Re: [asterisk-users] Asterisk uses Anonymous, but why? [SOLVED]
It is so embarrassing, but the Voicent gateway was running as a service on the Windoze. As soon as I stopped it, the outbound calls were working via Vonage. I am so sorry for wasting the time of Richard and Steve. But I am also grateful for their suggestions like using callerid in the AMI, and analyzing the Wireshark dump, etc. Overall I am so impressed by Asterisk that I will continue using it and hopefully one day contributing to it. Thanking you and best regards murthy Date: Thu, 6 Aug 2015 13:50:04 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why? On Thu, 6 Aug 2015, Steve Edwards wrote: On Thu, 6 Aug 2015, Steve Edwards wrote: Would comparing an INVITE from X-Lite or X-Pro with the INVITE from Asterisk yield any clues? On Thu, 6 Aug 2015, Murthy Gandikota wrote: For Asterisk INVITE please view http://pastebin.com/v15vMax4 For X-Lite INVITE please view http://pastebin.com/rmHZKu3N Just a quick glance (because I'm not a SIP expert)... (Asterisk) Request-Line: INVITE sip:1732xxx@69.59.234.67:5061 SIP/2.0 (X-Lite) Request-Line: INVITE sip:732xxx@69.59.234.67 SIP/2.0 This seems relevant to me. Some ISPs want the country code and some don't. Also, the port difference (5060 vs 5061) strikes me as curious. Not a smoking gun, just curious. On the production systems I run, I run OpenSIPS on 5060 and Asterisk on 5061 so you may be talking to 2 different endpoints. This also seems relevant: (Asterisk) Proxy-Authorization:... Authentication URI: sip:69.59.234.67 (X-Lite) Proxy-Authorization:... Authentication URI: sip:732xxx@69.59.234.67 -- Thanks in advance, - Steve Edwards sedwa...@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 -- _ -- 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
Re: [asterisk-users] PTT push to talk solution
Hi Jerry As others have eluded to, the 'PTT' feature can mean different things to different people depending on their background. Is it fair to say that you're looking for a one-touch button which initiates a call to the other end and causes the other end to automatically answer in speakerphone mode? If that would foot the bill then have a look at the auto-answer feature of Yealink and Snom phones (and others I'm sure). It can be easily triggered by adding a SIP header line into your dialplan so that the necessary header is included with the invite (this triggers the auto-answer at the remote end). If you have either of these brands to play with and need the dialplan code just sing out. Pete Mundy On 7/08/2015, at 3:09 AM, Jerry Geis ge...@pagestation.com wrote: I am looking for a push to talk solution does anyone know of a good PTT phone one that works with asterisk. I'm not talking about polycom fake PPT... I'm talking about a real call into Asterisk and having to push a button on a headset or the phone to actually talk. not multicast talk like polycom. I wish polycom had a real PTT headset but I cannot find one, I like their phones. Cisco has a PTT headset but seems only for 7960 model. Those phones are older and diffucult time find a new one and hard to get SIP on 7960. So is there a PTT phone out there that works great with asterisk ? Thanks so much. Jerry smime.p7s Description: S/MIME cryptographic signature -- _ -- 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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, 6 Aug 2015, Steve Edwards wrote: Would comparing an INVITE from X-Lite or X-Pro with the INVITE from Asterisk yield any clues? On Thu, 6 Aug 2015, Murthy Gandikota wrote: For Asterisk INVITE please view http://pastebin.com/v15vMax4 For X-Lite INVITE please view http://pastebin.com/rmHZKu3N Just a quick glance (because I'm not a SIP expert)... (Asterisk) Request-Line: INVITE sip:1732xxx@69.59.234.67:5061 SIP/2.0 (X-Lite) Request-Line: INVITE sip:732xxx@69.59.234.67 SIP/2.0 This seems relevant to me. Some ISPs want the country code and some don't. Also, the port difference (5060 vs 5061) strikes me as curious. Not a smoking gun, just curious. On the production systems I run, I run OpenSIPS on 5060 and Asterisk on 5061 so you may be talking to 2 different endpoints. -- Thanks in advance, - Steve Edwards sedwa...@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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
Date: Thu, 6 Aug 2015 12:37:36 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why? On Thu, 6 Aug 2015, Murthy Gandikota wrote: [trimming cruft nobody cares about anymore] 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? At the moment they are very similar. Now Vonage sends back 403 status (Forbidden) for the Request: INVITE sip:1732xxx@69.59.234.67:5061 The very similar request sent from X-Lite or X-Pro results in the message Trying... and it eventually succeeds. For Asterisk INVITE please view http://pastebin.com/v15vMax4 For X-Lite INVITE please view http://pastebin.com/rmHZKu3N 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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, 6 Aug 2015, Murthy Gandikota wrote: [trimming cruft nobody cares about anymore] 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 sedwa...@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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, 6 Aug 2015, Steve Edwards wrote: On Thu, 6 Aug 2015, Steve Edwards wrote: Would comparing an INVITE from X-Lite or X-Pro with the INVITE from Asterisk yield any clues? On Thu, 6 Aug 2015, Murthy Gandikota wrote: For Asterisk INVITE please view http://pastebin.com/v15vMax4 For X-Lite INVITE please view http://pastebin.com/rmHZKu3N Just a quick glance (because I'm not a SIP expert)... (Asterisk) Request-Line: INVITE sip:1732xxx@69.59.234.67:5061 SIP/2.0 (X-Lite) Request-Line: INVITE sip:732xxx@69.59.234.67 SIP/2.0 This seems relevant to me. Some ISPs want the country code and some don't. Also, the port difference (5060 vs 5061) strikes me as curious. Not a smoking gun, just curious. On the production systems I run, I run OpenSIPS on 5060 and Asterisk on 5061 so you may be talking to 2 different endpoints. This also seems relevant: (Asterisk) Proxy-Authorization:... Authentication URI: sip:69.59.234.67 (X-Lite) Proxy-Authorization:... Authentication URI: sip:732xxx@69.59.234.67 -- Thanks in advance, - Steve Edwards sedwa...@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
