Re: [asterisk-users] Parking in Asterisk 12.0.0
Thank you for reply, Matt and Leandro! The reasons I'm not using one step parking is that it will speak which parking extension it has parked the call on and I also want a channel variable to be set about the parked call, which I need in the agi-script (ParkAndAnnounce with a not valid dial string and setting a MASTER_CHANNEL() variable in the Macro solves these issues for me). Maybe I could do a hack in the Asterisk source to achive this if there is no other solution. The reason I tried to upgrade from 11.6 to 12.0 was that I experienced problems with lost (ignored) DTMF events after Asterisk has released the DTMF events for what it think can be a potential dynamic feature. This problem seems to be gone in Asterisk 12.0... [Jan 30 11:13:48] DEBUG[29442][C-]: features.c:3740 feature_interpret: Feature interpret: chan=SIP/at-tcty-ssw-, peer=SIP/vpn-sbc-0001, code=*8, sense=1, features=0, dynamic=parkswitch#parkswitch [Jan 30 11:13:49] DEBUG[29445][C-]: res_rtp_asterisk.c:2833 create_dtmf_frame: Creating BEGIN DTMF Frame: 50 (2), at 85.30.63.134:15870 [Jan 30 11:13:49] DTMF[29445][C-]: channel.c:4171 __ast_read: DTMF begin '2' received on SIP/at-tcty-ssw- *[Jan 30 11:13:49] DTMF[29445][C-]: channel.c:4185 __ast_read: DTMF begin ignored '2' on SIP/at-tcty-ssw-* [Jan 30 11:13:49] DEBUG[29445][C-]: res_rtp_asterisk.c:3165 ast_rtcp_read: Got RTCP report of 80 bytes [Jan 30 11:13:49] DEBUG[29445][C-]: res_rtp_asterisk.c:2833 create_dtmf_frame: Creating END DTMF Frame: 50 (2), at 85.30.63.134:15870 I spoke to Olle E Johansson about this issue today and he pointed me to a SVN branch he has made for Asterisk 1.8 which should solve the ignored DTMF events (http://svnview.digium.com/svn/asterisk/team/oej/rana-dtmf-duration-1.8/) I have now quickly tested his code and it seems to work, so I will propably go with Asterisk 1.8. Thanks again. -- Anders On Thu, Jan 30, 2014 at 2:58 PM, Leandro Dardini ldard...@gmail.com wrote: I have converted the normal Park application and I can only alert you about the syntax change. I suspect also in the ParkAndAnnounce command, the parameters are ordered completely different. Leandro Please go ahead an open an issue for this - issues.asterisk.org. The problem here is that you are attempting to enter into a Parking bridge while you are still technically in a bridge. The DTMF features that account for the 'normal' mechanism of doing this - the one touch parking feature - recognize that you are in a bridge and do a safe transfer from the existing bridge to the parking bridge. By jumping out to a macro/gosub and directly going in through the ParkAndAnnounce application, you are bypassing that logic. The code in bridge_channel_internal_join is preventing you from going into the parking bridge as it knows that you have not yet safely left the bridge you are in. We'll take a look and see if there's a way to allow this to happen again. For now, you should use the one touch parking feature. Matt -- _ -- 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 as a media gateway
I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting. This is what I was thinking of trying. 1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc). 2. The mains server will not handle RTP traffic, it will send the RTP traffic to another system (another asterisk box?) for processing. At the end of the day, what I am hoping for is to have 1 brain, and mutiple work horse audio gateways that can be added and removed as needed. Has this been done? Can anyone point me to some documentation on how others have done this? It's always fun to play Richard Seguin. -- _ -- 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] e911 Signalling
Hi,
We've got a dedicated T1 with two trunks running into our ILECs
selective router for 911. Split out of the T1 into two MF CAMA trunks
on ILEC side.
I'm trying to use asterisks e911 signaling, but I'm having trouble with
the dial command. (== Everyone is busy/congested at this time (1:1/0/0))
I'm missing something and I'm thinking it has to do with the hookstate
of the dahdi channel.
If anyone has a similar situation and want to provide some guidance, I'd
sure appreciate it.
Thanks!
