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murthy64 at hotmail.com
Guest
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 Posted: Wed Aug 05, 2015 4:37 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
 |
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Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank" class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
It was not working. So I downloaded X-PRO Vonage, the vonage sip phone, and wiresharked the port. I see that a significant difference is the vonage phone uses "Vonage User" where
asterisk uses "Anonymous". Is that the problem? The Inbound call works fine. Here is my sip.conf
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register =><did>:<password>@69.59.234.67:5060/202
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:<did>@69.59.234.67>;tag=as69898393'
ubuntu*CLI>
Thanks for your help
murthy
--
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murthy64 at hotmail.com
Guest
|
| Posted: Wed Aug 05, 2015 4:37 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank" class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
It was not working. So I downloaded X-PRO Vonage, the vonage sip phone, and wiresharked the port. I see that a significant difference is the vonage phone uses "Vonage User" where
asterisk uses "Anonymous". Is that the problem? The Inbound call works fine. Here is my sip.conf
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register =><did>:<password>@69.59.234.67:5060/202
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:<did>@69.59.234.67>;tag=as69898393'
ubuntu*CLI>
Thanks for your help
murthy
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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murthy64 at hotmail.com
Guest
|
| Posted: Wed Aug 05, 2015 4:38 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank" class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
It was not working. So I downloaded X-PRO Vonage, the vonage sip phone, and wiresharked the port. I see that a significant difference is the vonage phone uses "Vonage User" where
asterisk uses "Anonymous". Is that the problem? The Inbound call works fine. Here is my sip.conf
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register =><did>:<password>@69.59.234.67:5060/202
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:<did>@69.59.234.67>;tag=as69898393'
ubuntu*CLI>
Thanks for your help
murthy
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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murthy64 at hotmail.com
Guest
|
| Posted: Wed Aug 05, 2015 4:38 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank" class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
It was not working. So I downloaded X-PRO Vonage, the vonage sip phone, and wiresharked the port. I see that a significant difference is the vonage phone uses "Vonage User" where
asterisk uses "Anonymous". Is that the problem? The Inbound call works fine. Here is my sip.conf
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register =><did>:<password>@69.59.234.67:5060/202
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:<did>@69.59.234.67>;tag=as69898393'
ubuntu*CLI>
Thanks for your help
murthy
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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murthy64 at hotmail.com
Guest
|
| Posted: Thu Aug 06, 2015 11:56 am Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
Tested with X-Lite and it worked fiine. Is there some way to replace "Anonymous" with a config parameter?
Thanks for your kind help
----------------------------------------
| Quote: | From: murthy64@hotmail.com
To: asterisk-users@lists.digium.com
Subject: Asterisk uses "Anonymous", but why?
Date: Wed, 5 Aug 2015 21:38:16 +0000
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank" class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
It was not working. So I downloaded X-PRO Vonage, the vonage sip phone, and wiresharked the port. I see that a significant difference is the vonage phone uses "Vonage User" where
asterisk uses "Anonymous". Is that the problem? The Inbound call works fine. Here is my sip.conf
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register =><did>:<password>@69.59.234.67:5060/202
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:<did>@69.59.234.67>;tag=as69898393'
ubuntu*CLI>
Thanks for your help
murthy
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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rmudgett at digium.com
Guest
|
| Posted: Thu Aug 06, 2015 12:07 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64@hotmail.com (murthy64@hotmail.com)> wrote:
| Quote: | Tested with X-Lite and it worked fiine. Is there some way to replace "Anonymous" with a config parameter?
Thanks for your kind help
----------------------------------------
| Quote: |
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank" class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
It was not working. So I downloaded X-PRO Vonage, the vonage sip phone, and wiresharked the port. I see that a significant difference is the vonage phone uses "Vonage User" where
asterisk uses "Anonymous". Is that the problem? The Inbound call works fine. Here is my sip.conf
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register =><did>:<password>@69.59.234.67:5060/202
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:<did>@69.59.234.67>;tag=as69898393'
ubuntu*CLI>
|
|
Use the AMI Originate action or a call file. You can specify a caller id there. You cannot specify one from the command line.
