[asterisk-users] Why all the 401 Unauthorized
I have a connection between two asterisk boxes, both running 1.4.43 The connection is alive and good and working. however, I see a bunch of 401 Unauthorized messages using wireshark - then it eventually registers again just fine. Why would it not successfully register the first time - every time? 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] Why all the 401 Unauthorized
Hi, SIP registrations typically try to register, are them prompted for a password (via a 401 message) it then sends a new request with authentication . This is normal. Steve On 23 Oct 2012, at 13:26, Jerry Geis wrote: I have a connection between two asterisk boxes, both running 1.4.43 The connection is alive and good and working. however, I see a bunch of 401 Unauthorized messages using wireshark - then it eventually registers again just fine. Why would it not successfully register the first time - every time? 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 -- _ -- 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] dahdi dummy
I need to use the dahdi dummy driver. Its not being compiled at this time. When I go into tools subdirectory under dahdi-linux-complete-2.4.1 and do make menuselect all I get is CC= CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent nmenuselect make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Nothing to be done for `nmenuselect'. make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' CC= CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent gmenuselect make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Nothing to be done for `gmenuselect'. make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools' Terminal must be at least 80 x 27. menuselect changes NOT saved! make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools' How can I get the dahdi_dummy.c driver compiled? 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] dahdi dummy
If I remember correctly, dahdi dummy was removed and the functionally added by default when you load dahdi with no TDM cards installed. I could be wrong though. What do you need dummy for? Thanks, --Warren Selby, dCAP On Oct 23, 2012, at 10:28 AM, Jerry Geis ge...@pagestation.com wrote: I need to use the dahdi dummy driver. Its not being compiled at this time. When I go into tools subdirectory under dahdi-linux-complete-2.4.1 and do make menuselect all I get is CC= CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent nmenuselect make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Nothing to be done for `nmenuselect'. make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' CC= CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent gmenuselect make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Nothing to be done for `gmenuselect'. make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools' Terminal must be at least 80 x 27. menuselect changes NOT saved! make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools' How can I get the dahdi_dummy.c driver compiled? 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 -- _ -- 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] dahdi dummy
Need dummy to provide timing on machines that do not have a tdm board. Also meetme dependency was on dummy or one of the tdm card. I believe meetme has been rewritten since then. Mitul On Oct 23, 2012 9:58 PM, Warren Selby wcse...@selbytech.com wrote: If I remember correctly, dahdi dummy was removed and the functionally added by default when you load dahdi with no TDM cards installed. I could be wrong though. What do you need dummy for? Thanks, --Warren Selby, dCAP On Oct 23, 2012, at 10:28 AM, Jerry Geis ge...@pagestation.com wrote: I need to use the dahdi dummy driver. Its not being compiled at this time. When I go into tools subdirectory under dahdi-linux-complete-2.4.1 and do make menuselect all I get is CC= CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent nmenuselect make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Nothing to be done for `nmenuselect'. make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' CC= CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent gmenuselect make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Nothing to be done for `gmenuselect'. make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools/menuselect' make[1]: Entering directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools' Terminal must be at least 80 x 27. menuselect changes NOT saved! make[1]: Leaving directory `/home/silentm/MessageNet/digium/dahdi-linux-complete-2.4.1.2+2.4.1/tools' How can I get the dahdi_dummy.c driver compiled? 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 -- _ -- 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] dahdi dummy
If I remember correctly, dahdi dummy was removed and the functionally added by default when you load dahdi with no TDM cards installed. I could be wrong though. What do you need dummy for? I am using CentOS 5 on a machine and have no issue with alsa dropping audio. I dual booted and have CentOS 6 on a machine and after a littel while the audio drops out and then comes back. The machine is a client to the master asterisk and just plays files out the alsa or console/dsp port. I thought it was a dahdi_dummy driver issue. I dont know what else to look at. How do I tell for sure that a timer is being grabbed and used? I see nothing in dmesg 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] Call drop weirdness
I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get a data dump to see what actually happens. Some times there is a bit of artifacting which takes place just prior to the drop, but mostly nothing: it just drops. I've checked and rechecked firewall settings. Bandwidth consumption on the Inet link varies, but the dropped audio happens even on off-peak times. I'm considering giving the Asterisk box a public IP on one IF and bypassing the FW to rule out NAT weirdness. Any thoughts on things to look at would be greatly appreciated. Kind Regards, Chris I'm not sure if this is any help, but I had a familiar issue to this, except it involved transferring to another internal extension. The symptoms were the same though. Only outbound audio would cut out and it was very sporadic (~10% of transfers). The issue ended up being with the trunking service and their spotty support with UPDATE messages. We had to disable rpid_update in sip.conf and a couple other bits that I can't offhand remember. I'd check with the trunk provider on the issue. Best of luck, Brett Lehrer -- _ -- 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] Can't get Lua Pattern Matching to work
l can't see to get the Lua extension matching to work:
[Oct 23 19:13:12] NOTICE[4288]: chan_sip.c:23577 handle_request_invite:
Call from 'user' (XXX.XXX.XXX.XXX:33962) to extension '107' rejected
because extension not found in context 'luaentry'.
