[asterisk-users] VPN SIP Phone | PC Traffic
We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are connected to the mikrotik... user plugs router in, plugs phone and computer into router, and that traffic is encapsulated back to our office... simple and straighforward. We would like to remove the router from the equation... does anyone know of a SIP phone with a built in VPN client that can provide the tunnel for *both the phone and the pc traffic*? It would seem trivial to route a subnet down to the vpn client in the phone, that would be available to devices connected on the PC side of the telephone.. This would be tremendous for an at-home contact center agent..An added benefit would be to limit connections the connection on the PC side of the phone to a specific mac address.. We're aware of the opportunity to use a softphone on the pc with a vpn client. though, we're looking for a physical phone. -- _ -- 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] VPN SIP Phone | PC Traffic
On 9 April 2014 11:42, Positively Optimistic positivelyoptimis...@gmail.com wrote: We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are connected to the mikrotik... user plugs router in, plugs phone and computer into router, and that traffic is encapsulated back to our office... simple and straighforward. We would like to remove the router from the equation... does anyone know of a SIP phone with a built in VPN client that can provide the tunnel for *both the phone and the pc traffic*? It would seem trivial to route a subnet down to the vpn client in the phone, that would be available to devices connected on the PC side of the telephone.. This would be tremendous for an at-home contact center agent..An added benefit would be to limit connections the connection on the PC side of the phone to a specific mac address.. We're aware of the opportunity to use a softphone on the pc with a vpn client. though, we're looking for a physical phone. Yealink seems to support OpenVPN for the phone, although I am not sure if it extends to the PC port or not. Nice feature I agree. http://www.yealink.com/Upload/T2X/20131125/OpenVPN_Feature_on_Yealink_IP_Phones.pdf Hope this helps. -Barry Flanagan -- _ -- 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] VPN SIP Phone | PC Traffic
On 9/04/2014, at 10:42 pm, Positively Optimistic positivelyoptimis...@gmail.com wrote: We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are connected to the mikrotik... user plugs router in, plugs phone and computer into router, and that traffic is encapsulated back to our office... simple and straighforward. We would like to remove the router from the equation... does anyone know of a SIP phone with a built in VPN client that can provide the tunnel for both the phone and the pc traffic? It would seem trivial to route a subnet down to the vpn client in the phone, that would be available to devices connected on the PC side of the telephone.. This would be tremendous for an at-home contact center agent..An added benefit would be to limit connections the connection on the PC side of the phone to a specific mac address.. We use the Yealink phones - a number of which do openvpn, we had Mikrotiks in the middle but are removing them now, we haven’t tested the PC routing side though. http://www.yealink.com/Upload/T4X/GA/OpenVPN_Feature_on_Yealink_IP_Phones(Linux_Windows)_V71.pdf We're aware of the opportunity to use a softphone on the pc with a vpn client. though, we're looking for a physical phone. -- _ -- 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] is g729 codec free? or under license???
On Wed, Apr 09, 2014 at 10:19:59AM +0800, Steve Underwood wrote: Hi Jeff, On 04/08/2014 12:13 PM, Jeff Brower wrote: Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff Where did you get the cutoff date of June 8 1995, and how does 20 years from that date lead to the last of the patents expiring in 2014? Nobody uses G.729. They use G.729A. The G.729A spec is somewhat later than the original G.729, but I don't know if there are any additional patents which specifically relate to Annex A. You could use G.729 instead, but it roughly doubles the compute needed. If it allows me to avoid the trolls: I'll pay that performance hit. In many caces there are CPU cycles to spare. But the licensing is a hard limit. There are various things on the web saying the last of the patents on G.723.1, which was around in draft form long before G.729, expires in 2014. However, there seem to be patents related to that codec which don't really expire until some time in 2015. Its really hard to find solid information. The ITU patent database rarely identifies the actual patents being claimed, so its damned hard to look them up. Nice. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- 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] is g729 codec free? or under license???
