subject:"\[asterisk\-users\] hello"

Re: [asterisk-users] Hello again

2016-09-30 Thread A J Stiles

On Friday 30 Sep 2016, aaberga/gmail wrote:
> Hi,
> 
> after a long pause (Asterisk 1.8 times), I have started again playing with
> VOIP. A lot has changed since last time I did setup an Asterisk system!
> 
> So I am asking for some help.
[stuff deleted]
> [2102]
> type=endpoint
> context=internal
> ;disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> transport=transport-udp-nat
> auth=auth2102
> aors=2102
> rtp_symmetric=yes
> force_rport=yes
> ice_support=yes
> direct_media=no

You might want to comment out all references to g729  (which needs a special 
licence)  and just use alaw  (the native codec of the PSTN)  throughout.

If one of the phones is deciding to use g729 and your Asterisk doesn't have 
the relevant licence, then you might well get all manner of strange things 
happening.

Even if you have g729 licences, try and get it working with alaw first.  The 
fewer things there are that could go wrong, the better.  It's always best to 
get it working with the simplest possible setup first, and only then add 
sophistication.

-- 
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] Hello again

2016-09-30 Thread Joan Aymà [ackstorm]

Your external addres seems wrong. As you are doing natting, you need to
set to you external (natted behind firewall).


El 30/09/16 a les 13:07, aaberga/gmail ha escrit:
> Hi,
>
> after a long pause (Asterisk 1.8 times), I have started again playing with 
> VOIP. A lot has changed since last time I did setup an Asterisk system!
>
> So I am asking for some help.
>
> 
>
> PJSIP seems tougher..
>
> So my problem is that I do have a test system up in the cloud, behind a 
> firewall. I am trying to make the “Hello World!” mandatory call between two 
> iPhones (with the Bria SIP client).
>
> Outcomes are erratic.
>
> 
>
> This is the pjsip.conf file:
>
> ——
>
> [transport-udp-nat]
> type=transport
> protocol=udp
> bind=0.0.0.0
> local_net=10.2.12.3/32
> local_net=127.0.0.1/32
> external_media_address=10.2.12.2
> external_signaling_address=10.2.12.2
>
> ;===Messagenet TRUNK 
>
> [messagenet_reg]
> type=registration
> transport=transport-udp-nat
> outbound_auth=messagenet_auth
> server_uri=sip:xx...@sip.messagenet.it:5061
> client_uri=sip:xx...@sip.messagenet.it:5061
>  
> [messagenet_auth]
> type=auth
> auth_type=userpass
> password=
> username=
>  
> [messagenet_aor]
> type=aor
> contact=sip:sip.messagenet.it:5061
>  
> [messagenet]
> type=endpoint
> transport=transport-udp-nat
> context=messagenet_incoming
> disallow=all
> allow=ulaw
> allow=alaw
> outbound_auth=messagenet_auth
> aors=messagenet_aor
>  
> [messagenet_id]
> type=identify
> endpoint=messagenet
> match=sip.messagenet.it
>  
> ;===Extension 2102
>
> [2102]
> type=endpoint
> context=internal
> ;disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> transport=transport-udp-nat
> auth=auth2102
> aors=2102
> rtp_symmetric=yes
> force_rport=yes
> ice_support=yes
> direct_media=no
>
>
> [auth2102]
> type=auth
> auth_type=userpass
> password=xx
> username=2102
>  
> [2102]
> type=aor
> max_contacts=1
>  
> ;===Extension 2103
>
> [2103]
> type=endpoint
> context=internal
> ;disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> transport=transport-udp-nat
> auth=auth2103
> aors=2103
> rtp_symmetric=yes
> force_rport=yes
> ice_support=yes
> direct_media=no
>  
> [auth2103]
> type=auth
> auth_type=userpass
> password=xx
> username=2103
>  
> [2103]
> type=aor
> max_contacts=1
>  
> 
>
> This is a trace of what I do see from the console.
>
> First I let the Bria clients connect. Then I try to call terminal 1 from 
> terminal 2. Most of the times there is no route to the destination, even if 
> it appears as an online AOR (whatever that means!! Ahhh: Good olde times of 
> Peer, Friend, etc… ;-)
>
> A couple of times I got a connection, with the typical one side only audio of 
> NAT traversal problems.
>
> BTW: The iPhones are behind TWO nats (one is given by the broadband router, 
> one by the WiFi router that gives a better WiFi cover for in-house things).
>
> My understanding is that I did something wrong in letting the phones 
> ‘register’ them as present and available to receive calls. 
>
> If only I knew what is wrong… I have tried random combinations of 
> rtp_symmetric, force_rport, and friends; nothing final discovered...
>
> 
>
> Thanks in advance for any help,
> Aldo
>
>
> PS: Setting up Zoiper as IAX client works of course like a charm on the VOIP 
> side. The only catch is that Zoiper has less than optimal background support 
> on IOS… And I have no plan to make an IAX client myself!
>
> I want to get my old Asterisk apps back online and the VOIP client part makes 
> no sense to me..
>
>

