Re: [asterisk-users] Strange problem on ougoing call
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B Ipbx
Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
IP Adress: 172.16.0.70
Second IP: 172.16.1.70 (used for phone lan)
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Linksys SPA942 A:
IP Adress: 172.16.1.200
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User01
Linksys SPA942 B:
IP Adress: 172.16.1.220
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User02
On Server A Trader, we have two sip account:
accountname: USER01 for user in group 1
accountname: USER02 for user in group 2
On Server B Ipbx, i use registry:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
for two connection to the Trader Server. Registry is good:
on server A Trader:
trader*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
On server B Ipbx, i have into my sip.conf after the registry:
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
and in extensions.conf:
[I-User01]
exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
[I-User02]
exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
When i call with Linksys SPA942 A, i use the context I-User01 and
the call are sent
to SIP account USER01 and No problems.
When i call with Linksys SPA942 B, i use the context I-User02 and
the call are sent
to SIP account USER02 but Server A Trader reject the call
immediatly with this error:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
Olivier and 906280 is the information that i have on the Linksys
SPA942 B, 906280 is the username used between
best ? hihi
Olivier
Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you are
trying to achieve with proper IP addresses and declared sip ext. names.
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
Somehow it reminds of the same situation I always face when a peer is
declared with the same name as of the dialing one on second server - only
Its just not registered there instead registered on server-1.
So when the call comes in from server-1 to server-2 fromuser=olivier which
is not registered on server-2 but is declared. Server-2 thinks that this is
my valid extension but it is not registered with me and so lets authenticate
this one and here it fails and rejects the call.
BR,
Sammy.
On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
i see the registration:
ipbx*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr
Re: [asterisk-users] Strange problem on ougoing call
2012/4/25 Olivier CALVANO o.calv...@gmail.com
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B Ipbx
Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
IP Adress: 172.16.0.70
Second IP: 172.16.1.70 (used for phone lan)
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Linksys SPA942 A:
IP Adress: 172.16.1.200
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User01
Linksys SPA942 B:
IP Adress: 172.16.1.220
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User02
On Server A Trader, we have two sip account:
accountname: USER01 for user in group 1
accountname: USER02 for user in group 2
On Server B Ipbx, i use registry:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
for two connection to the Trader Server. Registry is good:
on server A Trader:
trader*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
On server B Ipbx, i have into my sip.conf after the registry:
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
and in extensions.conf:
[I-User01]
exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
[I-User02]
exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
When i call with Linksys SPA942 A, i use the context I-User01 and
the call are sent
to SIP account USER01 and No problems.
When i call with Linksys SPA942 B, i use the context I-User02 and
the call are sent
to SIP account USER02 but Server A Trader reject the call
immediatly with this error:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
Olivier and 906280 is the information that i have on the Linksys
SPA942 B, 906280 is the username used between
best ? hihi
Olivier
Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you
are
trying to achieve with proper IP addresses and declared sip ext. names.
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
Somehow it reminds of the same situation I always face when a peer is
declared with the same name as of the dialing one on second server - only
Its just not registered there instead registered on server-1.
So when the call comes in from server-1 to server-2 fromuser=olivier
which
is not registered on server-2 but is declared. Server-2 thinks that this
is
my valid extension but it is not registered with me and so lets
authenticate
this one and here it fails and rejects the call.
BR,
Sammy.
On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
i see the registration:
ipbx*CLI sip show registry
Host dnsmgr Username Refresh
[asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??
Hi i have a lot of error in the CLI of one of my Asterisk: [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted [Apr 25 09:30:47] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:49] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:50] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted [Apr 25 09:30:51] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x862b178 (len 886) to 172.16.251.46:5060 returned -1: Operation not permitted [Apr 25 09:30:53] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:54] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted anyone know what is this error ? thanks olivier -- _ -- 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] Strange problem on ougoing call
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :
2012/4/25 Olivier CALVANO o.calv...@gmail.com
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B Ipbx
Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
IP Adress: 172.16.0.70
Second IP: 172.16.1.70 (used for phone lan)
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Linksys SPA942 A:
IP Adress: 172.16.1.200
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User01
Linksys SPA942 B:
IP Adress: 172.16.1.220
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User02
On Server A Trader, we have two sip account:
accountname: USER01 for user in group 1
accountname: USER02 for user in group 2
On Server B Ipbx, i use registry:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
for two connection to the Trader Server. Registry is good:
on server A Trader:
trader*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
On server B Ipbx, i have into my sip.conf after the registry:
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
and in extensions.conf:
[I-User01]
exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
[I-User02]
exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
When i call with Linksys SPA942 A, i use the context I-User01 and
the call are sent
to SIP account USER01 and No problems.
