Re: [asterisk-users] externip nat audio sip trunk issue problem

[C F]

On Wed, Feb 1, 2012 at 9:14 PM, Gabriel Ortiz Lour
 wrote:
> Hi all,
>
>   I've tried search this problem on the list... no luck...
>
>   The case is:
>
> without externip/localnet config on sip.conf [general] my SIP trunk works,
> but with no audio NAT problem (asterisk sends the private 192 address to the
> outside...)
>
> when I configure externip/localnet correctly my SIP trunk simply disappear!
> Checking the signalling with tcpdump shows me that Im sending the packets to
> the correct SIP trunk IP but there is no response AT ALL from it...

Can you explain this?
What do you mean no response? Is it registering? Do you have a debug output?

>
> Anyone had this problem?
>
> Thanks,
> Gabriel
>
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Re: [asterisk-users] Is this doable?

[C F]

Whats asterick?

On Wed, Feb 1, 2012 at 7:48 PM, Josh  wrote:
> I am trying to configure Asterick, having the following system setup on
> the Asterick server:
>
> * eth0 faces the external Internet interface, *but* it does not have IP
> address (it has a private one given to it by my ISP's DHCP server);
> * eth1 faces my internal network (say 10.1.1.0/24);
> * tun0 serves all mobile smartphones and connects to the internal
> network (it has a different ip range, say 10.1.2.0/24) - they are all
> connected via the Internet using OpenVPN;
>
> I would like to configure Asterick for internal calls between ourselves
> (eth1<->tun0) and I think I have no problem with configuring this part.
> I would also like to use one external VOIP provider to which Asterick
> registers on startup. I think I know how to do that and use the
> "register" option in sip.conf, though I am not sure for the rest of the
> NAT-related entries (see below).
>
> The purpose of registering this external account is so that both the
> smart phones (tun0) and the internal net (eth1) users could use this
> account to make external calls (starting with "0", i.e "_0[0-9]."
> pattern in extensioins.conf). Obviously, I need these calls to be routed
> properly via the external VOIP account. In addition to that, I would
> also need to receive calls from that external account to a nominated
> internal one (say on extension 20).
>
> Is this achievable?
>
> If so, I am not completely clear on whether I need to explicitly specify
> my public IP address (via externip/externhost) or whether Asterick is
> able to find it without this option? If not, then my plan is to use
> external program to find it and then use a script in Asterick to set it
> up as an environment variable. Would that work? That external IP address
> is going to change, but only in rare circumstances and in such cases I
> have to restart a lot of stuff (including Asterick) on that server (this
> is usually triggered by a monitoring program), so it won't be a problem
> once it is setup initially. I am also not sure whether to specify
> "nat=yes" or just have "nat=route" only - any ideas?
>
> Is there a comprehensive list of all the options available in sip.conf
> and what they do, because I was unable to find such a list?
>
> If the above is doable, I would also like to add the following 2 features:
>
> 1. Secondary external VOIP account, though I have no idea how to specify
> its port in "register" (it uses port 5065 instead of the standard 5060).
> That account would need to be used on a separate interface (eth2) with a
> different public IP address. Would it be possible to use
> externip/externhost inside that external account section to specify it?
> If this is not possible, then I am thinking of running a separate
> instance of Asterick with the second VOIP account/public IP address set
> up - would that work?
>
> 2. I would like to be able to configure the following work flow: for a
> specific set of (external) calling numbers (including where no Caller ID
> is available):
> a) these callers to be prompted to specify the "reason" for their call;
> b) their response to be temporarily "recorded"/stored (a short message
> of, say no more than 10 seconds long or when they press '#' for that
> recording to stop);
> c) Asterick then rings the nominated number for external VOIP calls
> (extension 20) and play that recorded message back;
> d) then asks for one of four possible outcomes:
> - accept this call (pressing, say 1) in which case the call is connected
> as normal;
> - reject it with a message that that number/person is "unavailable"
> (say, by pressing 0);
> - ask the caller to leave a message by transferring them to a voicemail
> (say by pressing 2); or
> - end the initial call completely with a message that the caller/number
> has been "blacklisted" (say, by pressing the 9 key);
>
> Could this be achieved?
>
> One final question about binding: in order to be able to use both tun0
> and eth1 interfaces so that Asterick serves the calls from both eth1 and
> tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like
> specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the
> tun0 interface - is this possible?
>
> Many thanks in advance!
>
>
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[asterisk-users] externip nat audio sip trunk issue problem

[Gabriel Ortiz Lour]

Hi all,

  I've tried search this problem on the list... no luck...