-- 
Joan Aymà
Departamento de SAT
joan.a...@ackstorm.es 

ackstorm 
 
C/ Catalunya 72 Local, 08840 Viladecans - Barcelona
902 888 345  | i...@ackstorm.es
 | **ackstorm.es** 
 
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[asterisk-users] Hello Again - ooops

2016-09-30 Thread aaberga/gmail

Sorry forgot to attach the CLI trace:

=

CLI> pjsip show aors

  Aor:
Contact:  

 
=

  Aor:  210220

  Aor:  210320

  Aor:  messagenet_aor   0
Contact:  messagenet_aor/sip:sip.messagenet.it:5061  Unknown
   nan


-- Added contact 'sip:2103@192.168.155.5:63639;rinstance=eabdf84e26104b07' 
to AOR '2103' with expiration of 900 seconds
-- Removed contact 
'sip:2103@192.168.155.5:63639;rinstance=eabdf84e26104b07' from AOR '2103' due 
to request
-- Added contact 'sip:2103@192.168.155.5:63639;rinstance=420fa67d404d9816' 
to AOR '2103' with expiration of 900 seconds
-- Removed contact 
'sip:2103@192.168.155.5:63639;rinstance=420fa67d404d9816' from AOR '2103' due 
to request
-- Added contact 'sip:2103@37.228.255.229:60677;rinstance=635ece4650faa34e' 
to AOR '2103' with expiration of 900 seconds
-- Added contact 'sip:2102@192.168.155.5:60157;rinstance=0833518dac88d43b' 
to AOR '2102' with expiration of 900 seconds
-- Removed contact 
'sip:2102@192.168.155.5:60157;rinstance=0833518dac88d43b' from AOR '2102' due 
to request
-- Added contact 'sip:2102@192.168.155.5:60157;rinstance=917b91dc461a6eda' 
to AOR '2102' with expiration of 900 seconds
-- Removed contact 
'sip:2102@192.168.155.5:60157;rinstance=917b91dc461a6eda' from AOR '2102' due 
to request
-- Added contact 'sip:2102@37.228.255.229:60605;rinstance=fbd37b6a6d7cb4fb' 
to AOR '2102' with expiration of 900 seconds


-- Executing [2102@internal:1] Set("PJSIP/2103-0003", "ORIGIN=IP") in 
new stack
-- Executing [2102@internal:2] NoOp("PJSIP/2103-0003", "Declared 
CallerID=<"2103" <2103>>") in new stack
-- Executing [2102@internal:3] Set("PJSIP/2103-0003", 
"CALLERID(name)=Insicure-IP") in new stack
-- Executing [2102@internal:4] Set("PJSIP/2103-0003", 
"OriginalEXTEN=2102") in new stack
-- Executing [2102@internal:5] Set("PJSIP/2103-0003", 
"CDR(userfield)=2102") in new stack
-- Executing [2102@internal:6] Goto("PJSIP/2103-0003", 
"dialplan-switch,2102,1") in new stack
-- Goto (dialplan-switch,2102,1)
-- Executing [2102@dialplan-switch:1] NoOp("PJSIP/2103-0003", " 
Entering Dialplan Switch from  ") in new stack
-- Executing [2102@dialplan-switch:2] Dial("PJSIP/2103-0003", 
"PJSIP/2102") in new stack
[Sep 30 10:50:44] ERROR[19237]: res_pjsip.c:2106 
sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 
'transport-udp-nat' for endpoint '2102'
[Sep 30 10:50:44] ERROR[19237]: chan_pjsip.c:1788 request: Failed to create 
outgoing session to endpoint '2102'
[Sep 30 10:50:44] WARNING[19287][C-0003]: app_dial.c:2431 dial_exec_full: 
Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2102@dialplan-switch:3] Hangup("PJSIP/2103-0003", "") in 
new stack
  == Spawn extension (dialplan-switch, 2102, 3) exited non-zero on 
'PJSIP/2103-0003'
  