On 04/09/2014 06:54 PM, Tzafrir Cohen wrote: On Wed, Apr 09, 2014 at 10:19:59AM +0800, Steve Underwood wrote: Hi Jeff, On 04/08/2014 12:13 PM, Jeff Brower wrote: Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff Where did you get the cutoff date of June 8 1995, and how does 20 years from that date lead to the last of the patents expiring in 2014? Nobody uses G.729. They use G.729A. The G.729A spec is somewhat later than the original G.729, but I don't know if there are any additional patents which specifically relate to Annex A. You could use G.729 instead, but it roughly doubles the compute needed. If it allows me to avoid the trolls: I'll pay that performance hit. In many caces there are CPU cycles to spare. But the licensing is a hard limit. Well, you do get the benefit of higher quality for your extra compute. G.729 sounds distinctly better than G.729A on a lot of material. There are various things on the web saying the last of the patents on G.723.1, which was around in draft form long before G.729, expires in 2014. However, there seem to be patents related to that codec which don't really expire until some time in 2015. Its really hard to find solid information. The ITU patent database rarely identifies the actual patents being claimed, so its damned hard to look them up. Nice. Regards, Steve -- _ -- 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] VPN SIP Phone | PC Traffic
I did this with SNOM phones and a special firmware a while ago. The trick to get the VPN to extend to the PC port is bridge-utils. Worked very well. On Apr 9, 2014 7:40 AM, Positively Optimistic positivelyoptimis...@gmail.com wrote: We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are connected to the mikrotik... user plugs router in, plugs phone and computer into router, and that traffic is encapsulated back to our office... simple and straighforward. We would like to remove the router from the equation... does anyone know of a SIP phone with a built in VPN client that can provide the tunnel for *both the phone and the pc traffic*? It would seem trivial to route a subnet down to the vpn client in the phone, that would be available to devices connected on the PC side of the telephone.. This would be tremendous for an at-home contact center agent..An added benefit would be to limit connections the connection on the PC side of the phone to a specific mac address.. We're aware of the opportunity to use a softphone on the pc with a vpn client. though, we're looking for a physical phone. -- _ -- 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] PJSIP in dialog OPTIONS method handling
Hi, Anyone has a workaround? On Tue, Apr 8, 2014 at 9:17 PM, Yaron Nachum nachum.ya...@gmail.com wrote: Hi everyone, I am running asterisk with release 12.1.0.rc3 and PJSIP. I have a peer which sends OPTIONS method for session keep-alive, and the asterisk is not responding to it. That of course disconnects the call after a few minutes. Is there a settings in the PJSIP.conf to respond to in dialog OPTIONS method? Looking at the documentation I haven't seen it. Does anybody know a workaround? Thanks, Yaron. -- _ -- 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] Google Puts the Final Nail in the Google Voice Coffin
No tell me that's a jock ! I can't believe it: http://nerdvittles.com/?p=7940 -- _ -- 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] PJSIP usereqphone setting in config file
Hi everyone, I am starting to work with PJSIP on release 12.1.0.rc3. I used to have Asterisk 1.8 with the regular sip channel. I was using the usereqphone settings in order to set user=phone on from and to URIs. Is there a similar config in PJSIP? -- _ -- 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] I can't make outbound calls (status is 'CHANUNAVAIL')
Hello:
I have this situation: I can make calls internally, I can make inbound
calls but I can't make outbound calls.
Thanks in advance.
These are my devices:
* asterisk 11.8.1 = 192.168.1.22
* sipphone grandstream gxp2160 = 192.168.1.5
* gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4
port 1 (FXS) connected to an analog phone
port 3 (FXO) connected to the PSTN
These are my sip.conf and extensions.conf files:
sip.conf
[general]
context = incoming-call
allowguest = no
srvlookup = no
udpbindaddr = 0.0.0.0
tcpenable = no
qualify = yes
language = es
[office](!)
type = friend
context = internal-call
host = dynamic
nat = force_rport,comedia
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
[telefono](office)
description = grandstream gxp2160
secret = telefono
[celular](office)
description = samsung gt-s7562
secret = celular
[fxs](office)
description = fxs port1
secret = fxs
[pstn](!)