Richard |
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|
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RyanT at OscarWinski.com
Guest
|
| Posted: Thu Aug 06, 2015 12:12 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Thursday, August 06, 2015 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64@hotmail.com (murthy64@hotmail.com)> wrote:
Tested with X-Lite and it worked fiine. Is there some way to replace "Anonymous" with a config parameter?
Thanks for your kind help
----------------------------------------
| Quote: |
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank" class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
It was not working. So I downloaded X-PRO Vonage, the vonage sip phone, and wiresharked the port. I see that a significant difference is the vonage phone uses "Vonage User" where
asterisk uses "Anonymous". Is that the problem? The Inbound call works fine. Here is my sip.conf
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register =><did>:<password>@69.59.234.67:5060/202
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <[url=sip:%3cdid]sip:<did[/url]>@69.59.234.67>;tag=as69898393'
ubuntu*CLI>
|
Use the AMI Originate action or a call file. You can specify a caller id there. You cannot specify one from the command line.
Richard
[Ryan, Travis]
Are you sure? I have no issue with a PRI line and using the set command like so…Unless it’s a toll free number.
Set(CALLERID(num)=7656371111) |
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murthy64 at hotmail.com
Guest
|
| Posted: Thu Aug 06, 2015 12:34 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
________________________________
| Quote: | Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudgett@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota
<murthy64@hotmail.com<mailto:murthy64@hotmail.com>> wrote:
Tested with X-Lite and it worked fiine. Is there some way to replace
"Anonymous" with a config parameter?
Thanks for your kind help
----------------------------------------
| Quote: | From: murthy64@hotmail.com<mailto:murthy64@hotmail.com>
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Subject: Asterisk uses "Anonymous", but why?
Date: Wed, 5 Aug 2015 21:38:16 +0000
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage"
|
target="_blank"
class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
| Quote: |
It was not working. So I downloaded X-PRO Vonage, the vonage sip
|
phone, and wiresharked the port. I see that a significant difference is
the vonage phone uses "Vonage User" where
| Quote: | asterisk uses "Anonymous". Is that the problem? The Inbound call
|
works fine. Here is my sip.conf
| Quote: |
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register
|
=><did>:<password>@69.59.234.67:5060/202<http://69.59.234.67:5060/202>
| Quote: |
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160
|
handle_response_invite: Received response: "Forbidden" from
'"Anonymous"
<sip:<did>@69.59.234.67<http://69.59.234.67>>;tag=as69898393'
Use the AMI Originate action or a call file. You can specify a caller
id there. You cannot specify one from the command line.
Richard
|
Hi Richard
What should I use for extension? Since I am not bridging an extension with outbound, but making an outbound call and playing a sound file, what would be the extension?
Here is my Asterisk-Java code:
managerConnection.addEventListener(this);
originateAction = new OriginateAction();
originateAction.setChannel("SIP/"+ani);
originateAction.setContext("from-pstn");
originateAction.setExten(????);
originateAction.setPriority(new Integer(1));
originateAction.setCallerId("murthy");
originateAction.setTimeout(new Integer(30000));
// connect to Asterisk and log in
managerConnection.login();
// send the originate action and wait for a maximum of 30 seconds for Asterisk
// to send a reply
originateResponse = managerConnection.sendAction(originateAction, 30000);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxxxxxx => {
Answer();
Playback(welcomesystole);
Read(digito1,,3);
Playback(diastole);
Read(digito2,,3);
Agi(agi://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2});
Hangup()
}
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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murthy64 at hotmail.com
Guest
|
| Posted: Thu Aug 06, 2015 12:55 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
----------------------------------------
| Quote: | From: murthy64@hotmail.com
To: asterisk-users@lists.digium.com
Date: Thu, 6 Aug 2015 17:33:37 +0000
Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
________________________________
| Quote: | Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudgett@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota
<murthy64@hotmail.com<mailto:murthy64@hotmail.com>> wrote:
Tested with X-Lite and it worked fiine. Is there some way to replace
"Anonymous" with a config parameter?
Thanks for your kind help
----------------------------------------
| Quote: | From: murthy64@hotmail.com<mailto:murthy64@hotmail.com>
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Subject: Asterisk uses "Anonymous", but why?
