Re: [asterisk-users] PJSIP signaling question

[Kevin Long]

Thanks George I appreciate the info .  Being able to see what codec is in use 
for call in progress is very handy sometimes. 

As far as the RTP stats goes,  I see there is some info with “rtp” and “rtcp” 
commands which can be useful for troubleshooting. A running tally of # packets 
or bandwidth used would be awesome in along with the codec in "pjsip show 
channels" or something like that.


Im not certain, but I think the TLS signalling problem from this email may be 
happening to me again after patching for another pjsip/NAT issue which was with 
the external_media_address not working and the internal IP being sent in the 
SDP from asterisk - I applied this patch to the codebase and recompiled I am 
seeing the TLS “new transport”  issue again , I think.

Regards,

Kevin Long

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Re: [asterisk-users] RTP / NAT question ( pjsip )

[Kevin Long]

So the patch did resolve the audio RTP issue and I can make echo calls now,   
but it seems like the last issue I posted to the list,  (pjsip driver making 
new outbound TLS transports instead of using existing SIP connection, not NAT 
friendly)   is happening again ..   Could that be?


Thanks again,


Kevin Long

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[asterisk-users] Selecting timing source for Asterisk

[Carlos Chavez]

I have an Asterisk 13 installation with an E1 card and I thought 
that DAHDI would be the default timing source for the system:


pbxcore*CLI> module show like timing
Module Description  Use Count  Status 
Support Level
res_timing_dahdi.soDAHDI Timing Interface   
0  Running  core
res_timing_pthread.so  pthread Timing Interface 
0  Running  extended
res_timing_timerfd.so  Timerfd Timing Interface 
13 Running  core

3 modules loaded

How is the timing source selected?  I thought that if a DAHDI card 
was present it was automatically selected as the timing source for the 
system.


--
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Carlos Chávez
+52 (55)9116-91161


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Re: [asterisk-users] Rasterisk freeze on 4G link

[Duncan]

> With this new link, whenever I launch a vim, a nano or a rasterisk 
session, my terminal freezes (rasterisk) or remains empty (nano, vim).


>
> When a session is frozon, I can open a new one at the same so it
excludes a basic connectivity loss.
>
Usually incorrect MTU gives you this effect. Use ping with MTU
size set to test and find what works.

I think its value: it's 1272, which amazes me.


You will probably also break it with any large text dump eg cat
/var/log/syslog will also do it


Yes  "cat /var/log/syslog" also broke my console.

Why would my console break because of inadequate MTU and other PC on 
the same location, seem unaffected ?

Because, they most probably mostly use SMTP and HTTP ?

Is possible to simulate a given MTU on a LAN to reproduce such freezes ?
(the remote location is at the other side of the country and I would 
like to prepare things as much as possible).



I think you need to go through a router or some device that can 
constrain the MTU. But live changing your server MTU should be straight 
forward as openvpn should try and reconnect, and you can change the 
server back. I haven't lost connectivity before with this


Also the session is probably timed out rather than gone, in 10-15 mins 
maybe less it will come back (or does for me)


Cheers Duncan
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Re: [asterisk-users] Windstream SIP Trunk settings

[James Cass]

Here's what I ultimately got to work (in case it helps someone):

Name your trunk
Enter your outgoing CID

Under Outgoing settings-
Trunk name - whatever you choose to name it
PEER Details-
host=IP address of SIP gateway
type=friend
context=from-trunk
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite

Incoming settings -

none

Registration string -
username:passw...@xxx.xxx.xxx.xxx

James Cass 
jcas...@gmail.com


On Tue, Feb 23, 2016 at 12:20 PM, Rodrigo Ramírez Norambuena <
decipher...@gmail.com> wrote:

> February 23 2016 9:37 AM, "James Cass"  wrote:
> > Thanks everyone, all sound advice. Still can't even get the calls to
> show up on the console at all
> > - I suspect the issue is on the WS side, as I'm not having any issues
> with other carriers with
> > similar settings.
>
> You can debug SIP to detect the problem. May be exists some cause tell you
> more information in the
> trace SIP.
> --
> Rodrigo Ramírez Norambuena
> http://www.rodrigoramirez.com
>
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Re: [asterisk-users] RTP / NAT question ( pjsip )

[Kevin Long]

Hi Joshua,

This Asterisk 13 was pulled from git master branch just 2-3 days ago: 
GIT-13-d1495b . 