When i call with Linksys SPA942 B, i use the context I-User02 and
the call are sent
to SIP account USER02 but Server A Trader reject the call
immediatly with this error:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
Olivier and 906280 is the information that i have on the Linksys
SPA942 B, 906280 is the username used between
best ? hihi
Olivier
Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you
are
trying to achieve with proper IP addresses and declared sip ext. names.
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
Somehow it reminds of the same situation I always face when a peer is
declared with the same name as of the dialing one on second server -
only
Its just not registered there instead registered on server-1.
So when the call comes in from server-1 to server-2 fromuser=olivier
which
is not registered on server-2 but is declared. Server-2 thinks that this
is
my valid extension but it is not registered with me and so lets
authenticate
this one and here it fails and rejects the call.
BR,
Sammy.
On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has
different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
[asterisk-users] Hangup Cause and SIP Response Code
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
Thanks
Bryant
--
_
-- 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] CONNECTEDLINE() updated during SIP events?
Hi, I have read the excellent information here: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information and believe I have an understanding of what is offered. I have a couple of questions: - Is it possible to update COLP/COLR when a SIP redirect occurs, or when a SIP divert is in place? If so, how? - Is it possible to have the COLP/COLR information updated when a SIP attended transfer is completed? If so how? In both of the above cases, there is no obvious dialplan execution when the calls are redirected, diverted or masqueraded, so we cannot update the CONNECTEDLINE() information trivially. Or am I missing an obvious trick? Thanks, 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
[asterisk-users] Asterisk - Nortel transfer problem
Hi Carlos It could help if you can get a trace of the call transfer from Nortel to SIP extension on the Asterisk (1303), if no way to get from Nortel get from Asterisk. I guest operator try to make a bind call transfer, without wait complete DR2 signalling exchange at least minimal time exchange DR2 signalling between Nortel and Asterisk is about 5 sec. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.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] CONNECTEDLINE() updated during SIP events?
I have read the excellent information here: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information and believe I have an understanding of what is offered. I have a couple of questions: - Is it possible to update COLP/COLR when a SIP redirect occurs, or when a SIP divert is in place? If so, how? All redirecting activity is valid only before the associated calls are answered. After the calls are answered, it is connected-line updates. The redirecting interception macros are invoked before the outgoing call is answered when the outgoing call is redirected by an entity further down the line. If your Asterisk server is redirecting the call, the REDIRECTING information is updated by normal dialplan activity before placing the next outgoing call to the redirected to party. - Is it possible to have the COLP/COLR information updated when a SIP attended transfer is completed? If so how? Transfers generate connected line update events automatically. The connected line interception macros give you a chance to alter the connected line information as it is passed between the connected endpoints of the bridge. In both of the above cases, there is no obvious dialplan execution when the calls are redirected, diverted or masqueraded, so we cannot update the CONNECTEDLINE() information trivially. Or am I missing an obvious trick? This is the purpose of the interception macros. 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] CONNECTEDLINE() updated during SIP events?
On 25 April 2012 16:55, Richard Mudgett rmudg...@digium.com wrote: [snip] - Is it possible to have the COLP/COLR information updated when a SIP attended transfer is completed? If so how? Transfers generate connected line update events automatically. The connected line interception macros give you a chance to alter the connected line information as it is passed between the connected endpoints of the bridge. In both of the above cases, there is no obvious dialplan execution when the calls are redirected, diverted or masqueraded, so we cannot update the CONNECTEDLINE() information trivially. Or am I missing an obvious trick? This is the purpose of the interception macros. Ah, thank you. I was looking at it back-to-front. The key bit is Transfers generate connected line update events automatically. - I can now see this in the source code in ast_do_masquerade() and elsewhere. This then lets you use the macros as you describe. A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms out there as follows: - Update RPID and PAI (ITSP and trunks often understand this) - Update Contact: header (Aastra handsets use this) - A SIP INFO packet if Supported: callerid is specified (Older snom firmware uses this) - Update From: header if Supported: from-change is specified (RFC 4916, snom, Yealink) Are there existing plans to support any of these other methods? If not, I will almost certainly add them for my own use, and submit the code. 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] CONNECTEDLINE() updated during SIP events?