Date: Wed, 5 Aug 2015 21:38:16 +0000
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage"
|
target="_blank"
class="newlyinsertedlink">http://www.voip-info.org/wiki/view/Asterisk+and+Vonage</a>
| Quote: |
It was not working. So I downloaded X-PRO Vonage, the vonage sip
|
phone, and wiresharked the port. I see that a significant difference is
the vonage phone uses "Vonage User" where
| Quote: | asterisk uses "Anonymous". Is that the problem? The Inbound call
|
works fine. Here is my sip.conf
| Quote: |
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes
register
|
=><did>:<password>@69.59.234.67:5060/202<http://69.59.234.67:5060/202>
| Quote: |
[vonage-out]
username=<did>
type=friend
secret=<password>
port=5061
nat=yes
host=69.59.234.67
fromuser=<did>
fromdomain=69.59.234.67
dtmfmode=rfc2833
auth=md5
context=from-pstn
canreinvite=no
Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160
|
handle_response_invite: Received response: "Forbidden" from
'"Anonymous"
<sip:<did>@69.59.234.67<http://69.59.234.67>>;tag=as69898393'
Use the AMI Originate action or a call file. You can specify a caller
id there. You cannot specify one from the command line.
Richard
|
Hi Richard
What should I use for extension? Since I am not bridging an extension with outbound, but making an outbound call and playing a sound file, what would be the extension?
Here is my Asterisk-Java code:
managerConnection.addEventListener(this);
originateAction = new OriginateAction();
originateAction.setChannel("SIP/"+ani);
originateAction.setContext("from-pstn");
originateAction.setExten(????);
originateAction.setPriority(new Integer(1));
originateAction.setCallerId("murthy");
originateAction.setTimeout(new Integer(30000));
// connect to Asterisk and log in
managerConnection.login();
// send the originate action and wait for a maximum of 30 seconds for Asterisk
// to send a reply
originateResponse = managerConnection.sendAction(originateAction, 30000);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxxxxxx => {
Answer();
Playback(welcomesystole);
Read(digito1,,3);
Playback(diastole);
Read(digito2,,3);
Agi(agi://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2});
Hangup()
}
|
I used the "s" for exten, and added extension s to the from-pstn context thus:
managerConnection.addEventListener(this);
originateAction = new OriginateAction();
originateAction.setChannel("SIP/"+ani+"@vonage-out");
originateAction.setContext("from-pstn");
originateAction.setExten("s");
originateAction.setPriority(new Integer(1));
originateAction.setCallerId("Vonage User");
originateAction.setTimeout(new Integer(30000));
// connect to Asterisk and log in
managerConnection.login();
// send the originate action and wait for a maximum of 30 seconds for Asterisk
// to send a reply
originateResponse = managerConnection.sendAction(originateAction, 30000);
// print out whether the originate succeeded or not
System.out.println(originateResponse.getResponse());
context from-pstn {
s => {
Answer();
Playback(welcomesystole);
Read(digito1,,3);
Playback(diastole);
Read(digito2,,3);
Agi(agi://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2});
Hangup();
}
}
Now I get
[Aug 6 10:50:32] WARNING[25977][C-0000000b]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Vonage User" <sip:1619xxxxxxx@69.59.234.67>;tag=as46f9ddef'
ubuntu*CLI>
Regards
--
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rmudgett at digium.com
Guest
|
| Posted: Thu Aug 06, 2015 12:55 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64@hotmail.com (murthy64@hotmail.com)> wrote:
| Quote: |
________________________________
|
<snip>
Â
| Quote: | >> Here is the CLI command used:
| Quote: | | Quote: |
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160
|
handle_response_invite: Received response: "Forbidden" from
'"Anonymous"
<sip:<did>@69.59.234.67<http://69.59.234.67>>;tag=as69898393'
Use the AMI Originate action or a call file. You can specify a caller
id there. You cannot specify one from the command line.
Richard
|
Hi Richard
What should I use for extension? Since I am not bridging an extension with outbound, but making an outbound call and playing a sound file, what would be the extension?