  The case is:

without externip/localnet config on sip.conf [general] my SIP trunk works,
but with no audio NAT problem (asterisk sends the private 192 address to
the outside...)

when I configure externip/localnet correctly my SIP trunk simply disappear!
Checking the signalling with tcpdump shows me that Im sending the packets
to the correct SIP trunk IP but there is no response AT ALL from it...

Anyone had this problem?

Thanks,
Gabriel
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[asterisk-users] FXS hangup issues

[Ari Pollak]

Greetings,

I currently have an Asterisk 1.8.8.1 system set up with SIP accounts
as well as a Wildcard TDM400P REV I card with both FXS and FXO
ports - FXO is connected to outside lines, FXS connected to inside
analog phones. Everything about the setup works fine except one thing -
after making calls to or from any of the analog phones, and the other
side hangs up, the analog phone just gives a busy signal instead of
hanging up. On the Asterisk console, it seems to think it's hung up
the phone too:

== Spawn extension (from-office, 44, 50005) exited non-zero on 'DAHDI/5-1'
-- Hanging up on 'DAHDI/5-1'
-- Hungup 'DAHDI/5-1'

chan_dahdi.conf is mostly just the default with just the lines
defined, nothing too fancy, and this doesn't happen for SIP clients or remote
phones via the FXO ports.

Any ideas?

Thanks!
Ari

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[asterisk-users] Is this doable?

[Josh]

I am trying to configure Asterick, having the following system setup on
the Asterick server:

* eth0 faces the external Internet interface, *but* it does not have IP
address (it has a private one given to it by my ISP's DHCP server);
* eth1 faces my internal network (say 10.1.1.0/24);
* tun0 serves all mobile smartphones and connects to the internal
network (it has a different ip range, say 10.1.2.0/24) - they are all
connected via the Internet using OpenVPN;

I would like to configure Asterick for internal calls between ourselves
(eth1<->tun0) and I think I have no problem with configuring this part.
I would also like to use one external VOIP provider to which Asterick
registers on startup. I think I know how to do that and use the
"register" option in sip.conf, though I am not sure for the rest of the
NAT-related entries (see below).

The purpose of registering this external account is so that both the
smart phones (tun0) and the internal net (eth1) users could use this
account to make external calls (starting with "0", i.e "_0[0-9]."
pattern in extensioins.conf). Obviously, I need these calls to be routed
properly via the external VOIP account. In addition to that, I would
also need to receive calls from that external account to a nominated
internal one (say on extension 20).

Is this achievable?

If so, I am not completely clear on whether I need to explicitly specify
my public IP address (via externip/externhost) or whether Asterick is
able to find it without this option? If not, then my plan is to use
external program to find it and then use a script in Asterick to set it
up as an environment variable. Would that work? That external IP address
is going to change, but only in rare circumstances and in such cases I
have to restart a lot of stuff (including Asterick) on that server (this
is usually triggered by a monitoring program), so it won't be a problem
once it is setup initially. I am also not sure whether to specify
"nat=yes" or just have "nat=route" only - any ideas?

Is there a comprehensive list of all the options available in sip.conf
and what they do, because I was unable to find such a list?

If the above is doable, I would also like to add the following 2 features:

1. Secondary external VOIP account, though I have no idea how to specify
its port in "register" (it uses port 5065 instead of the standard 5060).
That account would need to be used on a separate interface (eth2) with a
different public IP address. Would it be possible to use
externip/externhost inside that external account section to specify it?
If this is not possible, then I am thinking of running a separate
instance of Asterick with the second VOIP account/public IP address set
up - would that work?