Adam
Here's my config:
DAHDI version 2.8.0.1
[root@e911 dahdi]# dahdi_hardware
pci::02:01.0 wct1xxp+ e159:0001 Digium Wildcard T100P T1/PRI
or E100P E1/PRA Board
e911*CLI core show version
Asterisk 11.7.0 built by root @ e911 on a x86_64 running Linux on
2014-01-28 15:50:19 UTC
/etc/dahdi/system.conf
span=1,1,0,esf,b8zs
em=1-2
/etc/asterisk/chan_dahdi.conf
[channels]
group=1
signalling=e911
channel=1-2
/etc/asterisk/extensions.conf
[InFromSIP]
exten = s,1,dial(DAHDI/1/${CALLERID(num)})
e911*CLI dahdi show status
Description Alarms IRQbpviol CRC
Fra Codi Options LBO
Digium Wildcard T100P T1/PRI Card 0 OK 0 0 0
ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1)
e911*CLI dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState Description
pseudodefaultdefault
In Service
1public default
In Service
2public default
In Service
e911*CLI dahdi show channel 1
Channel: 1
Description:
File Descriptor: 7
Span: 1
Extension:
Dialing: no
Context: public
Caller ID:
Calling TON: 0
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: E911 (MF)
Radio: 0
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Offhook
Here's a debug from a 911 call.
[Jan 31 11:29:53] DEBUG[9876][C-0005]: pbx.c:4890
pbx_extension_helper: Launching 'Dial'
-- Executing [s@InFromSIP:1] Dial(SIP/SIP-0005,
DAHDI/1/212001) in new stack
[Jan 31 11:29:53] DEBUG[9876][C-0005]: sig_analog.c:820
analog_available: analog_available 1
[Jan 31 11:29:53] DEBUG[9876][C-0005]: sig_analog.c:845
analog_available: Channel 1 off hook, can't use
[Jan 31 11:29:53] WARNING[9876][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'DAHDI' (cause 17 -
User busy)
== Everyone is busy/congested at this time (1:1/0/0)
[Jan 31 11:29:53] DEBUG[9876][C-0005]: app_dial.c:3100
dial_exec_full: Exiting with DIALSTATUS=BUSY.
-- Auto fallthrough, channel 'SIP/SIP-0005' status is 'BUSY'
--
_
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Re: [asterisk-users] Asterisk as a media gateway
On Fri, Jan 31, 2014 at 11:27 AM, richard.seg...@marisec.ca wrote: I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting. This is what I was thinking of trying. 1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc). 2. The mains server will not handle RTP traffic, it will send the RTP traffic to another system (another asterisk box?) for processing. At the end of the day, what I am hoping for is to have 1 brain, and mutiple work horse audio gateways that can be added and removed as needed. Has this been done? Can anyone point me to some documentation on how others have done this? It's always fun to play Yes, this is basically functionality for Asterisk. If you are using SIP, you want to REINVITE media away from your core Asterisk box. I suggest picking up the book[1] and reading the chapter on connecting multiple Asterisk boxes together. [1] http://www.asteriskdocs.org/ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] callfiles.call
hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06 MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten = s,1,Ringing() exten = s,n,Playback(hello-world) exten = s,n,Dial(SIP/105) exten = s,n,Hangup() it works with one number how can i do in order to create a callfiles with a lot of numbers i try to create a callfiles.call like that Channel: SIP/watara/0661xx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 Channel: SIP/watara/0669xx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 but he call only the last number, any help please thanks and regards -- _ -- 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] callfiles.call
On Fri, Jan 31, 2014 at 6:16 PM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, Hi i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06 MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten = s,1,Ringing() exten = s,n,Playback(hello-world) exten = s,n,Dial(SIP/105) exten = s,n,Hangup() it works with one number how can i do in order to create a callfiles with a lot of numbers i try to create a callfiles.call like that Channel: SIP/watara/0661xx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 Channel: SIP/watara/0669xx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 but he call only the last number, From my limited knowledge about call files, you need 1 file per call. So you'd make two files in this instance, https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files I may be wrong, if I am, someone will tell me so, but I'm pretty sure it's 1 file per call you want, as each file has 1 set of instructions for the dialplan any help please thanks and regards 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 -- _ -- 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] callfiles.call
On Fri, 31 Jan 2014, Salaheddine Elharit wrote: it works with one number how can i do in order to create a callfiles with a lot of numbers You don't. A call file is a request to call a channel and when the channel answers, either call another channel or execute an application. Thus, you need a separate call file for every call. One thing to watch for: create the call file on the same file system as your outgoing spool directory in a temporary directory and then 'mv' the file to your outgoing spool directory. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] callfiles.call