Here is my Asterisk-Java code:
 managerConnection.addEventListener(this);
         originateAction = new OriginateAction();
         originateAction.setChannel("SIP/"+ani);
         originateAction.setContext("from-pstn");
         originateAction.setExten(????);
         originateAction.setPriority(new Integer(1));
         originateAction.setCallerId("murthy");
         originateAction.setTimeout(new Integer(30000));
         // connect to Asterisk and log in
         managerConnection.login();
         // send the originate action and wait for a maximum of 30 seconds for Asterisk
         // to send a reply
         originateResponse = managerConnection.sendAction(originateAction, 30000);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
    1619xxxxxxx => {
|
This looks like a dialplan pattern match exten but you do not have a leading '_' to indicate
that it is a pattern so this exten will only match a literal "1619xxxxxxx".
Â
| Quote: | Â Â Â Â Â Â Â Â Answer();
        Playback(welcomesystole);
        Read(digito1,,3);
        Playback(diastole);
        Read(digito2,,3);
        Agi(agi://[url=http://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2}]10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2}[/url]);
        Hangup()
}
|
It is up to you where you want to send the originated call to in your dialplan. Since you
appear to want to send it to an extension that is a pattern you need to use a value that
the pattern will match such as 16190000000.
Richard |
|
| Back to top |
|
|
murthy64 at hotmail.com
Guest
|
| Posted: Thu Aug 06, 2015 1:26 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
________________________________
| Quote: | Date: Thu, 6 Aug 2015 12:55:28 -0500
From: rmudgett@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota
<murthy64@hotmail.com<mailto:murthy64@hotmail.com>> wrote:
________________________________
<snip>
| Quote: | | Quote: | Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160
|
handle_response_invite: Received response: "Forbidden" from
'"Anonymous"
|
<sip:<did>@69.59.234.67<http://69.59.234.67><http://69.59.234.67>>;tag=as69898393'
| Quote: |
Use the AMI Originate action or a call file. You can specify a caller
id there. You cannot specify one from the command line.
Richard
|
Hi Richard
What should I use for extension? Since I am not bridging an extension
with outbound, but making an outbound call and playing a sound file,
what would be the extension?
Here is my Asterisk-Java code:
managerConnection.addEventListener(this);
originateAction = new OriginateAction();
originateAction.setChannel("SIP/"+ani);
originateAction.setContext("from-pstn");
originateAction.setExten(????);
originateAction.setPriority(new Integer(1));
originateAction.setCallerId("murthy");
originateAction.setTimeout(new Integer(30000));
// connect to Asterisk and log in
managerConnection.login();
// send the originate action and wait for a maximum of
30 seconds for Asterisk
// to send a reply
originateResponse =
managerConnection.sendAction(originateAction, 30000);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxxxxxx => {
This looks like a dialplan pattern match exten but you do not have a
leading '_' to indicate
that it is a pattern so this exten will only match a literal "1619xxxxxxx".
Answer();
Playback(welcomesystole);
Read(digito1,,3);
Playback(diastole);
Read(digito2,,3);
Agi(agi://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2}<http://10.10.22.171:4573/hello.agi?systole=$%7bdigito1%7d&diastole=$%7bdigito2%7d>);
Hangup()
}
It is up to you where you want to send the originated call to in your
dialplan. Since you
appear to want to send it to an extension that is a pattern you need to
use a value that
the pattern will match such as 16190000000.
Richard
|
Hi Richard
Thank you for your suggestions. The responses received are:
[Aug 6 11:20:28] NOTICE[25977][C-0000001a]: chan_sip.c:23147 handle_response_invite: Failed to authenticate on INVITE to '"Vonage User" <sip:1619xxxxxxx@69.59.234.67>;tag=as0bf485e8'
> Channel SIP/vonage202-00000019 was never answered.
I don't understand the "Channel SIP/vonage202-00000019 was never answered".... your kind clarification is sought.