2. I would like to be able to configure the following work flow: for a
specific set of (external) calling numbers (including where no Caller ID
is available):
a) these callers to be prompted to specify the "reason" for their call;
b) their response to be temporarily "recorded"/stored (a short message
of, say no more than 10 seconds long or when they press '#' for that
recording to stop);
c) Asterick then rings the nominated number for external VOIP calls
(extension 20) and play that recorded message back;
d) then asks for one of four possible outcomes:
- accept this call (pressing, say 1) in which case the call is connected
as normal;
- reject it with a message that that number/person is "unavailable"
(say, by pressing 0);
- ask the caller to leave a message by transferring them to a voicemail
(say by pressing 2); or
- end the initial call completely with a message that the caller/number
has been "blacklisted" (say, by pressing the 9 key);

Could this be achieved?

One final question about binding: in order to be able to use both tun0
and eth1 interfaces so that Asterick serves the calls from both eth1 and
tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like
specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the
tun0 interface - is this possible?

Many thanks in advance!


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Re: [asterisk-users] Router that support Asterisk

[Gerardo Barajas]

On Wed, Feb 1, 2012 at 5:41 PM, C F  wrote:

> G
> Have you ever heard of Google?
> Here is a link on google:
> http://lmgtfy.com/?q=google
>
> JAJAJAJAJA!!
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Re: [asterisk-users] Router that support Asterisk

[James Sharp]

On 02/01/2012 02:17 PM, bilal ghayyad wrote:

Hi All;

I heard from some friends that there are a dsl router that has Linux OS
and it has asterisk on it, so the ip phone can register on this router,
also if the router has FXS or FXO ports then it can be used to place
calls through them.

Is it really? Where I can these routers? Did anyone try it to tell us if
it is stable and working fine?

Regards
Bilal


The Cisco DDR2200 that I just got from Centurylink for DSL appears to be 
just that.  I haven't tested the FXS ports on it yet, though.


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Re: [asterisk-users] Router that support Asterisk

[C F]

G
Have you ever heard of Google?
Here is a link on google:
http://lmgtfy.com/?q=google


On Wed, Feb 1, 2012 at 2:17 PM, bilal ghayyad  wrote:

> Hi All;
>
> I heard from some friends that there are a dsl router that has Linux OS
> and it has asterisk on it, so the ip phone can register on this router,
> also if the router has FXS or FXO ports then it can be used to place calls
> through them.
>
> Is it really? Where I can these routers? Did anyone try it to tell us if
> it is stable and working fine?
>
> Regards
> Bilal
>
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>
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Re: [asterisk-users] Problem with DTMF in Voicemail main

[Ira]

At 02:31 PM 2/1/2012, you wrote:
app_voicemail (on some systems) requires that res_adsi and res_smdi 
be built and loaded; if they are not enabled, then the 'checkbox' 
for app_voicemail changes to angle brackets.In menuselect, the 
presence of angle brackets instead of square brackets means that 
enabling the item in question is going to also automatically enable 
modules/features that it depends on that are themselves currently 
disabled (sorry for the long sentence... it's just how menuselect works).


Ah, that's what it means. thanks for the explanation.

Ira 



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Re: [asterisk-users] Problem with DTMF in Voicemail main

[Kevin P. Fleming]

On 02/01/2012 12:46 PM, Ira wrote:


I notice that comedian mail has <> instead of [] brackets. Does that
mean it's on its way to being deprecated?


I assume you are referring to how app_voicemail (not 'comedian mail')
is listed the menuselect tool. Umm... no, those are completely
unrelated. How did you reach that assumption?


It was not an assumption, it was a question. It's the first time I
remember app_voicemail not being checked by default and the first time I
remember seeing it with <> brackets.


app_voicemail (on some systems) requires that res_adsi and res_smdi be 
built and loaded; if they are not enabled, then the 'checkbox' for 
app_voicemail changes to angle brackets.In menuselect, the presence of 
angle brackets instead of square brackets means that enabling the item 
in question is going to also automatically enable modules/features that 
it depends on that are themselves currently disabled (sorry for the long 
sentence... it's just how menuselect works).