On Fri, 31 Jan 2014, Steve Edwards wrote: A call file is a request to call a channel and when the channel answers, either call another channel or execute an application. Sorry. Need another cup of tea. A call file is a request to call a channel and when the channel answers, either enter your dialplan or execute an application. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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-Tools 2.9.0.1 Hotfix Now Available
The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.9.0 DAHDI-Tools-v2.9.0.1 dahdi-linux-complete-2.9.0+2.9.0.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete This hotfix corrects a rare race condition bug where a span could be misconfigured when using dahdi_cfg -c. Shortlog of dahdi-tools changes since v2.9.0: Shaun Ruffell (1): hotplug: Do not run auto span configuration if spans are auto assigned. The diffstat from the dahdi-tools v2.9.0 release: hotplug/dahdi_span_config | 11 +++ 1 file changed, 11 insertions(+) For a full list of changes in these releases, please see the shortlog at: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.9.0 http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.9.0.1 Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira [1] https://issues.asterisk.org/jira/browse/DAHLIN [2] https://issues.asterisk.org/jira/browse/DAHTOOL Thank you for your continued support of Asterisk! -- _ -- 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] how to get full channel name - AMI cuts off
On Thu, Jan 30, 2014 at 5:45 PM, Justin Killen jkil...@allamericanasphalt.com wrote: Using Dahdi/PRI, I end up with channel names like 'DAHDI/i8/9995551212-4d6B', but when I do a 'core show channels' it cuts off those names to only 'DAHDI/i8/9995551212-'. This is the same for AMI. Is there a way to get the full channel name within AMI? I'm using asterisk 11.7.0 Since you are using AMI, use the AMI CoreShowChannels action instead of the AMI Command action to run the CLI core show channels command. Use: Action: CoreShowChannels ActionID: your-optional-id Instead of: Action: Command ActionID: your-optional-id Command: core show channels Using the AMI CoreShowChannels action will give you more information as well as not truncating long strings. It is also the recommended way to get that information. 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] e911 Signalling
Well, good news, it was the telco side. They had their ports disabled.
I ended up having to use signalling=fgccamamf.
Adam
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Vocks
Sent: Friday, January 31, 2014 11:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] e911 Signalling
Hi,
We've got a dedicated T1 with two trunks running into our ILECs
selective router for 911. Split out of the T1 into two MF CAMA trunks
on ILEC side.
I'm trying to use asterisks e911 signaling, but I'm having trouble with
the dial command. (== Everyone is busy/congested at this time (1:1/0/0))
I'm missing something and I'm thinking it has to do with the hookstate
of the dahdi channel.
If anyone has a similar situation and want to provide some guidance, I'd
sure appreciate it.
Thanks!
Adam
Here's my config:
DAHDI version 2.8.0.1
[root@e911 dahdi]# dahdi_hardware
pci::02:01.0 wct1xxp+ e159:0001 Digium Wildcard T100P T1/PRI
or E100P E1/PRA Board
e911*CLI core show version
Asterisk 11.7.0 built by root @ e911 on a x86_64 running Linux on
2014-01-28 15:50:19 UTC
/etc/dahdi/system.conf
span=1,1,0,esf,b8zs
em=1-2
/etc/asterisk/chan_dahdi.conf
[channels]
group=1
signalling=e911
channel=1-2
/etc/asterisk/extensions.conf
[InFromSIP]
exten = s,1,dial(DAHDI/1/${CALLERID(num)})
e911*CLI dahdi show status
Description Alarms IRQbpviol CRC
Fra Codi Options LBO
Digium Wildcard T100P T1/PRI Card 0 OK 0 0 0
ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1)
e911*CLI dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState Description
pseudodefaultdefault
In Service
1public default
In Service
2public default
In Service
e911*CLI dahdi show channel 1
Channel: 1
Description:
File Descriptor: 7
Span: 1
Extension:
Dialing: no
Context: public
Caller ID:
Calling TON: 0
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: E911 (MF)
Radio: 0
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Offhook
Here's a debug from a 911 call.
[Jan 31 11:29:53] DEBUG[9876][C-0005]: pbx.c:4890
pbx_extension_helper: Launching 'Dial'
-- Executing [s@InFromSIP:1] Dial(SIP/SIP-0005,
DAHDI/1/212001) in new stack
[Jan 31 11:29:53] DEBUG[9876][C-0005]: sig_analog.c:820
analog_available: analog_available 1
[Jan 31 11:29:53] DEBUG[9876][C-0005]: sig_analog.c:845
analog_available: Channel 1 off hook, can't use
[Jan 31 11:29:53] WARNING[9876][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'DAHDI' (cause 17 -
User busy)
== Everyone is busy/congested at this time (1:1/0/0)
[Jan 31 11:29:53] DEBUG[9876][C-0005]: app_dial.c:3100
dial_exec_full: Exiting with DIALSTATUS=BUSY.
-- Auto fallthrough, channel 'SIP/SIP-0005' status is 'BUSY'
--
_
-- 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:
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