Re: [asterisk-users] multiple users for jabber.conf
On Wed, 2012-09-12 at 00:01 -0500, Vladimir Mikhelson wrote: > Hans, > > I did not try 10 or 11 as I run 1.8.15. Following are the related > conf files. > > gtalk.conf > > [General] > context = default > allowguest = yes ; Required if you want to accept calls > from people Not on your contact list. > bindaddr= ;; These two settings are very critical for > getting > externip= ;; gtalk audio with Asterisk server behind > NAT > disallow=all > allow=ulaw > > [guest] ;;special account for options on > guest account > disallow=all > allow=ulaw > context=from-trunk > connection= > > jabber.conf > > [general] > debug=no ;;Turn on debugging by default. > autoprune=no ;;Auto remove users from buddy list. > autoregister=yes ;;Auto register users from buddy list. > > [Jab01] ;; Label > type = client ;; Client or Component connection > serverhost = talk.google.com > ;; Route to server > username = google-user-nam...@gmail.com>/asterisk;; Username with > optional resource. > secret = >;; Password > priority = 1 ;; Resource priority > port = 5222 ;; Port to use, defaults to 5222 > usetls = yes;; TLS is required by talk.google.com, > you'll get a 'socket read error' without > usesasl = yes ;; Use sasl or not > timeout=100 ;; Timeout on the message stack > status=available;; One of: chat, available, away, > xaway, or dnd > statusmessage = "Connected via Asterisk" ;; Custom status message > > [Jab02] > type = client > serverhost = talk.google.com > username = google-user-nam...@gmail.com/asterisk > secret = > priority = 1 ;; Resource priority > port = 5222 ;;Port to use, defaults to 5222 > usetls = yes ;;TLS is required by talk.google.com, > you'll get a 'socket read error' without > usesasl = yes ;;Use sasl or not > buddy=@gmail.com ;;Manual addition of buddy to list. > buddy=@gmail.com ;;Manual addition of buddy to list. > timeout=100 > status=available > statusmessage = "Connected via Asterisk" > > [Jab03] > > [Jab04] > > and so on. > > Reagrds, > Vladimir > Thanks Vladimir, Will digg up an 1.8 machine and give it a try! afaics the only diference is that i am using a local xmpp server (ejabberd) instead of google, but that should only make things easier i think... Hans -- _ -- 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-user] INTERNAL_OBJ error in asterisk 1.8.13
Hi All, Asterisk Version: 1.8.13.0 CentOs : 6.3 Continues getting this error while submitting cdr record. [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 -- Regards, Chandrakant Solanki -- _ -- 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] Asterisk HangUp not breaking incoming call for caller
On 9/12/2012 12:05 AM, Raj Mathur (??? ?) wrote: > On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote: >> Raj, >> >> I am just confirming it happens here as well. >> >> CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. >> >> Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) >> >> Loadzone = us >> >> The problem started manifesting itself after I switched to 1.8.x from >> 1.6.2.x >> >> Typical scenario: a caller apparently hangs up, dial plan goes into >> voice mail and records a 50 sec message with the CO tones. Then >> something happens and the line finally gets hung up. > I feel this is a different problem: > > *Problem 1*: Asterisk receives incoming call from PSTN. Caller hangs up. > Asterisk doesn't notice hangup and continues in the dialplan. > > *Problem 2*: Asterisk receives incoming call from PSTN. Asterisk > eventually executes HangUp(). Caller does not get hung up. > > Correct me if I'm wrong, but yours seems to be the first, whereas mine > is the second. > > Solved the first one by loading the appropriate zones, loading the > Digium card driver with the correct opermode (which must be one of the > best-kept secrets on the Internet!) and enabling busydetect in Dahdi. > > Stuck at the second one for now. > > Regards, > > -- Raj Raj, You are absolutely right. I am experiencing the *Problem 1*. I never experienced your *Problem 2*. In my opinion my problem is related to the call supervision (reverse polarity or voltage drop) being missed by Asterisk on occasion. I do not use busydetect as I do not want to have false positives. Good luck with your Problem 2 struggle. -Vladimir -- _ -- 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] Asterisk HangUp not breaking incoming call for caller
Raj, Problem 1 is where asterisk times out and line gets free However for problem 2 There is no way that this line frees up, as it depends upon remote side infra of caller, if they calling from pri ckt they possibly could identify our hangup signal, but if they calling from Analog exchange this event is noy reliable and doesnt work 90% of the time. It can als occur if our side of operator pulls up line from Analog exchange instead of Digital, they can claim it to be analog, but i really dont trust MTNL for sure, as they dont know why they r operators too :-) So well we gotta either live wth it or move on to 100% digital comm, i.e. E1 PRI ckts. Mitul On Sep 12, 2012 10:36 AM, "Raj Mathur (राज माथुर)" wrote: > On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote: > > Raj, > > > > I am just confirming it happens here as well. > > > > CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. > > > > Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) > > > > Loadzone = us > > > > The