extensions = {}
extensions.luaentry = {}
extensions.luaentry[_NXX] = function(c,e)
return app.goto(ael-dial-exten,
e, 1);
end
Shouldn't _NXX match 107?
I also tried:
extensions = {
[luaentry] = {
[NXX] = function(c,e)
app.verbose(Redirecting to internal extension)
return app.goto(ael-dial-exten, e, 1);
end;
[_1NXXNXX] = function(c,e)
internalDialOut(e)
end;
[_NXXNXX] = function(c,e)
internalDialOut(1..e)
end;
i = function()
app.playback(invalid)
app.hangup()
end;
};
};
And that didn't work either.
Any tips or tricks?
My users.conf looks like:
[503]
fullname = Cody Harris
email = qbasi...@gmail.com
secret = XX
vmsecret =
context = luaentry
hasvoicemail = yes
callwaiting = yes
hasip = yes
qualify=yes
nat=yes
host=dynamic
canreinvite=no
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Can't get Lua Pattern Matching to work
Sorry for the reply so quick,
I should mention I'm running on 10.9.0
On Tue, Oct 23, 2012 at 3:15 PM, Cody Harris qbasi...@gmail.com wrote:
l can't see to get the Lua extension matching to work:
[Oct 23 19:13:12] NOTICE[4288]: chan_sip.c:23577 handle_request_invite:
Call from 'user' (XXX.XXX.XXX.XXX:33962) to extension '107' rejected
because extension not found in context 'luaentry'.
extensions = {}
extensions.luaentry = {}
extensions.luaentry[_NXX] = function(c,e)
return app.goto(ael-dial-exten,
e, 1);
end
Shouldn't _NXX match 107?
I also tried:
extensions = {
[luaentry] = {
[NXX] = function(c,e)
app.verbose(Redirecting to internal extension)
return app.goto(ael-dial-exten, e, 1);
end;
[_1NXXNXX] = function(c,e)
internalDialOut(e)
end;
[_NXXNXX] = function(c,e)
internalDialOut(1..e)
end;
i = function()
app.playback(invalid)
app.hangup()
end;
};
};
And that didn't work either.
Any tips or tricks?
My users.conf looks like:
[503]
fullname = Cody Harris
email = qbasi...@gmail.com
secret = XX
vmsecret =
context = luaentry
hasvoicemail = yes
callwaiting = yes
hasip = yes
qualify=yes
nat=yes
host=dynamic
canreinvite=no
--
_
-- 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] Can't get Lua Pattern Matching to work
_NXX is only going to match a 3 digit number. I think you need _NXX. For
this case.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cody Harris
Sent: Tuesday, October 23, 2012 2:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Can't get Lua Pattern Matching to work
Sorry for the reply so quick,
I should mention I'm running on 10.9.0
On Tue, Oct 23, 2012 at 3:15 PM, Cody Harris qbasi...@gmail.com wrote:
l can't see to get the Lua extension matching to work:
[Oct 23 19:13:12] NOTICE[4288]: chan_sip.c:23577 handle_request_invite: Call
from 'user' (XXX.XXX.XXX.XXX:33962) to extension '107' rejected because
extension not found in context 'luaentry'.
extensions = {}
extensions.luaentry = {}
extensions.luaentry[_NXX] = function(c,e)
return app.goto(ael-dial-exten, e,
1);
end
Shouldn't _NXX match 107?
