (if it were not production, I would
definitely test).

Seeing consistent call operation plus your info below quells my concerns.
I'll just be happy without lights.

Cheers,
~Brian

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, March 08, 2011 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TDM410P  dahdi driver == no lights?

On 03/07/2011 08:22 PM, Brian Henning wrote:
 
 I have just installed an Asterisk server with a Digium TDM410P card with 3
 FXO modules (no module in the 4th slot).
 
 It's lived on two different machines (a test machine, which had Linux
kernel
 2.6.28, and a new dedicated machine which has Linux kernel 2.6.32).
 
 On the test machine (2.6.28), I used the Zaptel drivers.  Once the kernel
 modules were loaded, the lights on the TDM410P came on green for the
 installed FXO modules.
 
 On the new server, the Zaptel drivers wouldn't build so I switched over to
 dahdi.  Everything seems to be working, EXCEPT there are no lights on the
 TDM410P!  I guess I can ignore that the lights aren't lit up, because it
 seems to be functioning as expected (I can dial out and receive incoming
 calls)...but it's disconcerting that the lights aren't on.  Yes, the Molex
 power connector is connected (although I think that's only needed by FXS
 modules).
 
 I've tried google searches but haven't found anything mentioning this odd
 behavior.  Is this expected?
 

Brian,

I haven't run it, but looking through the code this appears to be a
regression I added in 2.4.1 / current trunk (introduced in r9720 [1]).
I've opened issue 18939 and attached some patches if you want to try
them.  That way I can add your reported by / tested-by information if
you would like.

[1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9720
[2] https://issues.asterisk.org/view.php?id=18939

Thanks,
-- 
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] TDM410P dahdi driver == no lights?

[Brian Henning]

Hello,

I have just installed an Asterisk server with a Digium TDM410P card with 3
FXO modules (no module in the 4th slot).

It's lived on two different machines (a test machine, which had Linux kernel
2.6.28, and a new dedicated machine which has Linux kernel 2.6.32).

On the test machine (2.6.28), I used the Zaptel drivers.  Once the kernel
modules were loaded, the lights on the TDM410P came on green for the
installed FXO modules.

On the new server, the Zaptel drivers wouldn't build so I switched over to
dahdi.  Everything seems to be working, EXCEPT there are no lights on the
TDM410P!  I guess I can ignore that the lights aren't lit up, because it
seems to be functioning as expected (I can dial out and receive incoming
calls)...but it's disconcerting that the lights aren't on.  Yes, the Molex
power connector is connected (although I think that's only needed by FXS
modules).

I've tried google searches but haven't found anything mentioning this odd
behavior.  Is this expected?

Many thanks,
~Brian Henning

-- 
  Brian Henning, Software Engineer

/\Pine Research Instrumentation 
   //\\   5908 Triangle Drive 
  ///\\\  Raleigh, NC 27617 
  USA 
|| 
||phone: 919.782.8320 
  fax:   919.782.8323 
  email: bhenn...@pineinst.com 
-- 



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[asterisk-users] Losing registration - ast 1.4.39 and innomedia 6328-2Re

[Brian C. Huffman]

All,

I'm having a problem with an Innomedia 6328-2Re (old Sunrocket Gizmo).  
It keeps losing registration after a period of time ranging from a few 
minutes to a few hours.  It seems that right before it loses 
registration, it fails to send a second register (after the 401 
unauthorized).  Here's a transcript from wireshark (at the end).  The 
last message is all that's received and asterisk now shows UNKNOWN ast 
the status:


1067617:13:05.255123innomediaasteriskSIPRequest: 
REGISTER sip:asterisk
1067717:13:05.255336asteriskinnomediaSIPStatus: 100 
Trying(0 bindings)
1067817:13:05.255388asteriskinnomediaSIPStatus: 401 
Unauthorized(0 bindings)
1067917:13:17.263367asteriskinnomediaSIPRequest: 
OPTIONS sip:1015@innomedia

1068017:13:17.410876innomediaasteriskSIPStatus: 200 OK

The innomedia is behind a Netgear WGR614v7 router with Comcast as the 
ISP.  But I've already put the innomedia into the DMZ of the router and 
it doesn't appear that it's a NAT issue b/c I can actually telnet into 
the innomedia through the router (DMZ) even after it's lost registration.