problem started manifesting itself after I switched to 1.8.x from > > 1.6.2.x > > > > Typical scenario: a caller apparently hangs up, dial plan goes into > > voice mail and records a 50 sec message with the CO tones. Then > > something happens and the line finally gets hung up. > > I feel this is a different problem: > > Problem 1: Asterisk receives incoming call from PSTN. Caller hangs up. > Asterisk doesn't notice hangup and continues in the dialplan. > > Problem 2: Asterisk receives incoming call from PSTN. Asterisk > eventually executes HangUp(). Caller does not get hung up. > > Correct me if I'm wrong, but yours seems to be the first, whereas mine > is the second. > > Solved the first one by loading the appropriate zones, loading the > Digium card driver with the correct opermode (which must be one of the > best-kept secrets on the Internet!) and enabling busydetect in Dahdi. > > Stuck at the second one for now. > > Regards, > > -- Raj > -- > Raj Mathur || r...@kandalaya.org || GPG: > http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 > It is the mind that moves || http://schizoid.in || D17F > > -- > _ > -- 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] Asterisk HangUp not breaking incoming call for caller
On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote: > Raj, > > I am just confirming it happens here as well. > > CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. > > Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) > > Loadzone = us > > The problem started manifesting itself after I switched to 1.8.x from > 1.6.2.x > > Typical scenario: a caller apparently hangs up, dial plan goes into > voice mail and records a 50 sec message with the CO tones. Then > something happens and the line finally gets hung up. I feel this is a different problem: Problem 1: Asterisk receives incoming call from PSTN. Caller hangs up. Asterisk doesn't notice hangup and continues in the dialplan. Problem 2: Asterisk receives incoming call from PSTN. Asterisk eventually executes HangUp(). Caller does not get hung up. Correct me if I'm wrong, but yours seems to be the first, whereas mine is the second. Solved the first one by loading the appropriate zones, loading the Digium card driver with the correct opermode (which must be one of the best-kept secrets on the Internet!) and enabling busydetect in Dahdi. Stuck at the second one for now. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] multiple users for jabber.conf
Hans, I did not try 10 or 11 as I run 1.8.15. Following are the related conf files. *gtalk.conf* [General] context = default allowguest = yes ; Required if you want to accept calls from people Not on your contact list. bindaddr= ;; These two settings are very critical for getting externip= ;; gtalk audio with Asterisk server behind NAT disallow=all allow=ulaw [guest] ;;special account for options on guest account disallow=all allow=ulaw context=from-trunk connection= *jabber.conf* [general] debug=no ;;Turn on debugging by default. autoprune=no ;;Auto remove users from buddy list. autoregister=yes ;;Auto register users from buddy list. [Jab01] ;; Label type = client ;; Client or Component connection serverhost = talk.google.com ;; Route to server username = google-user-nam...@gmail.com>/asterisk;; Username with optional resource. secret = ;; Password priority = 1 ;; Resource priority port = 5222 ;; Port to use, defaults to 5222 usetls = yes;; TLS is required by talk.google.com, you'll get a 'socket read error' without usesasl = yes ;; Use sasl or not timeout=100 ;; Timeout on the message stack status=available;; One of: chat, available, away, xaway, or dnd statusmessage = "Connected via Asterisk" ;; Custom status message [Jab02] type = client serverhost = talk.google.com username = google-user-nam...@gmail.com/asterisk secret = priority = 1 ;; Resource priority port = 5222 ;;Port to use, defaults to 5222 usetls = yes ;;TLS is required by talk.google.com, you'll get a 'socket read error' without usesasl = yes ;;Use sasl or not buddy=@gmail.com ;;Manual addition of buddy to list. buddy=@gmail.com ;;Manual addition of buddy to list. timeout=100 status=available statusmessage = "Connected via Asterisk" [Jab03] [Jab04] and so on. Reagrds, Vladimir On 9/11/2012 4:53 PM, Hans Witvliet wrote: > Hi all, > > Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and > 11 version of asterisk. > In each example i got the impression that the asterisk server is > registering on a XMPP server as a single user with the credentials as > specified in jabber.conf. > > Instead of a single xmpp-user, could that also be multiple users? > For instance, for each sip-user an xmpp-user? > > When i skim through most of the examples, the asteriskbox is used for > making an outbound call with the jingle protocol. > > > But how about incoming calls? > I presume you need multiple xmpp-accounts, in order to differentiate > multiple destinations. Not? > > Or to describe it in an other way: If you just do a single > xmpp-registration, how can you become a destination for different > end-users? how about multiple presence-states? > > > [utterly confused] Hans > > -- > _ > -- 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] Asterisk HangUp not breaking incoming call for caller
This has been happening since the asterisk 1.2 days, makes me believe it
has something to do with Analog FXO ckts provided.