--
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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] Asterisk-users caller ID

[motty.cruz]

Hello, 
I have a server that connects to my Voice Server provider so far is working
great! I have a second server that I want to set caller id to a different
number second server I'm going to call it server B. And server B will go
through server A which is connected to my Voice Server Provider. Thus far
I'm unsussessful! Can some one help? 

A$ ee extensions.conf
[outbound]
exten => _91NXXNXX,1,Set(CALLERID(num)=8006332211)
exten => _91NXXNXX,2,Dial(SIP/VSP/${EXTEN:1},80)
exten => _9NXX,1,Set(CALLERID(num)=8006332211)
exten => _9NXX,2,Dial(SIP/VSP/${EXTEN:1},80)


B$ ee extensions.conf
[outbound]
exten => _91NXXNXX,1,Set(CALLERID(num)=8007342323)
exten => _91NXXNXX,2,Dial(SIP/ServerA/${EXTEN}@serverBout)
exten => _9NXX,1,Set(CALLERID(num)=8007342323)
exten => _9NXX,2,Dial(SIP/ServerA/${EXTEN:1}@serverBout)


Every time I call my cell phone from server B I get the caller id from
server A, please help! 

Thanks, 
Motty


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Re: [asterisk-users] Problem with DTMF in Voicemail main

[Ira]

At 06:05 AM 1/31/2012, you wrote:

On 01/31/2012 12:17 AM, Ira wrote:

Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk.

On 10.1.0 and trunk, I can't successfully enter the password for any
mailbox in voicemailmain on my Aastra 480i phones. All four version work
with a Snom cordless SIP phone. In 10.0.0 and 10.0.1 the Aastra works
perfectly. So needless to say I'm back to running 10.0.1. The WAF is
very low for stuff like that.


It is quite unlikely that there were any changes between 10.0.1 and 
10.1.0 that would affect DTMF detection or app_voicemail itself, but 
it's certainly possible. That's why we have an issue reporting 
system, and it's also why we produce release candidates to get 
testing prior to making official releases.


Well, none the less, it's broken now and it wasn't before. I always 
try to run the release candidates but must have missed the one for 
10.1. I added a report to JIRA.



I notice that comedian mail has <> instead of [] brackets. Does 
that mean it's on its way to being deprecated?


I assume you are referring to how app_voicemail (not 'comedian 
mail') is listed the menuselect tool. Umm... no, those are 
completely unrelated. How did you reach that assumption?


It was not an assumption, it was a question. It's the first time I 
remember app_voicemail not being checked by default and the first 
time I remember seeing it with <> brackets. 



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Re: [asterisk-users] Getting one way audio even NAT is configured

[Warren Selby]

On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir  wrote:

> Hi all,
>
> I'm getting one way audio when calling over the SIP trunk i.e. end device
> B (remote end of SIP trunk) can hear device A (softphone registered with
> Asterisk) but device A can't hear device B. Even though I configured same
> NAT configurations on other servers and they are working good. The NAT
> configuration is listed below;
>
> localnet=130.0.0.0/130.0.0.0
> externhost=12.131.12.13
> externrefresh=10
> fromdomain=test.localhost.com
> nat=yes
> qualify=yes
> canreinvite=no
>
>
> NAT on device end i.e. my softphone (extension) has already set to yes
> with canreinvite=no  but still unable to resolve this issue. SIP traces are
> listed below;
>
>



>
> The Asterisk version I'm using is 1.8.5. Please assist me at earliest.
>

Which device (A or B) is behind NAT with regards to your asterisk server?
Is that the actual localnet= statement you're using, because to my
understanding that is not the proper format to use (should be
localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
y.y.y.y is your subnet for your local network).

-- 
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http://www.SelbyTech.com 
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[asterisk-users] Router that support Asterisk

[bilal ghayyad]

Hi All;

I heard from some friends that there are a dsl router that has Linux OS and it 
has asterisk on it, so the ip phone can register on this router, also if the 
router has FXS or FXO ports then it can be used to place calls through them.

Is it really? Where I can these routers? Did anyone try it to tell us if it is 
stable and working fine?