I used this very recent source code to overcome a pjsip problem (you can see my 
email list post from a few days ago)


Thanks again




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Re: [asterisk-users] Asterisk Call Forwarding

[Steve Edwards]

On Fri, 4 Mar 2016, Madushan Geethanga wrote:


What is redacted means?

same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void)


Censored. Ususally for political reasons. In this case, the OP didn't want 
to put a real phone number in a public list.


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Re: [asterisk-users] Asterisk Call Forwarding

[Madushan Geethanga]

Hi

What is redacted means?

same => n,GotoIf($["${CALLERID(num)}"="**"]?divert:void)

Thanks
Best Regards,
Madushan

On Thu, Mar 3, 2016 at 10:58 PM, Madushan Geethanga  wrote:

>
> Hi,
>
> Thanks Phil, I will implement this and get back to you.
>
> Best Regards,
> Madushan
>
> On Thu, Mar 3, 2016 at 4:12 PM, Phil Reynolds <
> phil-aster...@tinsleyviaduct.com> wrote:
>
>> On Thu, 3 Mar 2016 08:21:14 +0530
>> Madushan Geethanga  wrote:
>>
>> > Hi
>> > I have to setup call forwarding. How do we setup Call forwarding in
>> > asterisk?. Eg. user dials a number and insert some mobile number for
>> > forwarding and dial another number to cancel the forwarding. thanks a
>> > lot.
>>
>> I implemented this like so in my default context:
>>
>> exten => _*21.,1,Answer()
>> same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void)
>> same => n(divert),Set(DB(divert/${CALLERID(num):-4})=${EXTEN:3})
>> same => n,Gosub(divertactive,1)
>> same => n,Hangup()
>> same => n(void),Gosub(divertvoid,1)
>> exten => _#21,1,Answer()
>> same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void)
>> same => n(divert),Verbose(0,${DB_DELETE(divert/${CALLERID(num):-4})})
>> same => n,Gosub(divertoff,1)
>> same => n,Hangup()
>> same => n(void),Gosub(divertvoid,1)
>>
>> (note: use whatever you need in the GotoIf to validate that the phone
>> the call is from is permitted to set up call forwarding - unless you're
>> allowing it for all that can reach the context)
>>
>> The divert{off,active,void} subroutines are where I handle the
>> announcements - but you could probably easily implement your own.
>>
>> At the top of my [voicemail] context, I do this:
>>
>>
>> exten=>s,1,GotoIf(${DB_EXISTS(divert/${ARG3})}?outbound-standard,${DB_RESULT},1)
>>
>> (ARG3 contains the last four digits of the number the call came to in
>> my case, and a success passes the call via the "outbound-standard"
>> context which is in my dialplan. Your exact requirements may vary but
>> this may help.)
>>
>> --
>> Phil Reynolds
>> mail: phil-aster...@tinsleyviaduct.com
>> Web: http://phil.tinsleyviaduct.com/
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Rasterisk freeze on 4G link

[Pete Mundy]

These are not Asterisk related questions. It is a common problem. Google is 
your friend. Try something like 'console stalls with big packets'.

To answer your question "why", it's simple. "Because the big packets are being 
dropped".

Pete


> On 4/03/2016, at 7:15 am, Olivier  wrote:
> 
> Yes  "cat /var/log/syslog" also broke my console.
> 
> Why would my console break because of inadequate MTU and other PC on the same 
> location, seem unaffected ?
> Because, they most probably mostly use SMTP and HTTP ?
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Re: [asterisk-users] Rasterisk freeze on 4G link

[Olivier]

2016-03-03 18:08 GMT+01:00 Duncan Turnbull :

>
> > On 4/03/2016, at 5:31 AM, Olivier  wrote:
> >
> > Hello,
> >
> > I'm remotely managing an asterisk setup using an OpenVPN client on this
> Asterisk box, connecting to an OpenVPN server of mine).
> >
> > This box is mainly connected to PSTN.
> > It is also connected to the Internet, only for remote management.
> >
> > The former ADSL link has recently been replaced by a new 4G link (UMTS).
> >
> > I'm connecting to this box from a Debian Jessie/Gnome Terminal combo.
> >
> > With this new link, whenever I launch a vim, a nano or a rasterisk
> session, my terminal freezes (rasterisk) or remains empty (nano, vim).
> >
> > When a session is frozon, I can open a new one at the same so it
> excludes a basic connectivity loss.
> >
> Usually incorrect MTU gives you this effect. Use ping with MTU size set to
> test and find what works.
>
I think its value: it's 1272, which amazes me.