Mitul Limbani
On Sep 12, 2012 10:18 AM, "Vladimir Mikhelson" wrote:
> Raj,
>
> I am just confirming it happens here as well.
>
> CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1.
>
> Digium, Inc. Wildcard TDM410 4-port analog card (rev 11)
>
> Loadzone = us
>
> The problem started manifesting itself after I switched to 1.8.x from
> 1.6.2.x
>
> Typical scenario: a caller apparently hangs up, dial plan goes into
> voice mail and records a 50 sec message with the CO tones. Then
> something happens and the line finally gets hung up.
>
> Regards,
> Vladimir
>
>
>
>
> On 9/11/2012 12:08 PM, Raj Mathur (राज माथुर) wrote:
> > Hi,
> >
> > Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai.
> > Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
> >
> > When Asterisk executes HangUp() on an incoming call, the line remains
> > connected for the caller.
> >
> > Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to
> > start looking appreciated.
> >
> > Sample log:
> > [Sep 11 22:06:18] DEBUG[4759]: chan_dahdi.c:11895 do_monitor: Monitor
> doohicky got event Ring Begin on channel 1
> > [Sep 11 22:06:18] DEBUG[4759]: sig_analog.c:3621
> analog_handle_init_event: channel (1) - signaling (5) - event
> (ANALOG_EVENT_RINGBEGIN)
> > [Sep 11 22:06:19] DEBUG[4759]: chan_dahdi.c:11895 do_monitor: Monitor
> doohicky got event Ring/Answered on channel 1
> > [Sep 11 22:06:19] DEBUG[4759]: sig_analog.c:3621
> analog_handle_init_event: channel (1) - signaling (5) - event
> (ANALOG_EVENT_RINGOFFHOOK)
> > [Sep 11 22:06:19] DEBUG[4759]: dsp.c:471 ast_tone_detect_init: Setup
> tone 1100 Hz, 500 ms, block_size=160, hits_required=21
> > [Sep 11 22:06:19] DEBUG[4759]: dsp.c:471 ast_tone_detect_init: Setup
> tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
> > [Sep 11 22:06:19] DEBUG[4759]: dsp.c:1576 ast_dsp_set_busy_pattern: dsp
> busy pattern set to 0,0
> > [Sep 11 22:06:19] DEBUG[4737]: devicestate.c:340 _ast_device_state: No
> provider found, checking channel drivers for DAHDI - 1
> > [Sep 11 22:06:19] DEBUG[5414]: sig_analog.c:1769 __analog_ss_thread:
> __analog_ss_thread 1
> > -- Starting simple switch on 'DAHDI/1-1'
> > [Sep 11 22:06:19] DEBUG[4737]: devicestate.c:458 do_state_change:
> Changing state for DAHDI/1 - state 2 (In use)
> > [Sep 11 22:06:19] DEBUG[4737]: devicestate.c:438 devstate_event: device
> 'DAHDI/1' state '2'
> > [Sep 11 22:06:19] DEBUG[5414]: sig_analog.c:2392 __analog_ss_thread:
> Receiving DTMF cid on channel DAHDI/1-1
> > [Sep 11 22:06:19] DEBUG[4772]: app_queue.c:1487 handle_statechange:
> Device 'DAHDI/1' changed to state '2' (In use) but we don't care because
> they're not a
> > member of any queue.