nat = no
canreinvite = no
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
[pstn-in](pstn)
description = pstn-in port3
type = user
host = dynamic
secret = pstn-in
context = incoming-call
[pstn-out](pstn)
description = pstn-out port3
type = peer
host = 192.168.1.4
extensions.conf
---
[incoming-call]
exten = _24872006,1,Answer()
same = n,Dial(SIP/telefono)
same = n,Hangup()
[outgoing-call]
exten = _X.,1,Dial(SIP/${EXTEN}@pstn-out)
[internal-call]
exten = 101,1,Dial(SIP/telefono)
exten = 102,1,Dial(SIP/celular)
exten = 103,1,Dial(SIP/fxs)
exten = 104,1,Answer()
same = n,Playback(tt-weasels)
same = n,Hangup()
include = outgoing-call
This is the result of sip show peers
--
Name/usernameHost Dyn Forcerport ComediaACL Port
Status Description
celular/celular 192.168.1.21D YesYes
47747OK (6 ms) samsung gt-s7562
fxs/fxs 192.168.1.4 D YesYes5060
OK (27 ms) fxs port1
pstn-out 192.168.1.4No No 5060
OK (25 ms) pstn-out port3
telefono/telefono192.168.1.5 D YesYes1555
OK (3 ms) grandstream gxp2160
4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0 offline]
This is the result of sip show users
--
UsernameSecret Accountcode Def.Context ACL Forcerport
celular celular internal-callNo Yes
pstn-in pstn-in incoming-callNo No
fxs fxs internal-callNo Yes
telefonotelefono internal-callNo Yes
debian-asterisk*CLI
This is the result of sip set debug on when I try to make an outbound call:
--- SIP read from UDP:192.168.1.5:1555 ---
INVITE sip:@192.168.1.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport
From: sip:telefono@192.168.1.22;tag=1524540678
To: sip:@192.168.1.22
Call-ID: 667168938-155...@bjc.bgi.B.F
CSeq: 30 INVITE
Contact: sip:telefono@192.168.1.5:1555
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.0.17
Privacy: none
P-Preferred-Identity: sip:telefono@192.168.1.22
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 335
v=0
o=telefono 8000 8000 IN IP4 192.168.1.5
s=SIP Call
c=IN IP4 192.168.1.5
t=0 0
m=audio 5004 RTP/AVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (17 headers 16 lines) ---
Sending to 192.168.1.5:1555 (no NAT)
Sending to 192.168.1.5:1555 (no NAT)
Using INVITE request as basis request - 667168938-155...@bjc.bgi.B.F
Found peer 'telefono' for 'telefono' from 192.168.1.5:1555
--- Reliably Transmitting (NAT) to 192.168.1.5:1555 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.5:1555;branch=z9hG4bK2009427179;received=192.168.1.5;rport=1555
From: sip:telefono@192.168.1.22;tag=1524540678
To: sip:@192.168.1.22;tag=as50d1512e
Call-ID: 667168938-155...@bjc.bgi.B.F
CSeq: 30 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1032f9e6
Content-Length: 0
Scheduling destruction of SIP dialog '667168938-155...@bjc.bgi.B.F' in
6400 ms (Method: INVITE)
--- SIP read from UDP:192.168.1.5:1555 ---
ACK sip:@192.168.1.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport
From: sip:telefono@192.168.1.22;tag=1524540678
To:
[asterisk-users] automated response
I will not be in the office from April 10 to 14. I will get back to you when I'm back. Joseph -- _ -- 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] I can't make outbound calls (status is 'CHANUNAVAIL')
Check your trunk @pstn-out there's something reaching that server
192.168.1.4?
2014-04-09 12:06 GMT-05:00 Luis Eduardo Cortes luedcor...@gmail.com:
Hello:
I have this situation: I can make calls internally, I can make inbound
calls but I can't make outbound calls.
Thanks in advance.