Regards
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
rmudgett at digium.com
Guest
|
| Posted: Thu Aug 06, 2015 1:33 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota <murthy64@hotmail.com (murthy64@hotmail.com)> wrote:
| Quote: |
________________________________
| Quote: | Date: Thu, 6 Aug 2015 12:55:28 -0500
From: rmudgett@digium.com (rmudgett@digium.com)
To: asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota
<murthy64@hotmail.com (murthy64@hotmail.com)<mailto:murthy64@hotmail.com (murthy64@hotmail.com)>> wrote:
________________________________
<snip>
| Quote: | | Quote: | Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160
|
handle_response_invite: Received response: "Forbidden" from
'"Anonymous"
|
<sip:<did>@69.59.234.67<http://69.59.234.67><http://69.59.234.67>>;tag=as69898393'
| Quote: |
Use the AMI Originate action or a call file. You can specify a caller
id there. You cannot specify one from the command line.
Richard
|
Hi Richard
What should I use for extension? Since I am not bridging an extension
with outbound, but making an outbound call and playing a sound file,
what would be the extension?
Here is my Asterisk-Java code:
managerConnection.addEventListener(this);
originateAction = new OriginateAction();
originateAction.setChannel("SIP/"+ani);
originateAction.setContext("from-pstn");
originateAction.setExten(????);
originateAction.setPriority(new Integer(1));
originateAction.setCallerId("murthy");
originateAction.setTimeout(new Integer(30000));
// connect to Asterisk and log in
managerConnection.login();
// send the originate action and wait for a maximum of
30 seconds for Asterisk
// to send a reply
originateResponse =
managerConnection.sendAction(originateAction, 30000);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxxxxxx => {
This looks like a dialplan pattern match exten but you do not have a
leading '_' to indicate
that it is a pattern so this exten will only match a literal "1619xxxxxxx".
Answer();
Playback(welcomesystole);
Read(digito1,,3);
Playback(diastole);
Read(digito2,,3);
|
| Quote: | Agi(agi://[url=http://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2}]10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2}[/url]<http://10.10.22.171:4573/hello.agi?systole=$%7bdigito1%7d&diastole=$%7bdigito2%7d>);
Hangup()
}
It is up to you where you want to send the originated call to in your
dialplan. Since you
appear to want to send it to an extension that is a pattern you need to
use a value that
the pattern will match such as 16190000000.
Richard
|
Hi Richard
Thank you for your suggestions. The responses received are:
[Aug  6 11:20:28] NOTICE[25977][C-0000001a]: chan_sip.c:23147 handle_response_invite: Failed to authenticate on INVITE to '"Vonage User" <sip:1619xxxxxxx@69.59.234.67 ([email]sip%3A1619xxxxxxx@69.59.234.67[/email])>;tag=as0bf485e8'
    > Channel SIP/vonage202-00000019 was never answered.
 Â
I don't understand the "Channel SIP/vonage202-00000019 was never answered".... your kind clarification is sought.
|
What do you think "Failed to authenticate" on the call you just originated means?
Your call was rejected and thus the call was never answered. You have an
authentication problem. Vonage could not authenticate the call you originated.
Richard |
|
| Back to top |
|
|
asterisk.org at sedwar...
Guest
|
| Posted: Thu Aug 06, 2015 1:44 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
On Thu, 6 Aug 2015, Murthy Gandikota wrote:
[trimming cruft irrelvant to the current issue]
| Quote: | [Aug 6 11:20:28] NOTICE[25977][C-0000001a]: chan_sip.c:23147
handle_response_invite: Failed to authenticate on INVITE to '"Vonage
User" <sip:1619xxxxxxx@69.59.234.67>;tag=as0bf485e8' > Channel
SIP/vonage202-00000019 was never answered. I don't understand the
"Channel SIP/vonage202-00000019 was never answered".... your kind
clarification is sought.
|
"Failed to authenticate on INVITE"
Sounds like something you could work out with wireshark and Vonage
support.
My SIP needs are small, but I've always been happy with vitelity.com.