>
> You will probably also break it with any large text dump eg cat
> /var/log/syslog will also do it
>

Yes  "cat /var/log/syslog" also broke my console.

Why would my console break because of inadequate MTU and other PC on the
same location, seem unaffected ?
Because, they most probably mostly use SMTP and HTTP ?

Is possible to simulate a given MTU on a LAN to reproduce such freezes ?
(the remote location is at the other side of the country and I would like
to prepare things as much as possible).


> > What would you suggest ?
> >
> > Best regards
> >
> >
> > PS: I was about to determine best MTU value but I always thought a
> punishment for a bad MTU value would be a lower throughput, not a screen
> freeze. Is it correct ?
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>
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Re: [asterisk-users] Rasterisk freeze on 4G link

[Pete Mundy]

Oliver,

Not correct!

Duncan and Toufic are spot-on with their answers.

Pete

> On 4/03/2016, at 5:40 am, Toufic Gmail  wrote:
> 
>> 
>> PS: I was about to determine best MTU value but I always thought a 
>> punishment for a bad MTU value would be a lower throughput, not a screen 
>> freeze. Is it correct ?

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Re: [asterisk-users] Asterisk Call Forwarding

[Madushan Geethanga]

Hi,

Thanks Phil, I will implement this and get back to you.

Best Regards,
Madushan
On Thu, Mar 3, 2016 at 4:12 PM, Phil Reynolds <
phil-aster...@tinsleyviaduct.com> wrote:

> On Thu, 3 Mar 2016 08:21:14 +0530
> Madushan Geethanga  wrote:
>
> > Hi
> > I have to setup call forwarding. How do we setup Call forwarding in
> > asterisk?. Eg. user dials a number and insert some mobile number for
> > forwarding and dial another number to cancel the forwarding. thanks a
> > lot.
>
> I implemented this like so in my default context:
>
> exten => _*21.,1,Answer()
> same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void)
> same => n(divert),Set(DB(divert/${CALLERID(num):-4})=${EXTEN:3})
> same => n,Gosub(divertactive,1)
> same => n,Hangup()
> same => n(void),Gosub(divertvoid,1)
> exten => _#21,1,Answer()
> same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void)
> same => n(divert),Verbose(0,${DB_DELETE(divert/${CALLERID(num):-4})})
> same => n,Gosub(divertoff,1)
> same => n,Hangup()
> same => n(void),Gosub(divertvoid,1)
>
> (note: use whatever you need in the GotoIf to validate that the phone
> the call is from is permitted to set up call forwarding - unless you're
> allowing it for all that can reach the context)
>
> The divert{off,active,void} subroutines are where I handle the
> announcements - but you could probably easily implement your own.
>
> At the top of my [voicemail] context, I do this:
>
>
> exten=>s,1,GotoIf(${DB_EXISTS(divert/${ARG3})}?outbound-standard,${DB_RESULT},1)
>
> (ARG3 contains the last four digits of the number the call came to in
> my case, and a success passes the call via the "outbound-standard"
> context which is in my dialplan. Your exact requirements may vary but
> this may help.)
>
> --
> Phil Reynolds
> mail: phil-aster...@tinsleyviaduct.com
> Web: http://phil.tinsleyviaduct.com/
>
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Re: [asterisk-users] Rasterisk freeze on 4G link

[Duncan Turnbull]

> On 4/03/2016, at 5:31 AM, Olivier  wrote:
> 
> Hello,
> 
> I'm remotely managing an asterisk setup using an OpenVPN client on this 
> Asterisk box, connecting to an OpenVPN server of mine).
> 
> This box is mainly connected to PSTN.
> It is also connected to the Internet, only for remote management.
> 
> The former ADSL link has recently been replaced by a new 4G link (UMTS).
> 
> I'm connecting to this box from a Debian Jessie/Gnome Terminal combo.
> 
> With this new link, whenever I launch a vim, a nano or a rasterisk session, 
> my terminal freezes (rasterisk) or remains empty (nano, vim).
> 
> When a session is frozon, I can open a new one at the same so it excludes a 
> basic connectivity loss.
> 
Usually incorrect MTU gives you this effect. Use ping with MTU size set to test 
and find what works.

