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>
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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 at hotmail.com<mailto: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<mailto:murthy64 at hotmail.com> > To: asterisk-users at lists.digium.com<mailto: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<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 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<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) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150806/a7a479d4/attachment.html>
Murthy Gandikota
2015-Aug-06 17:33 UTC
[asterisk-users] Asterisk uses "Anonymous", but why?
________________________________> Date: Thu, 6 Aug 2015 12:07:35 -0500 > From: rmudgett at digium.com > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota > <murthy64 at hotmail.com<mailto: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<mailto:murthy64 at hotmail.com> >> To: asterisk-users at lists.digium.com<mailto: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<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 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<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. > > RichardHi 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() } ? ? ? ? ? ? ? ?
Murthy Gandikota
2015-Aug-06 17:54 UTC
[asterisk-users] Asterisk uses "Anonymous", but why?
----------------------------------------> From: murthy64 at hotmail.com > To: asterisk-users at lists.digium.com > Date: Thu, 6 Aug 2015 17:33:37 +0000 > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? > > > > ________________________________ >> Date: Thu, 6 Aug 2015 12:07:35 -0500 >> From: rmudgett at digium.com >> To: asterisk-users at lists.digium.com >> Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? >> >> >> >> On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota >> <murthy64 at hotmail.com<mailto: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<mailto:murthy64 at hotmail.com> >>> To: asterisk-users at lists.digium.com<mailto: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<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 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<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 > > > 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 at 69.59.234.67>;tag=as46f9ddef' ubuntu*CLI>? Regards
Richard Mudgett
2015-Aug-06 17:55 UTC
[asterisk-users] Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota <murthy64 at hotmail.com> wrote:> > > ________________________________ > > Date: Thu, 6 Aug 2015 12:07:35 -0500 > > From: rmudgett at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk uses "Anonymous", but why? ><snip>> >> 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<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 > > > 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}); > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150806/c99d5915/attachment.html>