  
-- Executing [2103@internal:1] Set("PJSIP/2102-0004", "ORIGIN=IP") in 
new stack
-- Executing [2103@internal:2] NoOp("PJSIP/2102-0004", "Declared 
CallerID=<"2102" <2102>>") in new stack
-- Executing [2103@internal:3] Set("PJSIP/2102-0004", 
"CALLERID(name)=Insicure-IP") in new stack
-- Executing [2103@internal:4] Set("PJSIP/2102-0004", 
"OriginalEXTEN=2103") in new stack
-- Executing [2103@internal:5] Set("PJSIP/2102-0004", 
"CDR(userfield)=2103") in new stack
-- Executing [2103@internal:6] Goto("PJSIP/2102-0004", 
"dialplan-switch,2103,1") in new stack
-- Goto (dialplan-switch,2103,1)
-- Executing [2103@dialplan-switch:1] NoOp("PJSIP/2102-0004", " 
Entering Dialplan Switch from  ") in new stack
-- Executing [2103@dialplan-switch:2] Dial("PJSIP/2102-0004", 
"PJSIP/2103") in new stack
[Sep 30 10:52:01] ERROR[19299]: res_pjsip.c:2106 
sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 
'transport-udp-nat' for endpoint '2103'
[Sep 30 10:52:01] ERROR[19299]: chan_pjsip.c:1788 request: Failed to create 
outgoing session to endpoint '2103'
[Sep 30 10:52:01] WARNING[19306][C-0004]: app_dial.c:2431 dial_exec_full: 
Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2103@dialplan-switch:3] Hangup("PJSIP/2102-0004", "") in 
new stack
  == Spawn extension (dialplan-switch, 2103, 3) exited non-zero on 
'PJSIP/2102-0004'


=



Tnx,
Aldo


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[asterisk-users] Hello again

2016-09-30 Thread aaberga/gmail

Hi,

after a long pause (Asterisk 1.8 times), I have started again playing with 
VOIP. A lot has changed since last time I did setup an Asterisk system!

So I am asking for some help.



PJSIP seems tougher..

So my problem is that I do have a test system up in the cloud, behind a 
firewall. I am trying to make the “Hello World!” mandatory call between two 
iPhones (with the Bria SIP client).

Outcomes are erratic.



This is the pjsip.conf file:

——

[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.2.12.3/32
local_net=127.0.0.1/32
external_media_address=10.2.12.2
external_signaling_address=10.2.12.2

;===Messagenet TRUNK 

[messagenet_reg]
type=registration
transport=transport-udp-nat
outbound_auth=messagenet_auth
server_uri=sip:xx...@sip.messagenet.it:5061
client_uri=sip:xx...@sip.messagenet.it:5061
 
[messagenet_auth]
type=auth
auth_type=userpass
password=
username=
 
[messagenet_aor]
type=aor
contact=sip:sip.messagenet.it:5061
 
[messagenet]
type=endpoint
transport=transport-udp-nat
context=messagenet_incoming
disallow=all
allow=ulaw
allow=alaw
outbound_auth=messagenet_auth
aors=messagenet_aor
 
[messagenet_id]
type=identify
endpoint=messagenet
match=sip.messagenet.it
 
;===Extension 2102

[2102]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2102
aors=2102
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no


[auth2102]
type=auth
auth_type=userpass
password=xx
username=2102
 
[2102]
type=aor
max_contacts=1
 
;===Extension 2103

[2103]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2103
aors=2103
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no
 
[auth2103]
type=auth
auth_type=userpass
password=xx
username=2103
 
[2103]
type=aor
max_contacts=1
 


This is a trace of what I do see from the console.

First I let the Bria clients connect. Then I try to call terminal 1 from 
terminal 2. Most of the times there is no route to the destination, even if it 
appears as an online AOR (whatever that means!! Ahhh: Good olde times of Peer, 
Friend, etc… ;-)

A couple of times I got a connection, with the typical one side only audio of 
NAT traversal problems.

BTW: The iPhones are behind TWO nats (one is given by the broadband router, one 
by the WiFi router that gives a better WiFi cover for in-house things).

My understanding is that I did something wrong in letting the phones ‘register’ 
them as present and available to receive calls. 

If only I knew what is wrong… I have tried random combinations of 
rtp_symmetric, force_rport, and friends; nothing final discovered...



Thanks in advance for any help,
Aldo


PS: Setting up Zoiper as IAX client works of course like a charm on the VOIP 
side. The only catch is that Zoiper has less than optimal background support on 
IOS… And I have no plan to make an IAX client myself!

I want to get my old Asterisk apps back online and the VOIP client part makes 
no sense to me..