You will probably also break it with any large text dump eg cat /var/log/syslog 
will also do it

> What would you suggest ?
> 
> Best regards
> 
> 
> PS: I was about to determine best MTU value but I always thought a punishment 
> for a bad MTU value would be a lower throughput, not a screen freeze. Is it 
> correct ?
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Re: [asterisk-users] Rasterisk freeze on 4G link

[Toufic Gmail]

Please set correct MTU at server side, it is definitely an MTU issue.  

Sent from my iPhone

> On Mar 3, 2016, at 5:31 PM, Olivier  wrote:
> 
> Hello,
> 
> I'm remotely managing an asterisk setup using an OpenVPN client on this 
> Asterisk box, connecting to an OpenVPN server of mine).
> 
> This box is mainly connected to PSTN.
> It is also connected to the Internet, only for remote management.
> 
> The former ADSL link has recently been replaced by a new 4G link (UMTS).
> 
> I'm connecting to this box from a Debian Jessie/Gnome Terminal combo.
> 
> With this new link, whenever I launch a vim, a nano or a rasterisk session, 
> my terminal freezes (rasterisk) or remains empty (nano, vim).
> 
> When a session is frozon, I can open a new one at the same so it excludes a 
> basic connectivity loss.
> 
> What would you suggest ?
> 
> Best regards
> 
> 
> PS: I was about to determine best MTU value but I always thought a punishment 
> for a bad MTU value would be a lower throughput, not a screen freeze. Is it 
> correct ?
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> _
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>   http://www.asterisk.org/hello
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[asterisk-users] Rasterisk freeze on 4G link

[Olivier]

Hello,

I'm remotely managing an asterisk setup using an OpenVPN client on this
Asterisk box, connecting to an OpenVPN server of mine).

This box is mainly connected to PSTN.
It is also connected to the Internet, only for remote management.

The former ADSL link has recently been replaced by a new 4G link (UMTS).

I'm connecting to this box from a Debian Jessie/Gnome Terminal combo.

With this new link, whenever I launch a vim, a nano or a rasterisk session,
my terminal freezes (rasterisk) or remains empty (nano, vim).

When a session is frozon, I can open a new one at the same so it excludes a
basic connectivity loss.

What would you suggest ?

Best regards


PS: I was about to determine best MTU value but I always thought a
punishment for a bad MTU value would be a lower throughput, not a screen
freeze. Is it correct ?
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Re: [asterisk-users] Dial your phone and contact phone from within outlook?

[Ryan, Travis]

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, March 03, 2016 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial your phone and contact phone from within 
outlook?

Is TAPI still available on Windows 10, for instance ?

2016-03-02 23:22 GMT+01:00 Neeraj Chand 
>:
Hi Travis,

Have a look at this:

http://www.ipcom.at/en/telephony/siptapi/

I have used this in the past to do something similar, unless you have an 
Exchange Enterprise setup in which case I would suggest exploring unified 
messaging

Thanks,

Neeraj

On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis 
> wrote:
I am wondering what the best solution is for initiating a call from Outlook 
Contacts. I imagine something that would start the call from the outlook card 
(or similar) and then dial the user’s extension and the contact’s phone number 
and place them in a bridge.

Anyone use something like this?

Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com
(765) 742-1102

FFYI, this worked great for me on Windows 7. Not sure if it works with any 
after as I’ve not tested.
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Re: [asterisk-users] Dial your phone and contact phone from within outlook?

[Olivier]

Is TAPI still available on Windows 10, for instance ?

2016-03-02 23:22 GMT+01:00 Neeraj Chand :

> Hi Travis,
>
> Have a look at this:
>
> http://www.ipcom.at/en/telephony/siptapi/
>
> I have used this in the past to do something similar, unless you have an
> Exchange Enterprise setup in which case I would suggest exploring unified
> messaging
>
> Thanks,
>
> Neeraj
>
> On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis 
> wrote:
>
>> I am wondering what the best solution is for initiating a call from
>> Outlook Contacts. I imagine something that would start the call from the
>> outlook card (or similar) and then dial the user’s extension and the
>> contact’s phone number and place them in a bridge.
>>
>>
>>
>> Anyone use something like this?
>>
>>
>>
>> Travis Ryan
>> Director of Information Technologies
>> Oscar Winski Company
>> 2407 North Ninth Street
>> Lafayette, IN 47905
>> ry...@oscarwinski.com
>> (765) 742-1102
>>
>>
>> *We're not the IT departmentWe're the I-TEAM department!*
>>
>>
>>
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>
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Re: [asterisk-users] Dial your phone and contact phone from within outlook?