Regards
Bilal --
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[asterisk-users] Getting one way audio even NAT is configured

[Ahmed Munir]

Hi all,

I'm getting one way audio when calling over the SIP trunk i.e. end device B
(remote end of SIP trunk) can hear device A (softphone registered with
Asterisk) but device A can't hear device B. Even though I configured same
NAT configurations on other servers and they are working good. The NAT
configuration is listed below;

localnet=130.0.0.0/130.0.0.0
externhost=12.131.12.13
externrefresh=10
fromdomain=test.localhost.com
nat=yes
qualify=yes
canreinvite=no


NAT on device end i.e. my softphone (extension) has already set to yes with
canreinvite=no  but still unable to resolve this issue. SIP traces are
listed below;

Reliably Transmitting (NAT) to 12.194.12.12:5060:
INVITE sip:173242@12.194.12.12 SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK1fbbab95;rport
Max-Forwards: 70
From: "77057" ;tag=as1fa9b502
To: 
Contact: 
Call-ID: 04ce1d566f1f17a221caba261e2af...@test.localhost.com
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.5.0)
Date: Wed, 01 Feb 2012 16:11:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 122642112 122642112 IN IP4 12.131.12.13
s=Asterisk PBX 1.8.5.0
c=IN IP4 12.131.12.13
t=0 0
m=audio 16006 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called SIP/ATTLABS-IP-FlexReach/173242

<--- SIP read from UDP:12.194.12.12:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060
From: "77057" ;tag=as1fa9b502
To: 
Call-ID: 04ce1d566f1f17a221caba261e2af...@test.localhost.com
CSeq: 102 INVITE

<->
--- (6 headers 0 lines) ---

<--- SIP read from UDP:12.194.12.12:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060
From: "77057" ;tag=as1fa9b502
To: ;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Call-ID: 04ce1d566f1f17a221caba261e2af...@test.localhost.com
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
Contact: 
Content-Length: 237
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 14862 4757 IN IP4 12.194.12.12
s=SIP Media Capabilities
c=IN IP4 12.194.12.12
t=0 0
m=audio 16534 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<->
--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 12.194.12.12:16534
-- SIP/ATTLABS-IP-FlexReach-0025 is making progress passing it to
SIP/2005-0024

<--- SIP read from UDP:12.194.12.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060
From: "77057" ;tag=as1fa9b502
To: ;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Call-ID: 04ce1d566f1f17a221caba261e2af...@test.localhost.com
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed
Contact: 
Content-Length: 237
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 14862 4757 IN IP4 12.194.12.12
s=SIP Media Capabilities
c=IN IP4 12.194.12.12
t=0 0
m=audio 16534 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<->
--- (12 headers 11 lines) ---
list_route: hop: 
set_destination: Parsing  for
address/port to send to
set_destination: set destination to 12.194.12.12:5060
Transmitting (NAT) to 12.194.12.12:5060:
ACK sip:12.194.12.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK483f052d;rport
Max-Forwards: 70
From: "77057" ;tag=as1fa9b502
To: ;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Contact: 
Call-ID: 04ce1d566f1f17a221caba261e2af...@test.localhost.com
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.5.0)
Content-Length: 0


---
-- SIP/ATTLABS-IP-FlexReach-0025 answered SIP/2005-0024
-- Locally bridging SIP/2005-0024 and
SIP/ATTLABS-IP-FlexReach-0025
Reliably Transmitting (NAT) to 12.194.12.12:5060:
OPTIONS sip:12.194.12.12 SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK06532068;rport
Max-Forwards: 70
From: "Unknown" ;tag=as054a7d2d
To: 
Contact: 
Call-ID: 767dcb7d4406d06c248a7056559ad...@test.localhost.com
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.5.0)
Date: Wed, 01 Feb 20

Re: [asterisk-users] Dynamically toggling ConfBridge recording from conference menu

[Kevin P. Fleming]

On 02/01/2012 11:42 AM, Josh Freeman wrote:

Hello,

I'm using ConfBridge in an application where I need a conference admin
to be able to start and stop recording using a conference menu option.