Re: [asterisk-users] PTT push to talk solution
This is for a government type end user. They wish to be having an internal meeting and be able to announce something - but require a push to talk button to speak. Thus the meeting can continue with the button released, then they can pause the meeting and push the button and speak more... Something like that is my understanding. Currently I have one of the new Ubiquity phones on my desk. There handsets have a mute button, or if you want a speak button, but a phone running under Android for government usage might leave some questions unanswered. If your phones have some functions keys, you'd have a look at the MuteAudio function and map the states to DTMF sequences, which in turn are mapped to the function keys. This makes you rather independent from any hardware and you might adapt the behavior depending on what your clients wishes will finally be, if they ever find out themselves. What I don't understand is why the normal mute button on most headsets is not sufficient. jg -- _ -- 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
Re: [asterisk-users] PTT push to talk solution
Hi Jerry As others have eluded to, the 'PTT' feature can mean different things to different people depending on their background. Is it fair to say that you're looking for a one-touch button which initiates a call to the other end and causes the other end to automatically answer in speakerphone mode? If that would foot the bill then have a look at the auto-answer feature of Yealink and Snom phones (and others I'm sure). It can be easily triggered by adding a SIP header line into your dialplan so that the necessary header is included with the invite (this triggers the auto-answer at the remote end). If you have either of these brands to play with and need the dialplan code just sing out. Pete Mundy Hi All, no - sorry if I wasn't clear. I am looking for a solution that I can pick up a headset, dial a number into asterisk and connect to whatever - BUT before I can actually speak I need to press a real button on the headset. if I then release the button on the headset no audio is passed but the call is still active. To speak again - I need to press the button on the headset. Like I mentioned. The fake PTT on polycom will not work. This is for a government type end user. They wish to be having an internal meeting and be able to announce something - but require a push to talk button to speak. Thus the meeting can continue with the button released, then they can pause the meeting and push the button and speak more... Something like that is my understanding. I hope that helps, Jerry -- _ -- 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
Re: [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: murth...@hotmail.com To: asterisk-users@lists.digium.com Subject: Asterisk uses Anonymous, but why? Date: Wed, 5 Aug 2015 21:38:16 + Hi All I am trying to dial out using SIP and Vonage using the instructions : a href=http#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage target=_blank class=newlyinsertedlinkhttp#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;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-@vonage-out application dial == Using SIP RTP CoS mark 5 [Aug 5 14:16:49] WARNING[32891][C-0006]: 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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 murth...@hotmail.commailto:murth...@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: murth...@hotmail.commailto:murth...@hotmail.com To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: Asterisk uses Anonymous, but why? Date: Wed, 5 Aug 2015 21:38:16 + 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=newlyinsertedlinkhttp://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@69.59.234.67:5060/202http://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-@vonage-out application dial == Using SIP RTP CoS mark 5 [Aug 5 14:16:49] WARNING[32891][C-0006]: chan_sip.c:23160 handle_response_invite: Received response: Forbidden from 'Anonymous sip:didsip:%3cdid@69.59.234.67http://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)=765637) -- _ -- 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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota murth...@hotmail.com
wrote:
Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudg...@digium.com
To: asterisk-users@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-@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-0006]: chan_sip.c:23160
handle_response_invite: Received response: Forbidden from
'Anonymous
sip:did@69.59.234.67http://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(3));
// 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, 3);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxx = {
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 1619xxx.
