Murthy Gandikota
2015-Aug-05 21:38 UTC
[asterisk-users] Asterisk uses "Anonymous", but why?
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 at 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
Murthy Gandikota
2015-Aug-06 16:56 UTC
[asterisk-users] Asterisk uses "Anonymous", but why?
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: murthy64 at hotmail.com > To: asterisk-users at 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 at 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 > > >
Richard Mudgett
2015-Aug-06 17:07 UTC
[asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at 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 > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at 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 at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150806/af1a07a9/attachment.html>