On 04/25/2012 11:54 AM, Steve Davies wrote: A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms out there as follows: - Update RPID and PAI (ITSP and trunks often understand this) - Update Contact: header (Aastra handsets use this) - A SIP INFO packet if Supported: callerid is specified (Older snom firmware uses this) - Update From: header if Supported: from-change is specified (RFC 4916, snom, Yealink) Are there existing plans to support any of these other methods? If not, I will almost certainly add them for my own use, and submit the code. No, we have no plans at this time to go beyond RPID and PAI support. Those two appear to cover all the current endpoints that we have been able to test with, and many community members have also used with other endpoints and had success. Changing the Contact header seems quite wrong; the display-name in a URI in the Contact header is pretty much irrelevant. Changing the From header also seems wrong; that should indicate who sent the initial INVITE, not who redirected it. I don't think we'd want to merge patches that added support for either of those mechanisms. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] meetme identify user number
Daniel wrote: Hi Group, is in MeetMe any option to identify the own number (from the view of a caller)? I would like to write an option to set on the telephone an request for voice, if the room is muted. That request should display on our Conference Control Website and an Admin should unmute this person. If you have the user menu enabled, and the user is muted, then option 2 sets a 'Requests the Floor' flag. I know that the conference display feature in Web-MeetMe can interpret that flag and display a message that the caller would like to be unmated. I don't know of any other conference management apps that do, but I really have not looked into it. The request the floor feature was added in one of the early 1.6 releases, so unless you are on a truly ancient version, the backend support should be there. Dan -- _ -- 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] Open source replacement for AudioCodes nCite 1000 SBC
List users, I have an AudioCodes nCite 1000 SBC that is end-of-life and I'm looking to replace it with open source software. I believe one of the SIP proxy projects will fit my needs, but I'm a bit overwhelmed by the number of choices and I'd like the advice of experienced users before I venture too far down any one path. The projects that came to mind first were Kamailio, OpenSIPS, SER, and SIP-Router, but I'm aware that there are others and I'm open to suggestions. Please keep in mind that I'm looking for something reasonably easy to setup and administer. I'm not looking to have it setup tomorrow, but it must be something that a single skilled Linux/Asterisk administrator could take on in addition to other daily tasks. The functionality that I'm currently using on the nCite 1000 is: SIP Proxy/B2BUA and RTP Proxy * Internal call routing (private IP-to-private IP) * External call routing (external IP-to-private IP and vice versa) with topology hiding * SIP header modification * Digit manipulation (delete digits/add prefixes based on matching criteria) Connectivity * NAT traversal * External registrations (registration bindings are maintained and ports on the far end firewall are kept open) Authentication * By source IP address or range * By destination SIP proxy Session Targets and Session Target Sets * Individual SIP entities (e.g. Asterisk servers, SIP trunks) are defined as session targets * Session targets are grouped into sets with call distribution based on priorities/weights Call Routing * Static Binding: All calls to an inbound SIP proxy are routed to the same session target set via the same outbound SIP proxy * Dial Pattern: All calls to an inbound SIP proxy are routed to different session target sets via different outbound SIP proxies based on dial patterns Future Considerations * TLS/SRTP support Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Hangup Cause and SIP Response Code
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what
version of Asterisk you are using. In some versions there is a SIP_CAUSE
feature that can be used to extract that information (although this has
been reimplemented for Asterisk 11 in a way that doesn't affect
performance as much as the old method did).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Hangup Cause and SIP Response Code
Kevin
I am using 1.8.x 10.x
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what
version of Asterisk you are using. In some versions there is a SIP_CAUSE
feature that can be used to extract that information (although this has been
reimplemented for Asterisk 11 in a way that doesn't affect performance as
much as the old method did).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
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[asterisk-users] Restart single dahdi span
Hi, Is it possible yet to restart a single Dahdi span in any version of Asterisk? (instead of all of them) Thanks. -- James -- _ -- 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] Hangup Cause and SIP Response Code
On 04/25/2012 04:45 PM, brya...@zktech.com wrote:
Kevin
I am using 1.8.x 10.x
Then you have SIP_CAUSE available, although you'll have to enable it
because it is off by default due to performance concerns.
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, Kevin P. Flemingkpflem...@digium.com wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what version
of Asterisk you are using. In some versions there is a SIP_CAUSE feature that
can be used to extract that information (although this has been reimplemented
for Asterisk 11 in a way that doesn't affect performance as much as the old
method did).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
_
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asterisk-users mailing list
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Hangup Cause and SIP Response Code
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 25, 2012 6:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code On 04/25/2012 04:45 PM, brya...@zktech.com wrote: Kevin I am using 1.8.x 10.x Then you have SIP_CAUSE available, although you'll have to enable it because it is off by default due to performance concerns. Does anyone know what kind of performance hit you take from SIP_CAUSE when you are using few or no calls using chan_local? -- _ -- 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