These are my devices:
* asterisk 11.8.1 = 192.168.1.22
* sipphone grandstream gxp2160 = 192.168.1.5
* gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4
port 1 (FXS) connected to an analog phone
port 3 (FXO) connected to the PSTN
These are my sip.conf and extensions.conf files:
sip.conf
[general]
context = incoming-call
allowguest = no
srvlookup = no
udpbindaddr = 0.0.0.0
tcpenable = no
qualify = yes
language = es
[office](!)
type = friend
context = internal-call
host = dynamic
nat = force_rport,comedia
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
[telefono](office)
description = grandstream gxp2160
secret = telefono
[celular](office)
description = samsung gt-s7562
secret = celular
[fxs](office)
description = fxs port1
secret = fxs
[pstn](!)
nat = no
canreinvite = no
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
[pstn-in](pstn)
description = pstn-in port3
type = user
host = dynamic
secret = pstn-in
context = incoming-call
[pstn-out](pstn)
description = pstn-out port3
type = peer
host = 192.168.1.4
extensions.conf
---
[incoming-call]
exten = _24872006,1,Answer()
same = n,Dial(SIP/telefono)
same = n,Hangup()
[outgoing-call]
exten = _X.,1,Dial(SIP/${EXTEN}@pstn-out)
[internal-call]
exten = 101,1,Dial(SIP/telefono)
exten = 102,1,Dial(SIP/celular)
exten = 103,1,Dial(SIP/fxs)
exten = 104,1,Answer()
same = n,Playback(tt-weasels)
same = n,Hangup()
include = outgoing-call
This is the result of sip show peers
--
Name/usernameHost Dyn Forcerport ComediaACL Port
Status Description
celular/celular 192.168.1.21D YesYes
47747OK (6 ms) samsung gt-s7562
fxs/fxs 192.168.1.4 D YesYes5060
OK (27 ms) fxs port1
pstn-out 192.168.1.4No No 5060
OK (25 ms) pstn-out port3
telefono/telefono192.168.1.5 D YesYes1555
OK (3 ms) grandstream gxp2160
4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0
offline]
This is the result of sip show users
--
UsernameSecret Accountcode Def.Context ACL Forcerport
celular celular internal-callNo Yes
pstn-in pstn-in incoming-callNo No
fxs fxs internal-callNo Yes
telefonotelefono internal-callNo Yes
debian-asterisk*CLI
This is the result of sip set debug on when I try to make an outbound
call:
--- SIP read from UDP:192.168.1.5:1555 ---
INVITE sip:@192.168.1.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport
From: sip:telefono@192.168.1.22;tag=1524540678
To: sip:@192.168.1.22
Call-ID: 667168938-155...@bjc.bgi.B.F
CSeq: 30 INVITE
Contact: sip:telefono@192.168.1.5:1555
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.0.17
Privacy: none
P-Preferred-Identity: sip:telefono@192.168.1.22
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 335
v=0
o=telefono 8000 8000 IN IP4 192.168.1.5
s=SIP Call
c=IN IP4 192.168.1.5
t=0 0
m=audio 5004 RTP/AVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (17 headers 16 lines) ---
Sending to 192.168.1.5:1555 (no NAT)
Sending to 192.168.1.5:1555 (no NAT)
Using INVITE request as basis request - 667168938-155...@bjc.bgi.B.F
Found peer 'telefono' for 'telefono' from 192.168.1.5:1555
--- Reliably Transmitting (NAT) to 192.168.1.5:1555 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.5:1555;branch=z9hG4bK2009427179;received=192.168.1.5;rport=1555
From: sip:telefono@192.168.1.22;tag=1524540678
To: sip:@192.168.1.22;tag=as50d1512e
Call-ID: 667168938-155...@bjc.bgi.B.F
CSeq: 30 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1032f9e6
Content-Length: 0
Scheduling
Re: [asterisk-users] I can't make outbound calls (status is 'CHANUNAVAIL')
Check your trunk @pstn-out there's something reaching that server
192.168.1.4?
2014-04-09 12:06 GMT-05:00 Luis Eduardo Cortes luedcor...@gmail.com:
Hello:
I have this situation: I can make calls internally, I can make inbound
calls but I can't make outbound calls.
Thanks in advance.