It *looks* like the problem is the innomedia since it didn't send 
another register.  But I figured I'd ask to see if anyone here knew what 
the problem could be.  Otherwise my next step is to buy a linksys PAP2T.


Thanks,
Brian

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[asterisk-users] Polycom 500 / MWI

[Brian C. Huffman]

All,

I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 
2.1.3.   I can not seem to get the Message Waiting Indicator to work 
reliably (and in my opinion correctly) with voicemail.


I've got the following in my phone.cfg:
reginfo
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.callBack=*97 msg.mwi.1.callBackMode=contact 
msg.mwi.1.subscribe= /mwi

/reginfo

and the indicator will come on if there is a new message but it won't go 
off when I delete the message.  I think that after a period of hours it 
may go off.  But the only way to make it go off quickly is to put some 
invalid chars into the subscribe string and reboot the phone and then 
switch it back and reboot again.


Does anyone know how to setup this phone to work with asterisk so that 
the indicator light comes on when there's a new message and goes off 
quickly (less than a minute) after the message is deleted?


Thanks,
Brian

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[asterisk-users] Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior?

[Brian Capouch]

The subject says it all.  I'm betting there's a way to do it, but so far 
I haven't found the dialplan runestone via web searching.

Thanks.

b.

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[asterisk-users] Exceptionally long queue length queuing . . . .

[Brian Capouch]

I wonder if anyone out there has a perspective on this.  There are a 
welter of tickets out there on the matter, most of them closed.

This problem began for me over a year ago, and continues up to the 
latest versions I've installed (1.6.2.13).

It happens randomly, and the suggestion on one of the bug tracker 
tickets that it is instigated by a small network leg looks to be on 
point to me, because while it happens way often, it doesn't always happen.

My ITSPs have all dropped IAX, and if they're experiencing this problem 
I can see why.  Once the first of these messages has occurred, it's 
goodbye audio for the rest of the call.

If anyone has a perspective on this longstanding problem, I'd sure be 
glad to hear it.

Thanks.

b.

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Re: [asterisk-users] What do you use for Invoicing?

[Brian C. Huffman]

 I'm still working on it, but I am using a2billing and making 
modifications to some of the PHP code.  I modified the layouts of their 
default invoices and and added PDF creation using dompdf 
(http://code.google.com/p/dompdf/downloads/list).


-b

On 08/03/2010 09:41 AM, Zeeshan Zakaria wrote:


I wanted the same, and so wrote my own. There is none free for this 
purpose.


Zeeshan A Zakaria

--
www.ilovetovoip.com http://www.ilovetovoip.com

On 2010-08-03 9:32 AM, bruce bruce bruceb...@gmail.com 
mailto:bruceb...@gmail.com wrote:


Yep, I seen that. That is probably the closet thing but looking at he 
interface it makes me not try to install it. Maybe too complicated. I 
wouldn't want to send customer the whole CDRs but rather a nice Bill 
like the telco sends out.


I am currently toying with NCH Invoicing. Those guys make a software 
for anything and everything.


Thanks,
Bruce

On Tue, Aug 3, 2010 at 9:17 AM, Zeeshan Zakaria zisha...@gmail.com 
mailto:zisha...@gmail.com wrote:



 I know someone who uses a billing solution called 'freeside',
and is happy with it. Personally I...

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[asterisk-users] Migrating from key system to asterisk

[Brian Kolaci]

I currently have 4 lines coming into the house.  We currently have an Avaya 
standard analog key system which has served us well, but running extensions is 
a major pain and requires a dedicated run per extension.  I have ethernet run 
throughout the house though.