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3509 analog_exception:
> analog_exception 1
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3603 analog_exception:
> Exception on 16, channel 1
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2660 __analog_handle_event:
> __analog_handle_event 1
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2687 __analog_handle_event:
> Got event ANALOG_EVENT_RINGBEGIN(12) on channel 1 (index 0)
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3509 analog_exception:
> analog_exception 1
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3603 analog_exception:
> Exception on 16, channel 1
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2660 __analog_handle_event:
> __analog_handle_event 1
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2687 __analog_handle_event:
> Got event ANALOG_EVENT_RINGOFFHOOK(2) on channel 1 (index 0)
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3043 __analog_handle_event:
> Ring detected
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2441 __analog_ss_thread: CID
> got string ''
> > [Sep 11 22:06:21] DEBUG[5414]: callerid.c:207 callerid_get_dtmf: No cid
> detected
> > [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2443 __analog_ss_thread: CID
> is '', flags 8
> > [Sep 11 22:06:21] DEBUG[5414]: pbx.c:3239 ast_str_retrieve_variable:
> Result of 'EXTEN' is 's'
> > [Sep 11 22:06:21] DEBUG[4737]: devicestate.c:340 _ast_device_state: No
> provider found, checking channel drivers for DAHDI - 1
> > [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper:
> Launching 'NoOp'
> > [Sep 11 22:06:21] DEBUG[4737]: devicestate.c:458 do_state_change:
> Changing state for DAHDI/1 - state 2 (In use)
> > [Sep 11 22:06:21] DEBUG[4737]: devicestate.c:438 devstate_event: device
> 'DAHDI/1' state '2'
> > -- Executing [s@incoming:1] NoOp("DAHDI/1-1", "Incoming s") in new
> stack
> > [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4058
> pbx_substitute_variables_helper_full: Function result is ''
> > [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper:
> Launching 'Verbose'
> > -- Executing [s@incoming:2] Verbose("DAHDI/1-1", "CID ") in new
> stack
> > CID
> > [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_ext
Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller
Raj,
I am just confirming it happens here as well.
CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1.
Digium, Inc. Wildcard TDM410 4-port analog card (rev 11)
Loadzone = us
The problem started manifesting itself after I switched to 1.8.x from
1.6.2.x
Typical scenario: a caller apparently hangs up, dial plan goes into
voice mail and records a 50 sec message with the CO tones. Then
something happens and the line finally gets hung up.
Regards,
Vladimir
On 9/11/2012 12:08 PM, Raj Mathur (राज माथुर) wrote:
> Hi,
>
> Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai.
> Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
>
> When Asterisk executes HangUp() on an incoming call, the line remains
> connected for the caller.
>
> Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to
> start looking appreciated.
>
> Sample log:
> [Sep 11 22:06:18] DEBUG[4759]: chan_dahdi.c:11895 do_monitor: Monitor
> doohicky got event Ring Begin on channel 1
> [Sep 11 22:06:18] DEBUG[4759]: sig_analog.c:3621 analog_handle_init_event:
> channel (1) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN)
> [Sep 11 22:06:19] DEBUG[4759]: chan_dahdi.c:11895 do_monitor: Monitor
> doohicky got event Ring/Answered on channel 1
> [Sep 11 22:06:19] DEBUG[4759]: sig_analog.c:3621 analog_handle_init_event:
> channel (1) - signaling (5) - event (ANALOG_EVENT_RINGOFFHOOK)
> [Sep 11 22:06:19] DEBUG[4759]: dsp.c:471 ast_tone_detect_init: Setup tone
> 1100 Hz, 500 ms, block_size=160, hits_required=21
> [Sep 11 22:06:19] DEBUG[4759]: dsp.c:471 ast_tone_detect_init: Setup tone
> 2100 Hz, 2600 ms, block_size=160, hits_required=116
> [Sep 11 22:06:19] DEBUG[4759]: dsp.c:1576 ast_dsp_set_busy_pattern: dsp busy
> pattern set to 0,0
> [Sep 11 22:06:19] DEBUG[4737]: devicestate.c:340 _ast_device_state: No
> provider found, checking channel drivers for DAHDI - 1
> [Sep 11 22:06:19] DEBUG[5414]: sig_analog.c:1769 __analog_ss_thread:
> __analog_ss_thread 1
> -- Starting simple switch on 'DAHDI/1-1'
> [Sep 11 22:06:19] DEBUG[4737]: devicestate.c:458 do_state_change: Changing
> state for DAHDI/1 - state 2 (In use)
> [Sep 11 22:06:19] DEBUG[4737]: devicestate.c:438 devstate_event: device
> 'DAHDI/1' state '2'
> [Sep 11 22:06:19] DEBUG[5414]: sig_analog.c:2392 __analog_ss_thread:
> Receiving DTMF cid on channel DAHDI/1-1
> [Sep 11 22:06:19] DEBUG[4772]: app_queue.c:1487 handle_statechange: Device
> 'DAHDI/1' changed to state '2' (In use) but we don't care because they're not
> a
> member of any queue.