I also tried:
extensions = {
[luaentry] = {
[NXX] = function(c,e)
app.verbose(Redirecting to internal extension)
return app.goto(ael-dial-exten, e, 1);
end;
[_1NXXNXX] = function(c,e)
internalDialOut(e)
end;
[_NXXNXX] = function(c,e)
internalDialOut(1..e)
end;
i = function()
app.playback(invalid)
app.hangup()
end;
};
};
And that didn't work either.
Any tips or tricks?
My users.conf looks like:
[503]
fullname = Cody Harris
email = qbasi...@gmail.com
secret = XX
vmsecret =
context = luaentry
hasvoicemail = yes
callwaiting = yes
hasip = yes
qualify=yes
nat=yes
host=dynamic
canreinvite=no
--
_
-- 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] Can't get Lua Pattern Matching to work
On Tue, Oct 23, 2012 at 2:18 PM, Danny Nicholas da...@debsinc.com wrote: _NXX is only going to match a 3 digit number. I think you need _NXX. For this case. ** Wouldn't _NXX match 107? That's what he's saying isn't working. ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Cody Harris *Sent:* Tuesday, October 23, 2012 2:17 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Can't get Lua Pattern Matching to work Shouldn't _NXX match 107?** ** -- ** -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] followme ldap realtime problems
Hi all, I am having problems configuring followme with realtime ldap database. The error I get is: [2012-10-23 21:01:40] WARNING[16004]: res_config_ldap.c:967 realtime_multi_ldap: realtime retrieval requires at least 1 parameter and 1 value to search on. [2012-10-23 21:01:40] WARNING[16004]: app_followme.c:1175 app_exec: Profile requested, adomaitis, not found in the configuration. cli commands: realtime load followme name adomaitis realtime load followme_numbers name adomaitis works as expected, meaning that shows informations fetched from ldap. relevant config are extconfig.conf followme = ldap,ou=People,dc=example,dc=com,followme followme_numbers = ldap,ou=People,dc=example,dc=com,followmenumbers res_ldap.conf: [followme] name = AstFollowMeName musicclass = AstFollowMeMusic musiconhold = AstFollowMeMusic music = AstFollowMeMusic context = AstFollowMeContext takecall = AstFollowMeTakeCall declinecall = AstFollowMeDeclineCall call_from_prompt = AstFollowMePromptCallFrom norecording_prompt = AstFollowMePromptNoRecording options_prompt = AstFollowMePromptOptions hold_prompt = AstFollowMePromptHold status_prompt = AstFollowMePromptStatus sorry_prompt = AstFollowMePromptSorry additionalFilter = (objectClass=asteriskFollowMe) [followmenumbers] name = AstFollowMeProfileName ordinal = AstFollowMeNumberPriority phonenumber = AstFollowMeNumber timeout = AstFollowMeNumberTimeOut additionalFilter = (objectClass=asteriskFollowMeNumbers) I have defined schema for followme. Does anyone know what can be the issue? Thanks Liutauras -- _ -- 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] Can't get Lua Pattern Matching to work
l can't see to get the Lua extension matching to work:
[Oct 23 19:13:12] NOTICE[4288]: chan_sip.c:23577
handle_request_invite: Call from 'user' (XXX.XXX.XXX.XXX:33962) to
extension '107' rejected because extension not found in context
'luaentry'.
extensions = {}
extensions.luaentry = {}
extensions.luaentry[_NXX] = function(c,e)
return app.goto(ael-dial-exten, e, 1);
end
Shouldn't _NXX match 107?
No. I wouldn't expect it to match 107.
X
Matches any single digit from 0 to 9.
Z
Matches any single digit from 1 to 9.
N
Matches any single digit from 2 to 9.