These are my devices:
* asterisk 11.8.1 = 192.168.1.22
* sipphone grandstream gxp2160 = 192.168.1.5
* gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4
port 1 (FXS) connected to an analog phone
port 3 (FXO) connected to the PSTN
These are my sip.conf and extensions.conf files:
sip.conf
[general]
context = incoming-call
allowguest = no
srvlookup = no
udpbindaddr = 0.0.0.0
tcpenable = no
qualify = yes
language = es
[office](!)
type = friend
context = internal-call
host = dynamic
nat = force_rport,comedia
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
[telefono](office)
description = grandstream gxp2160
secret = telefono
[celular](office)
description = samsung gt-s7562
secret = celular
[fxs](office)
description = fxs port1
secret = fxs
[pstn](!)
nat = no
canreinvite = no
dtmfmode = auto
disallow = all
allow = g722
allow = alaw
allow = ulaw
[pstn-in](pstn)
description = pstn-in port3
type = user
host = dynamic
secret = pstn-in
context = incoming-call
[pstn-out](pstn)
description = pstn-out port3
type = peer
host = 192.168.1.4
extensions.conf
---
[incoming-call]
exten = _24872006,1,Answer()
same = n,Dial(SIP/telefono)
same = n,Hangup()
[outgoing-call]
exten = _X.,1,Dial(SIP/${EXTEN}@pstn-out)
[internal-call]
exten = 101,1,Dial(SIP/telefono)
exten = 102,1,Dial(SIP/celular)
exten = 103,1,Dial(SIP/fxs)
exten = 104,1,Answer()
same = n,Playback(tt-weasels)
same = n,Hangup()
include = outgoing-call
This is the result of sip show peers
--
Name/usernameHost Dyn Forcerport ComediaACL Port
Status Description
celular/celular 192.168.1.21D YesYes
47747OK (6 ms) samsung gt-s7562
fxs/fxs 192.168.1.4 D YesYes5060
OK (27 ms) fxs port1
pstn-out 192.168.1.4No No 5060
OK (25 ms) pstn-out port3
telefono/telefono192.168.1.5 D YesYes1555
OK (3 ms) grandstream gxp2160
4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0
offline]
This is the result of sip show users
--
UsernameSecret Accountcode Def.Context ACL Forcerport
celular celular internal-callNo Yes
pstn-in pstn-in incoming-callNo No
fxs fxs internal-callNo Yes
telefonotelefono internal-callNo Yes
debian-asterisk*CLI
This is the result of sip set debug on when I try to make an outbound
call:
--- SIP read from UDP:192.168.1.5:1555 ---
INVITE sip:@192.168.1.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:1555;branch=z9hG4bK2009427179;rport
From: sip:telefono@192.168.1.22;tag=1524540678
To: sip:@192.168.1.22
Call-ID: 667168938-155...@bjc.bgi.B.F
CSeq: 30 INVITE
Contact: sip:telefono@192.168.1.5:1555
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.0.17
Privacy: none
P-Preferred-Identity: sip:telefono@192.168.1.22
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO,
REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 335
v=0
o=telefono 8000 8000 IN IP4 192.168.1.5
s=SIP Call
c=IN IP4 192.168.1.5
t=0 0
m=audio 5004 RTP/AVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (17 headers 16 lines) ---
Sending to 192.168.1.5:1555 (no NAT)
Sending to 192.168.1.5:1555 (no NAT)
Using INVITE request as basis request - 667168938-155...@bjc.bgi.B.F
Found peer 'telefono' for 'telefono' from 192.168.1.5:1555
--- Reliably Transmitting (NAT) to 192.168.1.5:1555 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.5:1555;branch=z9hG4bK2009427179;received=192.168.1.5;rport=1555
From: sip:telefono@192.168.1.22;tag=1524540678
To: sip:@192.168.1.22;tag=as50d1512e
Call-ID: 667168938-155...@bjc.bgi.B.F
CSeq: 30 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1032f9e6
Content-Length: 0
Scheduling
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Dear friends, In Asterisk 11.7.0, is it possible to receive CEL APP_START and APP_END events from Park application? Queue and Dial apps are generation these events, but Park no. It doesn't seem to make difference when configured. cel.conf ... apps = dial,queue,confbridge,park events = ALL ... localhost*CLI cel show status ... CEL Tracking Application: queue CEL Tracking Application: confbridge CEL Tracking Application: dial CEL Tracking Application: park ... Tks. -- _ -- 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
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