[Tech Support]

Hey;

I’ve used Camrivox in the past and it is an excellent product, the best 
I’ve seen. However, it is commercial software, so you’ll have to determine if 
it's within your budget or not. You can check it out at http://www.camrivox.com.

Regards;

John V.

 

Tech Support

Tech Support

VoIP Business Solutions

240-215-3479 x325

supp...@voipbusiness.us  

 

 

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand
Sent: Wednesday, March 02, 2016 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial your phone and contact phone from within 
outlook?

 

Hi Travis, 

 

Have a look at this: 

 

http://www.ipcom.at/en/telephony/siptapi/

 

I have used this in the past to do something similar, unless you have an 
Exchange Enterprise setup in which case I would suggest exploring unified 
messaging 

 

Thanks, 

 

Neeraj

 

On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis  wrote:

I am wondering what the best solution is for initiating a call from Outlook 
Contacts. I imagine something that would start the call from the outlook card 
(or similar) and then dial the user’s extension and the contact’s phone number 
and place them in a bridge. 

 

Anyone use something like this?

 

Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
  ry...@oscarwinski.com
(765) 742-1102


We're not the IT departmentWe're the I-TEAM department!

 


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Re: [asterisk-users] RTP / NAT question ( pjsip )

[Joshua Colp]

Kevin Long wrote:

Hi Joshua,



Looking at the transmitted SIP packets from Asterisk,  it looks like
Asterisk is only sending it’s own internal IP (it is behind a NAT
too, with proper port forwarding) .

I did set in my transport the external_signaling_address and
external_media_address  ,  and I have now put transport= into my
endpoint configuration hoping they will “inherit” the correct public
IP for the media .

But Asterisk is still sending RTP to the wrong IP .


I am trying to test a “real world” scenario of public IP and NAT
traversal,  but I do have split tunnel VPN in my environment so the
endpoint and the asterisk server *could* reach each other by the
private IP ,but I am actually trying to avoid this with a proper
configuration since my real users will not be on any  VPN, mostly.


What version of 13 are you also using?

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


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Re: [asterisk-users] Asterisk Call Forwarding

[Phil Reynolds]

On Thu, 3 Mar 2016 08:21:14 +0530
Madushan Geethanga  wrote:

> Hi
> I have to setup call forwarding. How do we setup Call forwarding in
> asterisk?. Eg. user dials a number and insert some mobile number for
> forwarding and dial another number to cancel the forwarding. thanks a
> lot.

I implemented this like so in my default context:

exten => _*21.,1,Answer()
same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void)
same => n(divert),Set(DB(divert/${CALLERID(num):-4})=${EXTEN:3})
same => n,Gosub(divertactive,1)
same => n,Hangup()
same => n(void),Gosub(divertvoid,1)
exten => _#21,1,Answer()
same => n,GotoIf($["${CALLERID(num)}"=""]?divert:void)
same => n(divert),Verbose(0,${DB_DELETE(divert/${CALLERID(num):-4})})
same => n,Gosub(divertoff,1)
same => n,Hangup()
same => n(void),Gosub(divertvoid,1)

(note: use whatever you need in the GotoIf to validate that the phone
the call is from is permitted to set up call forwarding - unless you're
allowing it for all that can reach the context)

The divert{off,active,void} subroutines are where I handle the
announcements - but you could probably easily implement your own.

At the top of my [voicemail] context, I do this:

exten=>s,1,GotoIf(${DB_EXISTS(divert/${ARG3})}?outbound-standard,${DB_RESULT},1)

(ARG3 contains the last four digits of the number the call came to in
my case, and a success passes the call via the "outbound-standard"
context which is in my dialplan. Your exact requirements may vary but
this may help.)

-- 
Phil Reynolds
mail: phil-aster...@tinsleyviaduct.com
Web: http://phil.tinsleyviaduct.com/

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Re: [asterisk-users] Dial your phone and contact phone from within outlook?

[A J Stiles]

On Wednesday 02 Mar 2016, Ryan, Travis wrote:
> I am wondering what the best solution is for initiating a call from Outlook
> Contacts. I imagine something that would start the call from the outlook
> card (or similar) and then dial the user's extension and the contact's
> phone number and place them in a bridge.
> 
> Anyone use something like this?