Richard
--
_
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Re: [asterisk-users] Can't get Lua Pattern Matching to work
Nope – see page 138 of the Asterisk manual – N matches 2-9 and X matches 0-9 so the N excludes numbers starting with 0 or 1. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Tuesday, October 23, 2012 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get Lua Pattern Matching to work On Tue, Oct 23, 2012 at 2:18 PM, Danny Nicholas da...@debsinc.com wrote: _NXX is only going to match a 3 digit number. I think you need _NXX. For this case. Wouldn't _NXX match 107? That's what he's saying isn't working. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cody Harris Sent: Tuesday, October 23, 2012 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can't get Lua Pattern Matching to work Shouldn't _NXX match 107? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Can't get Lua Pattern Matching to work
On Tue, Oct 23, 2012 at 2:34 PM, Danny Nicholas da...@debsinc.com wrote: Nope – see page 138 of the Asterisk manual – N matches 2-9 and X matches 0-9 so the N excludes numbers starting with 0 or 1. ** Ah, sorry, I was thrown off by you suggesting _NXX. which wouldn't have matched either. So Cody needs _ZXX as the pattern. ** On Tue, Oct 23, 2012 at 2:18 PM, Danny Nicholas da...@debsinc.com wrote: ** _NXX is only going to match a 3 digit number. I think you need _NXX. For this case. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Wireshark AMI Dissector
Hello everyone, Does anyone know of a Wireshark AMI (Asterisk Manager Interface) dissector? Decode as telnet and display filter telnet.data kind of work but TCP reassembly can't happen without a better understanding of the protocol... Thanks! -- Kristian Kielhofner -- _ -- 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] Can't get Lua Pattern Matching to work
Ah great, thanks guys for the help! I have it working now! On Tue, Oct 23, 2012 at 3:37 PM, Christopher Harrington ch...@acsdi.comwrote: On Tue, Oct 23, 2012 at 2:34 PM, Danny Nicholas da...@debsinc.com wrote: Nope – see page 138 of the Asterisk manual – N matches 2-9 and X matches 0-9 so the N excludes numbers starting with 0 or 1. ** Ah, sorry, I was thrown off by you suggesting _NXX. which wouldn't have matched either. So Cody needs _ZXX as the pattern. ** On Tue, Oct 23, 2012 at 2:18 PM, Danny Nicholas da...@debsinc.com wrote: ** _NXX is only going to match a 3 digit number. I think you need _NXX. For this case. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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
[asterisk-users] as soon as Phone rings I'm disconnected yet phone rings two more times
One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal
adapter onto my asterisk. I have the 8x8 box connected to the Internet, and
the phone line connected to an fxo port on a Cisco router:
voice-port 0/2/0
connection plar opx 5000
caller-id enable
dial-peer voice 200 voip
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
!
sip-ua
sip-server ipv4:172.16.200.212 -- Asterisk server
When I make a call from the PSTN to the 8x8 box, it does send ring back to the
asterisk server and the Digium phone does ring. However, as soon as the phone
rings the call disconnects yet the actual phone, extension 5000, rings two
times before it hangs up, also.
The following output is what I see on the Asterisk console:
asterisk*CLI
== Using SIP RTP CoS mark 5
[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite: Call
from '' (172.16.200.1:65451) to extension '5000' rejected because extension not
found in context 'default'.
== Using SIP RTP CoS mark 5
-- Executing [5000@pstn-incoming:1] Dial(SIP/172.16.200.1-0006,
SIP/5000,20|p) in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/5000
-- SIP/5000-0007 is ringing
== Spawn extension (pstn-incoming, 5000, 1) exited non-zero on
'SIP/172.16.200.1-0006'
The 172.16.200.1 is my router.
sip.conf excerpt:
[5000]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
secret=cisco123
mailbox=5000@phones
[172.16.200.1]
context=pstn-incoming
type=friend
host=172.16.200.1
dtmfmode=rfc2833
disallow=all
allow=ulaw
[phones]
exten = 5000,1,Dial(SIP/${EXTEN},20|p)
exten = 5000,n,Hangup
[pstn-incoming]
include=phones
Any help would be greatly appreciated,
Thanks,--
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