The first two lines are home lines, however line1 I use for my consulting 
during the day, and line2 is used by my wife for her work during the day.  
Line3 is a separate business that nobody else in the household should be 
picking up, however I would like to be able to pick it up from any extension if 
required.  Line4 is a fax, but is used for outgoing calls (like conference 
calls, etc.) so that the inbound lines can be left free.

With the current system, I can choose which outbound trunk is used by just 
selecting the line button I prefer.  My wife just picks up the phone for an 
outgoing call and it will choose line2, and if busy then line1.  She would 
never use line3 or line4.  The kids should never touch lines 3 and 4 either.  
They all know enough to only answer lines 1 and 2 and ignore lines 3 and 4, and 
can tell since each line has a different ring as well as the line lights that 
blink.  We also regularly need to be able to barge in on each other's calls 
converting a regular call into a conference.

I know what I'm looking for is SLA (Shared Line Appearance), and I've been 
trying to get that working for a couple of weeks.  It supposedly works 
somewhat, but I cannot find enough coherent documentation as to exactly how to 
set up each of the extensions and the physical Polycom phones.  I do see 
there's still bugs being worked on (even today) thats being tested:  
https://issues.asterisk.org/view.php?id=11688

So there's two questions here.  First, is there a way to accomplish the 
requirements I have above to a generic asterisk installation, or are we talking 
a very large and complicated configuration?  Second, is there anywhere I can 
find some detailed documentation on configuring SLA on Polycom phones?  (For 
example, for each line on the Polycom phone, does it log in with an extension 
number or specifically the string station1_line1, and does the sip.conf get a 
numbered extension for each line on the phone, or do you specifically put 
[statsion1_line1] in there?)

Thanks in advance...


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Re: [asterisk-users] Voicemail : mail attachment to multiple mail-addresses

[Brian C. Huffman]

Create a local mail alias that sends to what you need and then use the
alias in the vmail config.

-b

On Tue, 2010-06-01 at 20:47 +0200, Jonas Kellens wrote:
 I am no programmer, and very happy with what Asterisk holds in it.
 Just hoped that mailing multiple mail-addresses was an easy
 configuration...
 
 
 On 06/01/2010 06:06 PM, Steve Howes wrote: 
  On 1 Jun 2010, at 16:53, Jonas Kellens wrote:

   Sounds... p
   
  Perhaps you could contribute a patch? ;)
  
  S
  



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[asterisk-users] DAHDI not detecting hangup

[Brian C. Huffman]

I've got an old analog PBX and I'm trying to connect an FXO port on my
asterisk server to one of the extensions on the old PBX.  This should
work as en extension on the old PBX should be providing dialtone,
battery current and ring voltage.

However, when the old PBX hangs up asterisk doesn't appear to be
detecting the hangup (the DAHDI channel stays in use).

Can anyone help?

Thanks,
Brian




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Re: [asterisk-users] DAHDI not detecting hangup

[Brian C. Huffman]

Can I apply that to only one channel?

I'm on trixbox and I'm not sure which file should get these changes:
/etc/dahdi/system.conf
/etc/asterisk/chan_dahdi.conf 
/etc/asterisk/dahdi-channels.conf

Thanks,
Brian

On Mon, 2010-05-10 at 09:40 -0500, Danny Nicholas wrote:
 Hanguponpolarityswitch=yes in dahdi.conf?
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian C.
 Huffman
 Sent: Monday, May 10, 2010 9:35 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] DAHDI not detecting hangup
 
 I've got an old analog PBX and I'm trying to connect an FXO port on my
 asterisk server to one of the extensions on the old PBX.  This should
 work as en extension on the old PBX should be providing dialtone,
 battery current and ring voltage.
 
 However, when the old PBX hangs up asterisk doesn't appear to be
 detecting the hangup (the DAHDI channel stays in use).
 
 Can anyone help?
 
 Thanks,
 Brian



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[asterisk-users] Pass MWI from analog line through DAHDI?

[Brian C. Huffman]

Is it possible to pass the Message Waiting indicator from an analog line
to a DAHDI port and into asterisk (and to an extension)?

Thanks,
Brian


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