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 1619000.
Richard
--
_
-- 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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
From: murth...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Thu, 6 Aug 2015 17:33:37 +
Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why?
Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudg...@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
murth...@hotmail.commailto:murth...@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: murth...@hotmail.commailto:murth...@hotmail.com
To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Subject: Asterisk uses Anonymous, but why?
Date: Wed, 5 Aug 2015 21:38:16 +
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=newlyinsertedlinkhttp://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@69.59.234.67:5060/202http://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-@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-0006]: chan_sip.c:23160
handle_response_invite: Received response: Forbidden from
'Anonymous
sip:did@69.59.234.67http://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(3));
// 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, 3);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxx = {
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(3));
// 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, 3);
// 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);
Re: [asterisk-users] Asterisk uses Anonymous, but why?
Date: Thu, 6 Aug 2015 12:55:28 -0500
From: rmudg...@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
murth...@hotmail.commailto:murth...@hotmail.com wrote:
Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudg...@digium.commailto:rmudg...@digium.com
To: asterisk-users@lists.digium.commailto:asterisk-users@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-@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-0006]: chan_sip.c:23160
handle_response_invite: Received response: Forbidden from
'Anonymous
sip:did@69.59.234.67http://69.59.234.67http://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(3));
// 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, 3);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxx = {
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 1619xxx.
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=$%7bdigito1%7ddiastole=$%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 1619000.
Richard
Hi Richard
Thank you for your suggestions. The responses received are:
[Aug 6 11:20:28] NOTICE[25977][C-001a]: chan_sip.c:23147
handle_response_invite: Failed to authenticate on INVITE to 'Vonage User
sip:1619xxx@69.59.234.67;tag=as0bf485e8'
Channel SIP/vonage202-0019 was never answered.
I don't understand the Channel SIP/vonage202-0019 was never answered
your kind clarification is sought.
Regards
--
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Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, 6 Aug 2015, Murthy Gandikota wrote: [trimming cruft irrelvant to the current issue] [Aug 6 11:20:28] NOTICE[25977][C-001a]: chan_sip.c:23147 handle_response_invite: Failed to authenticate on INVITE to 'Vonage User sip:1619xxx@69.59.234.67;tag=as0bf485e8' Channel SIP/vonage202-0019 was never answered. I don't understand the Channel SIP/vonage202-0019 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 sedwa...@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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
Date: Thu, 6 Aug 2015 13:33:11 -0500
From: rmudg...@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
murth...@hotmail.commailto:murth...@hotmail.com wrote:
Date: Thu, 6 Aug 2015 12:55:28 -0500
From: rmudg...@digium.commailto:rmudg...@digium.com
To: asterisk-users@lists.digium.commailto: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
murth...@hotmail.commailto:murth...@hotmail.commailto:murth...@hotmail.commailto:murth...@hotmail.com
wrote:
Date: Thu, 6 Aug 2015 12:07:35 -0500
From:
rmudg...@digium.commailto:rmudg...@digium.commailto:rmudg...@digium.commailto:rmudg...@digium.com
To:
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.commailto:asterisk-users@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-@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-0006]: chan_sip.c:23160
handle_response_invite: Received response: Forbidden from
'Anonymous
sip:did@69.59.234.67http://69.59.234.67http://69.59.234.67http://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(3));
// 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, 3);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxx = {
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 1619xxx.
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=$%7bdigito1%7ddiastole=$%7bdigito2%7dhttp://10.10.22.171:4573/hello.agi?systole=$%7bdigito1%7ddiastole=$%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 1619000.
Richard
Hi Richard
Thank you for your suggestions. The responses received are:
[Aug 6 11:20:28] NOTICE[25977][C-001a]: chan_sip.c:23147
handle_response_invite: Failed to authenticate on INVITE to 'Vonage
User
sip:1619xxx@69.59.234.67mailto:sip%3A1619xxx@69.59.234.67;tag=as0bf485e8'
Channel SIP/vonage202-0019 was never answered.
I don't understand the Channel SIP/vonage202-0019 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
--
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Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota murth...@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: murth...@hotmail.com To: asterisk-users@lists.digium.com Subject: Asterisk uses Anonymous, but why? Date: Wed, 5 Aug 2015 21:38:16 + 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-@vonage-out application dial == Using SIP RTP CoS mark 5 [Aug 5 14:16:49] WARNING[32891][C-0006]: 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 -- _ -- 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
Re: [asterisk-users] Asterisk uses Anonymous, but why?
Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudg...@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
murth...@hotmail.commailto:murth...@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: murth...@hotmail.commailto:murth...@hotmail.com
To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Subject: Asterisk uses Anonymous, but why?
Date: Wed, 5 Aug 2015 21:38:16 +
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=newlyinsertedlinkhttp://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@69.59.234.67:5060/202http://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-@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-0006]: chan_sip.c:23160
handle_response_invite: Received response: Forbidden from
'Anonymous
sip:did@69.59.234.67http://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(3));
// 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, 3);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxx = {
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
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Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota murth...@hotmail.com
wrote:
Date: Thu, 6 Aug 2015 12:55:28 -0500
From: rmudg...@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
murth...@hotmail.commailto:murth...@hotmail.com wrote:
Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudg...@digium.commailto:rmudg...@digium.com
To: asterisk-users@lists.digium.commailto:
asterisk-users@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-@vonage-out application dial
== Using SIP RTP CoS mark 5
[Aug 5 14:16:49] WARNING[32891][C-0006]: chan_sip.c:23160
handle_response_invite: Received response: Forbidden from
'Anonymous
sip:did@69.59.234.67http://69.59.234.67http://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(3));
// 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, 3);
I get error with this.
Here is from-pstn context in extensions.ael
context from-pstn {
1619xxx = {
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
1619xxx.
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=$%7bdigito1%7ddiastole=$%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 1619000.
Richard
Hi Richard
Thank you for your suggestions. The responses received are:
[Aug 6 11:20:28] NOTICE[25977][C-001a]: chan_sip.c:23147
handle_response_invite: Failed to authenticate on INVITE to 'Vonage User
sip:1619xxx@69.59.234.67;tag=as0bf485e8'
Channel SIP/vonage202-0019 was never answered.
I don't understand the Channel SIP/vonage202-0019 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
--
_
-- 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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Re: [asterisk-users] asterisk queue - skills based routing (patch updated)
Hello, Le 2015-08-06 09:24, Marek Cervenka a écrit : hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 You can find the latest version we maintain here : https://github.com/xivo-pbx/asterisk/blob/master/debian/patches/xivo_skill_queues (asterisk 13.5) We originally wrote this patch for xivo and it's included by default. Sylvain -- _ -- 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
[asterisk-users] asterisk queue - skills based routing (patch updated)
hi, there is updated skills based routing patch for asterisk queue please test if you have time https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22 -- --- Marek Cervenka === -- _ -- 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
[asterisk-users] PTT push to talk solution
I am looking for a push to talk solution does anyone know of a good PTT phone one that works with asterisk. I'm not talking about polycom fake PPT... I'm talking about a real call into Asterisk and having to push a button on a headset or the phone to actually talk. not multicast talk like polycom. I wish polycom had a real PTT headset but I cannot find one, I like their phones. Cisco has a PTT headset but seems only for 7960 model. Those phones are older and diffucult time find a new one and hard to get SIP on 7960. So is there a PTT phone out there that works great with asterisk ? Thanks so much. Jerry -- _ -- 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
Re: [asterisk-users] PTT push to talk solution
I am looking for a push to talk solution does anyone know of a good PTT phone one that works with asterisk. I'm not talking about polycom fake PPT... I'm talking about a real call into Asterisk and having to push a button on a headset or the phone to actually talk. not multicast talk like polycom. I wish polycom had a real PTT headset but I cannot find one, I like their phones. Cisco has a PTT headset but seems only for 7960 model. Those phones are older and diffucult time find a new one and hard to get SIP on 7960. So is there a PTT phone out there that works great with asterisk ? I am not sure whether I really understood your question. It looks to me that the PTT functionality can easily be achieved using the mute button that most phones and headsets have. One could even implement it independent of any specific phone using the Asterisk function MuteAudio(). One could use the DTMF features for signaling. jg -- _ -- 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
Re: [asterisk-users] PTT push to talk solution
On Thursday 06 Aug 2015, Jerry Geis wrote: I am looking for a push to talk solution does anyone know of a good PTT phone one that works with asterisk. Um . Asterisk supports full-duplex telephony, so there's no need for any of that over to you, roger and out business -- you can actually talk in both directions at once (assuming your voices are pitched far enough apart for each of you to be able to understand what the other is saying). You could try wiring the normally-closed contact of a change-over switch across the microphone in the handset of any compatible phone . that should silence it until the switch is pressed. -- AJS The best programming language is 63% tin, 37% lead. -- _ -- 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