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[asterisk-users] hello

2013-07-07 Thread Safarifone Technical Support Hassan Caynte




Sent from Windows Mail--
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[asterisk-users] hello

2011-04-03 Thread ALAEDDINE abbech


I have bought a pair of Apple phones on---www.ofenno.com---they have pretty 
good quality. Here I would like to recommend it to you. Their company is 
holding a promotion activity now, so you can buy anything you want on it with 
free delivery charges. There must be anything you like, I hope you would not 
miss this  chance.
All The Best--
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Re: [asterisk-users] hello

2011-04-03 Thread Bradley D. Thornton

-BEGIN PGP SIGNED MESSAGE-
Hash: RIPEMD160

WTF does that have to do with Asterisk?

On 04/03/2011 05:56 AM, ALAEDDINE abbech wrote:
 
 I have bought a pair of Apple phones on---www.ofenno.com---they have pretty 
 good quality. Here I would like to recommend it to you. Their company is 
 holding a promotion activity now, so you can buy anything you want on it with 
 free delivery charges. There must be anything you like, I hope you would not 
 miss this  chance.
 All The Best
 
 
 
 --
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 asterisk-users mailing list
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- -- 
Bradley D. Thornton
Manager Network Services
NorthTech Computer
TEL: +1.760.666.2703  (US)
TEL: +44.702.405.1909 (UK)
http://NorthTech.US

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Version: GnuPG v1.4.10 (GNU/Linux)
Comment: Find this cert at x-hkp://pool.sks-keyservers.net

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Re: [asterisk-users] hello

2011-04-03 Thread Doug Lytle


Bradley D. Thornton wrote:

WTF does that have to do with Asterisk?

   


It's called spam.  And either he doesn't know what non-commercial 
discussion means or he signed up just to send this.  Doubtful we'll see 
him again.


Doug


--
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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[asterisk-users] Hello and Music on Hold question

2009-06-11 Thread Ishfaq Malik

Hello all, I have just joined this list and I'm currently working with 
asterisk 1.4.17 using RealTime. Not quite sure the level of queries you 
get but hopefully I'll be able to help with some input as well as 
questions of my own.

Now to my first query. I'm changing the hold music on our system and 
I've done this by deleting the old sound file that was in the music on 
hold directory as defined by the /etc/asterisk/musiconhold.conf and put 
a new one in there but when going on hold I can still hear the old 
music. Is there a cache somewhere that I need to clear?

I have tried moh reload from the console but that just reloads the conf 
file.

Thanks

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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[asterisk-users] Hello Asterisk list. BerkeleyTIP you

2009-01-03 Thread john_re

Hi Asterisk list readers  :) 

I thought you might like to be aware that:

The Asterisk VOIP conferencing system is being investigated for
communication for a global GNU(Linux), BSD  All Free SW  HW monthly
meeting, called BerkeleyTIP.

Below is some more info.

I'll also send 2 following emails:

1) BerkeleyTIP - Hello, Introduction, Monthly Global GNU(Linux) BSD
FreeSWHW meeting

2) BerkeleyTIP TODAY Jan 3 Sat- Party Time  :)  Video Talks: Asterisk,
GPU


=
Here's an alert to several things you might be interested in:

1) There is a global monthly GNU(Linux) BSD Free SW  HW meeting, called
BerkeleyTIP
2) TIP = Talks, Installfest, Potluck  ProgrammingParty.
3) It's an educational, productive, social, fun event.
4) We communicate by VOIP, using a VOIP conference system (Ekiga's,
iirc), using Ekiga client sw.
5) You might enjoy attending the meeting.  , you're invited.
6) The VOIP conference server is suboptimal, so I  as many people as I
can rope into it are going to maybe try  learn about voip conference
servers  clients,  improve them to better get BTIPs needs served.  -
Like using Ekiga  Asterisk (for conference server) sw.
7) You might want keep up on the goings on.
8) http://groups.google.com/group/BerkTIP 
http://groups.google.com/group/BerkTIPGlobal
are a good place to start.

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[asterisk-users] Hello Everybody, my problem with voicemail.conf

2007-01-26 Thread Ashish Barot


Hello everybody i am Ashish here.
i am new to this mailing list.
so dont know rules and regulation, just trying to post my problem of
voicemail.conf


Actuallt right now i am using Asterisk 1.2 on my LAN environment.
i am able to call all my extension very nicely.

Right now i am trying to deploying voicemail facility for all
extensions, so if anybody is not present, then he/she can leave
message, and that message will be e-mail to particular e-mail id which
i am using in
extension.conf

Upto this moment the voicemail is generating, but it is not e-mail to
any email id. But it comes on
[EMAIL PROTECTED]
that i had check with K-mail application.
Whole message is coming in .wav file extension. on [EMAIL PROTECTED]
also i get few text message from [EMAIL PROTECTED]
on my gmail's spam folder. but in gmail a.c no attachment is coming.

so pl. any body can help me for it

below i am sending my sip.conf , extensions.conf and voicemail.conf

Pl,do the needful.