Currently, I'm doing this by defining ConfBridge menu options

7=dialplan_exec(conference_functions,rec_start,1)
9=dialplan_exec(conference_functions,rec_stop,1)

The rec_start and rec_stop extensions simply start/stop MixMonitor on
the channel of the admin who presses 7/9. However, what I'd really like
to do is to be able to execute the equivalent of the CLI "confbridge
record start " command, so that the recording would be independent
of the participant channel.

I suppose I could do this with a System call, something like
System(asterisk -rx "confbridge record start ") - but is there a
better, less-roundabout way of getting there?


Unfortunately at this time there isn't any 'record' control available 
directly in the ConfBridge menu system, although it would surely not be 
hard to add.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
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] Dynamically toggling ConfBridge recording from conference menu

[Josh Freeman]

Hello,

I'm using ConfBridge in an application where I need a conference admin
to be able to start and stop recording using a conference menu option.

Currently, I'm doing this by defining ConfBridge menu options

7=dialplan_exec(conference_functions,rec_start,1)
9=dialplan_exec(conference_functions,rec_stop,1)

The rec_start and rec_stop extensions simply start/stop MixMonitor on
the channel of the admin who presses 7/9. However, what I'd really like
to do is to be able to execute the equivalent of the CLI "confbridge
record start " command, so that the recording would be independent
of the participant channel.

I suppose I could do this with a System call, something like
System(asterisk -rx "confbridge record start ") - but is there a
better, less-roundabout way of getting there?

Thanks,
Josh

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Re: [asterisk-users] SIP Provider Russia, Ukraine, Poland

[Markus]

Can't help you with the SIP account but for geographical numbers in all 
3 countries that you mentioned try http://www.globalnumbers.de - 
forwarding to any SIP destination is free.



Am 01.02.2012 13:29, schrieb Christian Gansberger:

Hello List!

I'm searching for SIP-Providers in the following countries:
Russia
Ukraine
Poland

I need a geographical number for each country, maybe a prepaid
SIP-Account, trunking is not important.
Has anyone some experience with these countries?

yours
christian

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[asterisk-users] SIP Provider Russia, Ukraine, Poland

[Christian Gansberger]

Hello List!

I'm searching for SIP-Providers in the following countries:
Russia
Ukraine
Poland

I need a geographical number for each country, maybe a prepaid
SIP-Account, trunking is not important.
Has anyone some experience with these countries?

yours
christian

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Re: [asterisk-users] CA Issued Certificates / TLS + SRTP

[Daniel Pocock]

On 01/02/12 10:58, Stuart Elvish wrote:
> Thanks for the clarification. I have looked at Polycom's website and
> saw which phones have the latest firmware (or at least a firmware that
> supports TLS) available.
> 
> Didn't get around to the testing with the chained certificate but will
> try again this evening.
> 
> 

One thing that frustrates people about Polycom is the very limited list
of root CAs they support - it was probably OK when they first started
doing SSL, but things have changed a lot now

The latest phones (e.g. IP321) have more memory than those they replace
(e.g. IP320) and so they should be able to handle a larger list of built
in root CAs (which Polycom can distribute through the firmware update).
 The obvious ones that are missing are the budget CAs:

- CaCert.org (all certs are free)
- startssl.com  (which has some free certs)
- GoDaddy

These budget CAs are now supported by the various Linux distributions
and Android phones, so they are clearly above a certain threshold of
stability

Polycom phones should also be able to handle 4096 bit certs with the
extra memory, but that appears to need remediation in the firmware (I
tried installing a custom 4096 bit cert and it didn't accept it)

If anyone is registered with Polycom as a reseller, they can quote these
issue numbers:

EXT-3192 GoDaddy root CA cert
https://jira.polycom.com:8443/browse/EXT-3192

EXT-3193 CACert root CA cert
https://jira.polycom.com:8443/browse/EXT-3193

EXT-3238 Support for 4096 bit keys
https://jira.polycom.com:8443/browse/EXT-3238

As in most commercial enterprises, the more customers who request fixes
on these issues, the higher it will go on their priority list

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[asterisk-users] Asterisk 10.0 Realtime

[Andrew Nowrot]

Hi

I have noticed new behaviour of asterisk 10.0 realtime.
In 1.6 when I was using realtime:

"""
[somecontext]

 exten => someexten1..
 exten => someexten2..
 exten => someexten3..
 exten => someexten4..

switch => Realtime/${CONTEXT}@extensions
"""

switch statement was executed after lines above (so there was a
precedence of the lines declared in a extensions.conf over the ones in
database).

