[asterisk-users] how do i enable call features??
Hi. i am running asterisk 11 and i would like to have features access codes
in my system such as call waiting(all types) (enable/disable), call forward
(enable/disable) and DND. my dialplan is pretty simple and it is the
following
[DefaultPlan]exten =>
_XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten =>
_XX,1,Busy()
exten => _4XX,2,Answer()exten =>
_4XX,3,VoiceMail(${EXTEN}@Office,su)exten => _4XX,4,HangUp()exten =>
_4XX,102,Answer()exten => _4XX,103,VoiceMail(${EXTEN}@Office,sb)exten
=> _4XX,104,HangUp()
i would like to enable/disable call waiting by typing eg. *70/*71
DND for the extension *72 enable, *73 to disable.
Regarding call waiting, at the moment it is disabled (default value). Now
if an extension is busy, a busy message is send back to the caller. I would
like have the following behavior:
-in the event were the extension is busy, then send a message indication to
the extension and the caller to hear from the SIP provider the default
early media for call waiting due to busy. Then after some period of time eg
30 secs send busy.
-in the event where the extension is busy, send the early media to the
caller and waiting indication to the extension. If the extension decides to
get the call then get the 2nd call and send the 1st to hear moh.
My phones are mainly softphones (zoiper), a few IP phones and 2 SPA3000 for
analog devices.
could someone help me please with this task, please?
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Re: [asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)
On Tue, Jan 9, 2018, at 10:01 AM, Benoit Panizzon wrote: > Hi Jushua > > > The rtp_ipv6 option is not needed, in current versions things will > > automatically be updated to reflect the signaling. Remove it and give > > it a try. The option itself actually had the bug that you are seeing. > > Ok, commented out rtp_ipv6 in the config and did try again: > > IPv6 Registered client. > > c=IN IP6 2001:4060:1:4133:204:13ff:fe30:2a80 > > Reply from * > > c=IN IP6 2001:4060:dead:beef::1 > > IPv4 registered client: > > c=IN IP4 157.161.4.172 > > Reply from * > > c=IN IP4 157.161.57.1 > > Perfect! It didn't occur to me to completely comment out that option as > I believed it was needed for rtp to work over ipv6. > > Thank you for that exceptional quick help. It used to be required but as part of Asterisk 14 work was done in DNS land (failover to different targets, including between IPv6 and IPv4) and based on discussions I had with other people at SIPit I made it automatic so that media family = signaling family. To keep things better in line and to provide a better experience the change was also done in Asterisk 13. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)
Hi Jushua > The rtp_ipv6 option is not needed, in current versions things will > automatically be updated to reflect the signaling. Remove it and give > it a try. The option itself actually had the bug that you are seeing. Ok, commented out rtp_ipv6 in the config and did try again: IPv6 Registered client. c=IN IP6 2001:4060:1:4133:204:13ff:fe30:2a80 Reply from * c=IN IP6 2001:4060:dead:beef::1 IPv4 registered client: c=IN IP4 157.161.4.172 Reply from * c=IN IP4 157.161.57.1 Perfect! It didn't occur to me to completely comment out that option as I believed it was needed for rtp to work over ipv6. Thank you for that exceptional quick help. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)
On Tue, Jan 9, 2018, at 9:48 AM, Benoit Panizzon wrote: > Dear List > > I fear I stumbled over a bug in asterisk 13.14.1. > > My 'phones' are roaming around, sometimes some are connecting from ipv6 > enabled networks, another time they are not. > > If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat > problems. > > I have not specified a transport in the endpoint section, so that the > appropriate transport which corresponds to the registration can be used. > > Now I have noticed, if an phone is registered from an ipv4 only > endpoint and is performing an outgoing call, my asterisk server is > answering with an IP4 RTP IPv6 address: The rtp_ipv6 option is not needed, in current versions things will automatically be updated to reflect the signaling. Remove it and give it a try. The option itself actually had the bug that you are seeing. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)
Dear List I fear I stumbled over a bug in asterisk 13.14.1. My 'phones' are roaming around, sometimes some are connecting from ipv6 enabled networks, another time they are not. If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat problems. I have not specified a transport in the endpoint section, so that the appropriate transport which corresponds to the registration can be used. Now I have noticed, if an phone is registered from an ipv4 only endpoint and is performing an outgoing call, my asterisk server is answering with an IP4 RTP IPv6 address: Example: <--- Received SIP request (1235 bytes) from UDP:157.161.4.172:5060 ---> [...] v=0 o=MxSIP 0 20 IN IP4 157.161.4.172 s=SIP Call c=IN IP4 157.161.4.172 t=0 0 m=audio 6018 RTP/AVP 9 8 97 91 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 CLEARMODE/8000 a=rtpmap:91 X-CLEAR-CHANNEL/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv <--- Transmitting SIP response (914 bytes) to UDP:157.161.4.172:5060 ---> [...] v=0 o=- 0 22 IN IP4 2001:4060:dead:beef::1 s=Asterisk c=IN IP4 2001:4060:dead:beef::1 t=0 0 m=audio 16172 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv Errr... how should my client @ 157.161.4.172 send udp to 2001:4060:dead:beef::1? Also when I compare with a real IPv6 client I notice from the client: c=IN IP6 2001:4060:1:4133:204:13ff:fe30:2a80 from the Asterisk: c=IN IP4 2001:4060:dead:beef::1 Shouldn't that also be IP6 from the asterisk? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP: identify endpoint by authentication username?
Hi George > [global] > endpoint_identifier_order = auth_username,username,ip,anonymous > > [endpoint_x] > identify_by = auth_username Thank you, I missed that config option, works perfectly! Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP: identify endpoint by authentication username?
On Tue, Jan 9, 2018 at 5:38 AM, Benoit Panizzon wrote: > Dear fellow list readers > > This is the situation: > > ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP > > The Patton GW resides on a dynamic IP address, so I cannot really use > match=ip in the identify section. > > The Patton does not send a line parameter. > > The ISDN Devices behind the patton have different MSN and should be > able to send them in the From: Header, so the default endpoint > identification mechanism which matches the From username with the > endplaint fails. > > So what are the options to solve that issue? > > I see the asterisk sending out a challenge and getting a proper reply > from the patton, but then stills complains about the endpoint not > matching. > > According to the manual there is no > > type=identify > match=authentication_username > There is no need for a separate "identify" object in this case. In the pjsip.conf "global" section set "endpoint_identifier_order" to include "auth_username" and in each endpoint's section set "identify_by" to include "auth_username". [global] endpoint_identifier_order = auth_username,username,ip,anonymous [endpoint_x] identify_by = auth_username > > or similar. > > Mit freundlichen Grüssen > > -Benoît Panizzon- > -- > I m p r o W a r e A G-Leiter Commerce Kunden > __ > > Zurlindenstrasse 29 Tel +41 61 826 93 00 > CH-4133 PrattelnFax +41 61 826 93 01 > Schweiz Web http://www.imp.ch > __ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP: identify endpoint by authentication username?
Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint identification mechanism which matches the From username with the endplaint fails. So what are the options to solve that issue? I see the asterisk sending out a challenge and getting a proper reply from the patton, but then stills complains about the endpoint not matching. According to the manual there is no type=identify match=authentication_username or similar. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