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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murthy64 at hotmail.com
Guest
|
| Posted: Thu Aug 06, 2015 1:54 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
________________________________
| Quote: | Date: Thu, 6 Aug 2015 13:33:11 -0500
From: rmudgett@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota
<murthy64@hotmail.com<mailto:murthy64@hotmail.com>> wrote:
________________________________
| Quote: | Date: Thu, 6 Aug 2015 12:55:28 -0500
From: rmudgett@digium.com<mailto:rmudgett@digium.com>
To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota
|
<murthy64@hotmail.com<mailto:murthy64@hotmail.com><mailto:murthy64@hotmail.com<mailto:murthy64@hotmail.com>>>
wrote:
| Quote: |
________________________________
| Quote: | Date: Thu, 6 Aug 2015 12:07:35 -0500
From:
|
|
rmudgett@digium.com<mailto:rmudgett@digium.com><mailto:rmudgett@digium.com<mailto:rmudgett@digium.com>>
asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com><mailto:asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>>
| Quote: | | Quote: | Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why?
|
<snip>
| Quote: | | Quote: | Here is the CLI command used:
ubuntu*CLI> originate SIP/732-xxx-xxxx@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160
|
handle_response_invite: Received response: "Forbidden" from
'"Anonymous"
|
|
<sip:<did>@69.59.234.67<http://69.59.234.67><http://69.59.234.67><http://69.59.234.67>>;tag=as69898393'
| Quote: | | Quote: |
Use the AMI Originate action or a call file. You can specify a caller
id there. You cannot specify one from the command line.
Richard
|
Hi Richard
What should I use for extension? Since I am not bridging an extension
with outbound, but making an outbound call and playing a sound file,
what would be the extension?
Here is my Asterisk-Java code:
managerConnection.addEventListener(this);
originateAction = new OriginateAction();
originateAction.setChannel("SIP/"+ani);
originateAction.setContext("from-pstn");
originateAction.setExten(????);
originateAction.setPriority(new Integer(1));
originateAction.setCallerId("murthy");
originateAction.setTimeout(new Integer(30000));
// connect to Asterisk and log in
managerConnection.login();
// send the originate action and wait for a maximum of
30 seconds for Asterisk
// to send a reply
originateResponse =
managerConnection.sendAction(originateAction, 30000);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxxxxxx => {
This looks like a dialplan pattern match exten but you do not have a
leading '_' to indicate
that it is a pattern so this exten will only match a literal "1619xxxxxxx".
Answer();
Playback(welcomesystole);
Read(digito1,,3);
Playback(diastole);
Read(digito2,,3);
|
Agi(agi://10.10.22.171:4573/hello.agi?systole=${digito1}&diastole=${digito2}<http://10.10.22.171:4573/hello.agi?systole=$%7bdigito1%7d&diastole=$%7bdigito2%7d><http://10.10.22.171:4573/hello.agi?systole=$%7bdigito1%7d&diastole=$%7bdigito2%7d>);
| Quote: | Hangup()
}
It is up to you where you want to send the originated call to in your
dialplan. Since you
appear to want to send it to an extension that is a pattern you need to
use a value that
the pattern will match such as 16190000000.
Richard
|
Hi Richard
Thank you for your suggestions. The responses received are:
[Aug 6 11:20:28] NOTICE[25977][C-0000001a]: chan_sip.c:23147
handle_response_invite: Failed to authenticate on INVITE to '"Vonage
User"
<sip:1619xxxxxxx@69.59.234.67<mailto:sip%3A1619xxxxxxx@69.59.234.67>>;tag=as0bf485e8'
| Quote: | Channel SIP/vonage202-00000019 was never answered.
|
I don't understand the "Channel SIP/vonage202-00000019 was never
answered".... your kind clarification is sought.
What do you think "Failed to authenticate" on the call you just
originated means?
Your call was rejected and thus the call was never answered. You have an
authentication problem. Vonage could not authenticate the call you
originated.
Richard
|
I use the same password for INBOUND and it works fine! Something amiss with Asterisk OUTBOUND
because I used the same password with X-Lite and X-Pro Vonage soft phones with successful calls.
Regards
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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| Back to top |
|
|
asterisk.org at sedwar...
Guest
|
| Posted: Thu Aug 06, 2015 2:38 pm Post subject: [asterisk-users] Asterisk uses "Anonymous", but wh |
|
|
On Thu, 6 Aug 2015, Murthy Gandikota wrote:
[trimming cruft nobody cares about anymore]
| Quote: | I use the same password for INBOUND and it works fine! Something amiss
with Asterisk OUTBOUND because I used the same password with X-Lite and
X-Pro Vonage soft phones with successful calls.
|
Would comparing an INVITE from X-Lite or X-Pro with the INVITE from
Asterisk yield any clues?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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