In asterisk 10.0 switch is executed before extens declared in the
extensions.conf file.

Is there a way to change that and have previous behaviour?


Cheers

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Re: [asterisk-users] read digits during recording / DTMF in conference?

[isrlgb]

M…
-Original Message-
From: Kingsley Tart 
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 01 Feb 2012 10:34:07 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: [asterisk-users] read digits during recording / DTMF in conference?

Hi,

I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they'd be diverted to
the extension number they entered.

I realise this can be done with the voicemail app with operator=yes but
the problem with this is that the caller has to press 0 while the
announcement is being played. If they're too slow and recording has
started, they've missed the opportunity.

So I played around with ConfBridge and a couple of call files, just to
see if I could get it to work. It's a bit convoluted but the idea is
that the caller gets silently put into a conference, then two call files
make asterisk silently connect to other calls into the same conference,
with one doing the recording and the other using Read() to collect
digits.

If I just had the caller and one of the other calls in the conference
(the one doing Read()) then this worked - Read() managed to read the
DTMF digits and assign them to a variable.

However, when the 'recording' call is also in the conference, the 'read'
call can no longer recognise the DTMF digits. To test, I made the 'read'
call play a sound before calling Read() and I could hear this being
played so the call was definitely there. However, regardless of the
number of digits I pressed, Read() didn't notice any of them, even if I
introduced a delay so that the other channels were quiet before the call
to Read().

I realise this might seem a bit like a mad solution but can anyone else
think of a way to get Asterisk to read (and react to) DTMF digits during
a recording?

This is with Asterisk 1.8.7.

-- 
Cheers,
Kingsley.


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[asterisk-users] read digits during recording / DTMF in conference?

[Kingsley Tart]

Hi,

I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they'd be diverted to
the extension number they entered.

I realise this can be done with the voicemail app with operator=yes but
the problem with this is that the caller has to press 0 while the
announcement is being played. If they're too slow and recording has
started, they've missed the opportunity.

So I played around with ConfBridge and a couple of call files, just to
see if I could get it to work. It's a bit convoluted but the idea is
that the caller gets silently put into a conference, then two call files
make asterisk silently connect to other calls into the same conference,
with one doing the recording and the other using Read() to collect
digits.

If I just had the caller and one of the other calls in the conference
(the one doing Read()) then this worked - Read() managed to read the
DTMF digits and assign them to a variable.

However, when the 'recording' call is also in the conference, the 'read'
call can no longer recognise the DTMF digits. To test, I made the 'read'
call play a sound before calling Read() and I could hear this being
played so the call was definitely there. However, regardless of the
number of digits I pressed, Read() didn't notice any of them, even if I
introduced a delay so that the other channels were quiet before the call
to Read().

I realise this might seem a bit like a mad solution but can anyone else
think of a way to get Asterisk to read (and react to) DTMF digits during
a recording?

This is with Asterisk 1.8.7.

-- 
Cheers,
Kingsley.


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Re: [asterisk-users] CA Issued Certificates / TLS + SRTP

[Stuart Elvish]

Thanks for the clarification. I have looked at Polycom's website and
saw which phones have the latest firmware (or at least a firmware that
supports TLS) available.

Didn't get around to the testing with the chained certificate but will
try again this evening.



