Re: [asterisk-users] Queue Ring inuse is shared ?
Hi, I found in another mail that setting call-limit=1 in the sip configuration works. I tried that. It works but in that case the agents are not able to transfer the call to another extension, because only one call is allowed at a time. Any other methods ? Thanks & Regards Shanavaz. From: Shanavaz E A To: "asterisk-users@lists.digium.com" Sent: Saturday, June 22, 2013 1:11 PM Subject: [asterisk-users] Queue Ring inuse is shared ? Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until the call gets answered. Is this a normal behaviour ? Can we prevent it? Can we set "not to ring" any queue member if he is answering a call either in the same queue or a different queue? Pls guide me. Regards Shanavaz. -- _ -- 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] IAX2 netsock error with name resolution
After changing my dialplan as suggested, there is no socket error, but
getting Busy/Congested, and the call is hanging up, let me check that
part...
Earlier my dialplan was,
;exten => _2XXX,1,Dial(SIP/${EXTEN}@${MANIAX},30)
and I changed like this exten => _2XXX,1,Dial(${MANIAX}/${EXTEN},30)
whether the SIP matters?
And now since its a SIP extension in other side, am getting failed because
the extension is not able to find.
Regards.
On Sun, Jun 23, 2013 at 5:22 PM, Alec Davis wrote:
>
> > -- Executing [2001@Test:1] Dial("SIP/4090-0005",
> "SIP/2001@IAX2/IND-MAN,30") in new stack
> > [Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
> sip_request_call: Conflicting extension values given. Using '2001' and not
> 'IND-MAN'
> > == Using SIP RTP CoS mark 5
> > [Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
> ast_sockaddr_resolve: getaddrinfo("IAX2", "(null)", ...): Temporary failure
> in name resolution
> > [Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
> No such host: IAX2
> > [Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
> dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
> Subscriber absent)
> > == Everyone is busy/congested at this time (1:0/0/1)
>
> Try this syntax Dial(IAX2/IND-MAN/2001,30)
> Where IND-MAN is the name of a peer/friend [IND-MAN] defined in iax.conf
> and 2001 is the extension on the remote system 'IND-MAN' where 2001 dials
> SIP/2001
>
> Alec Davis
>
>
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[asterisk-users] Upgrading from 1.4 to 11.4.0
Hi
After upgrading from 1.4 to 11.4.0, I am able to receive calls
and direct them to extensions via defined trunks.
However, when making outgoing calls I receive the following
error:
-- Executing [00044111@default:4] Dial("SIP/fixedline-0004",
"SIP/mydevice/0044111,60,w") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/mydevice/0044111
WARNING[13053][C-0002]: channel.c:6164 ast_channel_make_compatible_helper:
No path to translate from SIP/mydevice-0005 to SIP/hardphone-0004
And when I try to try to initiate a call with a manager script, I receive an
authentication error from the script.
How might I find more info to help diagnose either or both of these issues?
--
Eric Smith
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Re: [asterisk-users] IAX2 netsock error with name resolution
> -- Executing [2001@Test:1] Dial("SIP/4090-0005",
"SIP/2001@IAX2/IND-MAN,30") in new stack
> [Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
sip_request_call: Conflicting extension values given. Using '2001' and not
'IND-MAN'
> == Using SIP RTP CoS mark 5
> [Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo("IAX2", "(null)", ...): Temporary failure
in name resolution
> [Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
No such host: IAX2
> [Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
> == Everyone is busy/congested at this time (1:0/0/1)
Try this syntax Dial(IAX2/IND-MAN/2001,30)
Where IND-MAN is the name of a peer/friend [IND-MAN] defined in iax.conf
and 2001 is the extension on the remote system 'IND-MAN' where 2001 dials
SIP/2001
Alec Davis
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[asterisk-users] IAX2 netsock error with name resolution
Am getting netsock error like this when using IAX2,
Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid =
4270)
== Using SIP RTP CoS mark 5
-- Executing [2001@Test:1] Dial("SIP/4090-0005",
"SIP/2001@IAX2/IND-MAN,30")
in new stack
[Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
sip_request_call: Conflicting extension values given. Using '2001' and not
'IND-MAN'
== Using SIP RTP CoS mark 5
[Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo("IAX2", "(null)", ...): Temporary failure
in name resolution
[Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
No such host: IAX2
[Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
My hostname are proper,
in /etc/hostname and /etc/sysconfig/network
Even then am not able to find why am getting this error. Also am able to
ping with my own hostname.
Regards,
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