I'm not familiar with Outlook; but I imagine it must have the ability to run 
external helper programs, substituting placeholders in the command line with 
fields from the contact's database record, to perform actions such as dialling 
numbers.

Here, we use Kontact  (part of KDE)  which definitely has such an ability.  So 
I configured it to use wget to fire off a request to a CGI script which 
generates 
a call file; then you just click to dial a contact's number, your phone begins 
to ring, you answer it and the remote contact's phone begins to ring.

The backend script is fairly simple.  We can determine the extension number 
nearest to the user from their IP address  (the DHCP server is configured 
always to offer the same IP address to the same hardware address, so 
workstations effectively have static IP addresses),  which will be passed to 
the script by the Apache server.

If you can fire off a wget request when clicking on a number, you should be 
good 
to go.  (You can request a non-existent script at first; the request will show 
up in /var/log/apache2/error.log .)  If you need help with the script, ask 
again  :)

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?

[Vitor Mazuco]

Humm ok

But my monden not appear in /dev/ and it not show like ttyUSB

I have to install the driver before? Or is not necessary?

Thanks in advanced
Em 03/03/2016 06:13, "Frank Vanoni"  escreveu:

> On Wed, 2016-03-02 at 19:12 -0300, Vitor Mazuco wrote:
>
> > I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but
> > my Huawei E153 is not working in my Asterisk.
> > But not successes.
>
>
> A little more information from you would be helpful to identify the
> problem.
>
> I have a Huawei USB 3G-stick and it works fine on Asterisk 11.
>
> Take a look here:
>
>
> http://www.raspberry-asterisk.org/documentation/gsm-voip-gateway-with-chan_dongle/comment-page-1/
>
> Not all Huawei USB modems work out of the box, on some of them voice
> calling capability has to be enabled first, some need to be upgraded
> with the latest firmware. Details on this can be found on the original
> chan_dongle wiki.
>
> https://github.com/bg111/asterisk-chan-dongle/wiki/Preparation
>
> Before inserting the SIM into your modem please deactivate the PIN on
> your card. This can be done with any phone. Insert the SIM into your
> phone, deactivate PIN and you’re done.
>
>
>
>
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[asterisk-users] How to build and automatic attendant with ASR ?

[Olivier]

Hello,

I'm currently evaluating if it would possible/not too difficult to build
and maintain an automatic attendant application.

More precisely, my requirements are:
- must work with Asterisk
- should be installable on debian or centos
- works this way :
  . caller is asked to tell the first and last names of the person he or
she is after
  . caller answers (with for instance, "John Doe", "Doe John", "Mr Doe",
...)
  . application gets from Asterisk caller's answer and replies to Asterisk
with a code (more probably an extension number) identifying the extension
to dial.

I've seen here and there ASR software compatible with Asterisk.

My questions are:
- should ASR software support caller's language ?
- do ASR allow sysadmins to enter several phonetics per directory entry as
some names can be pronounced differently by different speakers ?
- shall I use an ASR or full automatic attendant app, if such app exists ?
- suggestions ?

Best regards
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Re: [asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?

[Frank Vanoni]

On Wed, 2016-03-02 at 19:12 -0300, Vitor Mazuco wrote:

> I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but
> my Huawei E153 is not working in my Asterisk.
> But not successes.


A little more information from you would be helpful to identify the
problem.

I have a Huawei USB 3G-stick and it works fine on Asterisk 11.

Take a look here:

http://www.raspberry-asterisk.org/documentation/gsm-voip-gateway-with-chan_dongle/comment-page-1/

Not all Huawei USB modems work out of the box, on some of them voice
calling capability has to be enabled first, some need to be upgraded
with the latest firmware. Details on this can be found on the original
chan_dongle wiki.

https://github.com/bg111/asterisk-chan-dongle/wiki/Preparation

Before inserting the SIM into your modem please deactivate the PIN on
your card. This can be done with any phone. Insert the SIM into your
phone, deactivate PIN and you’re done. 




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Re: [asterisk-users] How to install Huawei E153 in a Asterisk 11 or 13?

[Francis Mendoza]

Hi Vitor,

you can try this if it will work for you

http://www.mytechrepublic.com/?p=738


On Thursday, 3 March 2016, Vitor Mazuco  wrote:

> Hi everyone!
>
> I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but
> my Huawei E153 is not working in my Asterisk.
>
> I fallow this rules
> http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14
>
> But not successes.
>
> Thanks in advanced,
>
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