>
>> * And, is it necessary to use both my server specific certificate and
>> the intermediate certificate on the telephones or will the telephones
>> only require the server specific certificate?
> The phones should already have the root certificate for Geotrust, you
> should not deploy intermediate roots into the phones if you can
> avoid it
 If I understand this correctly (and the other emails you sent), the
 Polycom does not need any preloaded certificates / keys, it will ask the
 CA and then evaluate the certificate provided by Asterisk during TLS
 setup; is that correct? Makes it much easier. (Unfortunately my Polycom
 is a bit old so I will have to see if I can upgrade it.)
>
>
>
> By `preloaded', I mean you should not have to load any certificates or
> key pairs manually into the phones
>
> The phones should have the default CA certs that come in the firmware
>
> Most recent Polycom phones also have a client certificate and private
> key built in.  This allows you to secure the provisioning process.
>
> Some of the older Polycoms went end-of-life, some don't have client
> certs built in, so you'll have to research all that carefully on their
> support site.  E.g. the IP 300, IP 430 and IP 500 are too old for proper
> TLS, while the IP321, IP 450 and IP550 are good
>
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Re: [asterisk-users] CA Issued Certificates / TLS + SRTP

[Daniel Pocock]

> * And, is it necessary to use both my server specific certificate and
> the intermediate certificate on the telephones or will the telephones
> only require the server specific certificate?
 The phones should already have the root certificate for Geotrust, you
 should not deploy intermediate roots into the phones if you can
 avoid it
>>> If I understand this correctly (and the other emails you sent), the
>>> Polycom does not need any preloaded certificates / keys, it will ask the
>>> CA and then evaluate the certificate provided by Asterisk during TLS
>>> setup; is that correct? Makes it much easier. (Unfortunately my Polycom
>>> is a bit old so I will have to see if I can upgrade it.)



By `preloaded', I mean you should not have to load any certificates or
key pairs manually into the phones

The phones should have the default CA certs that come in the firmware

Most recent Polycom phones also have a client certificate and private
key built in.  This allows you to secure the provisioning process.

Some of the older Polycoms went end-of-life, some don't have client
certs built in, so you'll have to research all that carefully on their
support site.  E.g. the IP 300, IP 430 and IP 500 are too old for proper
TLS, while the IP321, IP 450 and IP550 are good

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Re: [asterisk-users] Proposed changes to Asterisk release and support cycles

[Hans Witvliet]

On Tue, 2012-01-31 at 15:52 -0500, John Knight wrote:
> > I like the idea of LTR release more often that would have the
> > feature patches baked in.  Case in point the new conference app
> > requires a jump to version 10 while the 1.8 conference app is quite
> > useless but 1.8 is my LTR version so I am stuck without the
> > conference app in my mainline systems for two years. 
> 
> Well said!  This is also true of any type of long term supported
> release whether if it's an operating system, application, etc.  In the
> "LTS" name, it conjurs up thoughts of Ubuntu, but comparisons to
> RHEL/Fedora are far more appropriate I would think as Ubuntu focuses
> nearly exclusively on new point releases while backporting new
> features is what a company like Red Hat excels at and should be the
> prime example of how to run dual software channels (enterprise release
> in RHEL vs. hobby release in Fedora). 
> 


> 
> I know distros and applications are two fundamentally different
> things, with entirely different goals and requirements, but I still
> think Red Hat provides the best example because 1) they have turned it
> into a science how smooth their development process goes in ratio to
> satisfied customers and 2) it's the only other open source software
> project I can think of that can accurately compare.  In a past meeting
> I had with Digium while working for another company, they too directly
> drew a correlation between the new LTS idea and ubuntu lts/non-lts and
> rhel/fedora.
> 
> The conference app changes since 1.4 I haven't been thrilled with, but
> in the whole time I've been supporting 1.8.x for my customers, I've
> come up with a very stable solution building on it and I haven't had
> any surprises come my way.   
> 
very well said indeed.
Some (...) distro's think dat LTS implies a complete feature freeze.
Others are more flexibel about it, that besides current versions of
applications, they are willing to support both elder _and_ newer
versions. (as example, i'm refering to the fact that hours after the
anouncement, firefox10 became available for sles11)

As said, re-written features like conference, are that important that
one shouldn't have to wait years for the next LTS. So this overlap of
multiple LTS-versions looks very much attractive

Having said that, i do understand that multiple versions of
features/applications puts an huge extra burden on the people who have
to maintain both versions, as the original version (as the term LTS
implies) should be maintained with all its limitations also.

hw

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