From: [email protected] 
[mailto:[email protected]] 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 
<[email protected]<mailto:[email protected]>> 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

----------------------------------------
> From: [email protected]<mailto:[email protected]>
> To: [email protected]<mailto:[email protected]>
> 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<http://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<sip:%3cdid>>@69.59.234.67<http://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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