Thanks.
With Warm Regards,
Ashish Barot.

-
*sip.conf*

[general]
bindport=5060
context=worldbiz
[EMAIL PROTECTED]
nat=yes
allow=all

[]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
[EMAIL PROTECTED]

[1112]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
[EMAIL PROTECTED]

[1113]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
[EMAIL PROTECTED]

-
*extensions.conf

*[general]
static=yes
writeprotect=no


[worldbiz]
exten = _111X,1,Dial(SIP/${EXTEN},4)
exten = s-BUSY,2,Goto(s,1)
exten = _111X,2,VoiceMail([EMAIL PROTECTED])
exten = _111X,3,SendText(Hello I am Ashish Barot here)
exten = _111X,4,Hangup()
exten = _111X,103,SendText(This is my test voice mail message. Try to reply
me)
exten = _111X,104,Hangup()*



voicemail.conf
**
*[general]
attach=yes
[EMAIL PROTECTED]
format=wav
minmessage=0
maxmessage=0


[worldbiz]
 = 1234,Barot,[EMAIL PROTECTED]
1112 = 1234,Ashish Barot,[EMAIL PROTECTED],,attach=yes|format=wav

-

i had search lots of config. files on net from all of them i had prepare
above files. but still not getting it resolve.
so pl. try to reply.

Thanks.
Ashish Barot.
*
**



**
*
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Re: [asterisk-users] Hello Everybody, my problem with voicemail.conf

2007-01-26 Thread Stefan Schmidt


hello,

maybe you should try adding this to your voicemail configuration.

mailcmd=/usr/sbin/sendmail -t

or whereever your sendmail is located.

then your mails should be send to the wanted adress.

Best regards.

Stefan


Ashish Barot schrieb:

Hello everybody i am Ashish here.
i am new to this mailing list.
so dont know rules and regulation, just trying to post my problem of 
voicemail.conf



snipped



Für weitere Fragen stehen wir natürlich gerne unter [EMAIL PROTECTED] oder
059944 - 2010 zur Verfügung.

Mit freundlichen Grüssen
--
Stefan Schmidt
Support/VOIP // [EMAIL PROTECTED] // Tel 059944-2010  //
-
SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 //
A-1160 Wien // Fax 059944-9000 // www.sil.at  //
-

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Re: [asterisk-users] Hello Everybody, my problem with voicemail.conf

2007-01-26 Thread Andy Davidson



On 26 Jan 2007, at 10:43, Ashish Barot wrote:

Upto this moment the voicemail is generating, but it is not e-mail  
to any email id. But it comes on [EMAIL PROTECTED]

[...]

[worldbiz]
exten = _111X,1,Dial(SIP/${EXTEN},4)
exten = s-BUSY,2,Goto(s,1)
exten = _111X,2,VoiceMail([EMAIL PROTECTED])

[...]

[worldbiz]
 = 1234,Barot,[EMAIL PROTECTED]
1112 = 1234,Ashish Barot,[EMAIL PROTECTED],,attach=yes|format=wav



All of your extensions are configured to pass voicemail into mailbox  
 - this is configured to send VM notifications to [EMAIL PROTECTED]  
(i.e. no voicemail is ever hitting mailbox 1112).


-a

--
Regards, Andy Davidson
http://www.devonshire.it/  -  0844 704 704 7  - Sheffield, UK


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[Asterisk-Users] hello

2005-11-23 Thread harry gaillac

hello






___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com
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Re: [Asterisk-Users] hello

2005-11-23 Thread Matt Riddell

harry gaillac wrote:
 hello

Hi there! How are you today?

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

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Re: [Asterisk-Users] hello

2005-11-23 Thread pdhales

 harry gaillac wrote:
  hello
 
 Hi there! How are you today?

Very well, thank you.

PaulH
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[Asterisk-Users] Hello Everyone

2005-03-25 Thread Bagan Jermal

would like to test this e-mail list.
anyway, have anybody here install and run [EMAIL PROTECTED] how was it?
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RE: [Asterisk-Users] Hello Everyone

2005-03-25 Thread Ariel Batista

Welcome,

Yes I have used it. It's great to get started. Give it a try.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bagan Jermal
Sent: Friday, March 25, 2005 6:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hello Everyone

would like to test this e-mail list.

anyway, have anybody here install and run [EMAIL PROTECTED] how was it?

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[Asterisk-Users] hello

2004-11-19 Thread Rogerio Santos











New user * 



Test Brasil 






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