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3509 analog_exception:
> analog_exception 1
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3603 analog_exception: Exception
> on 16, channel 1
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2660 __analog_handle_event:
> __analog_handle_event 1
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2687 __analog_handle_event: Got
> event ANALOG_EVENT_RINGBEGIN(12) on channel 1 (index 0)
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3509 analog_exception:
> analog_exception 1
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3603 analog_exception: Exception
> on 16, channel 1
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2660 __analog_handle_event:
> __analog_handle_event 1
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2687 __analog_handle_event: Got
> event ANALOG_EVENT_RINGOFFHOOK(2) on channel 1 (index 0)
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3043 __analog_handle_event: Ring
> detected
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2441 __analog_ss_thread: CID got
> string ''
> [Sep 11 22:06:21] DEBUG[5414]: callerid.c:207 callerid_get_dtmf: No cid
> detected
> [Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2443 __analog_ss_thread: CID is
> '', flags 8
> [Sep 11 22:06:21] DEBUG[5414]: pbx.c:3239 ast_str_retrieve_variable: Result
> of 'EXTEN' is 's'
> [Sep 11 22:06:21] DEBUG[4737]: devicestate.c:340 _ast_device_state: No
> provider found, checking channel drivers for DAHDI - 1
> [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper: Launching
> 'NoOp'
> [Sep 11 22:06:21] DEBUG[4737]: devicestate.c:458 do_state_change: Changing
> state for DAHDI/1 - state 2 (In use)
> [Sep 11 22:06:21] DEBUG[4737]: devicestate.c:438 devstate_event: device
> 'DAHDI/1' state '2'
> -- Executing [s@incoming:1] NoOp("DAHDI/1-1", "Incoming s") in new stack
> [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4058
> pbx_substitute_variables_helper_full: Function result is ''
> [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper: Launching
> 'Verbose'
> -- Executing [s@incoming:2] Verbose("DAHDI/1-1", "CID ") in new stack
> CID
> [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper: Launching
> 'Set'
> -- Executing [s@incoming:3] Set("DAHDI/1-1", "SPYGROUP=queue-01") in new
> stack
> [Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper: Launching
> 'Answer'
> -- Executing [s@incoming:4] Answer("DAHDI/1-1", "") in new stack
> [Sep 11 22:06:21] DEBU
Re: [asterisk-users] asterisk boxes looses registration
Try adding qualify=yes -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Tuesday, September 11, 2012 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk boxes looses registration I have a couple asterisk boxes, running sip between both boxes. 1.4.43 on both. both are installed from source, both have default settings, My config for one box is: [devgeis] type=friend defaultname=devgeis username=devgeis secret=yes disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 host=192.168.1.8 context=panel The other box is the same. There are times when "sip show peers" has Unspecified like: devgeis/devgeis (Unspecified)D a 0Unmonitored So the registration is lost. But a short time later I look again and the registration is present. How can I keep these boxes 100% registered. Thanks, 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] html/js/flash/air SIP clients?
On 7/08/2012, at 7:38 PM, Arstan Jusupov wrote: > Correct me if I'm wrong but phono works with voxeo tropo. Phono can also be used to make SIP calls directly to and from your Asterisk servers via Voxeo. I use it in my CRM package to provide a softphone that logs into call queues and makes/receives calls. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] multiple users for jabber.conf
Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim through most of the examples, the asteriskbox is used for making an outbound call with the jingle protocol. But how about incoming calls? I presume you need multiple xmpp-accounts, in order to differentiate multiple destinations. Not? Or to describe it in an other way: If you just do a single xmpp-registration, how can you become a destination for different end-users? how about multiple presence-states? [utterly confused] Hans -- _ -- 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 boxes looses registration
I have a couple asterisk boxes, running sip between both boxes. 1.4.43 on both. both are installed from source, both have default settings, My config for one box is: [devgeis] type=friend defaultname=devgeis username=devgeis secret=yes disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 host=192.168.1.8 context=panel The other box is the same. There are times when "sip show peers" has Unspecified like: devgeis/devgeis (Unspecified)D a 0Unmonitored So the registration is lost. But a short time later I look again and the registration is present. How can I keep these boxes 100% registered. Thanks, 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
[asterisk-users] Linebreaks in cdr_custom.conf / cdr_sqlite3_custom.conf
In the cdr_custom.conf / cdr_sqlite3_custom.conf configuration files, is it somehow possible to split the enumeration of CDR fields over multiple lines? I get parsing errors when using more then a single line, but a single line is very confusing if one has many CDR fields. So, is there something like "\" on the shell? Thanks and best regards Stefan -- _ -- 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 HangUp not breaking incoming call for caller
Hi,
Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai.
Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
When Asterisk executes HangUp() on an incoming call, the line remains
connected for the caller.
Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to
start looking appreciated.
Sample log:
[Sep 11 22:06:18] DEBUG[4759]: chan_dahdi.c:11895 do_monitor: Monitor doohicky
got event Ring Begin on channel 1
[Sep 11 22:06:18] DEBUG[4759]: sig_analog.c:3621 analog_handle_init_event:
channel (1) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN)
[Sep 11 22:06:19] DEBUG[4759]: chan_dahdi.c:11895 do_monitor: Monitor doohicky
got event Ring/Answered on channel 1
[Sep 11 22:06:19] DEBUG[4759]: sig_analog.c:3621 analog_handle_init_event:
channel (1) - signaling (5) - event (ANALOG_EVENT_RINGOFFHOOK)
[Sep 11 22:06:19] DEBUG[4759]: dsp.c:471 ast_tone_detect_init: Setup tone 1100
Hz, 500 ms, block_size=160, hits_required=21
[Sep 11 22:06:19] DEBUG[4759]: dsp.c:471 ast_tone_detect_init: Setup tone 2100
Hz, 2600 ms, block_size=160, hits_required=116
[Sep 11 22:06:19] DEBUG[4759]: dsp.c:1576 ast_dsp_set_busy_pattern: dsp busy
pattern set to 0,0
[Sep 11 22:06:19] DEBUG[4737]: devicestate.c:340 _ast_device_state: No provider
found, checking channel drivers for DAHDI - 1
[Sep 11 22:06:19] DEBUG[5414]: sig_analog.c:1769 __analog_ss_thread:
__analog_ss_thread 1
-- Starting simple switch on 'DAHDI/1-1'
[Sep 11 22:06:19] DEBUG[4737]: devicestate.c:458 do_state_change: Changing
state for DAHDI/1 - state 2 (In use)
[Sep 11 22:06:19] DEBUG[4737]: devicestate.c:438 devstate_event: device
'DAHDI/1' state '2'
[Sep 11 22:06:19] DEBUG[5414]: sig_analog.c:2392 __analog_ss_thread: Receiving
DTMF cid on channel DAHDI/1-1
[Sep 11 22:06:19] DEBUG[4772]: app_queue.c:1487 handle_statechange: Device
'DAHDI/1' changed to state '2' (In use) but we don't care because they're not a
member of any queue.
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3509 analog_exception:
analog_exception 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3603 analog_exception: Exception on
16, channel 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2660 __analog_handle_event:
__analog_handle_event 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2687 __analog_handle_event: Got
event ANALOG_EVENT_RINGBEGIN(12) on channel 1 (index 0)
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3509 analog_exception:
analog_exception 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3603 analog_exception: Exception on
16, channel 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2660 __analog_handle_event:
__analog_handle_event 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2687 __analog_handle_event: Got
event ANALOG_EVENT_RINGOFFHOOK(2) on channel 1 (index 0)
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3043 __analog_handle_event: Ring
detected
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2441 __analog_ss_thread: CID got
string ''
[Sep 11 22:06:21] DEBUG[5414]: callerid.c:207 callerid_get_dtmf: No cid detected
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2443 __analog_ss_thread: CID is '',
flags 8
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:3239 ast_str_retrieve_variable: Result of
'EXTEN' is 's'
[Sep 11 22:06:21] DEBUG[4737]: devicestate.c:340 _ast_device_state: No provider
found, checking channel drivers for DAHDI - 1
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper: Launching 'NoOp'
[Sep 11 22:06:21] DEBUG[4737]: devicestate.c:458 do_state_change: Changing
state for DAHDI/1 - state 2 (In use)
[Sep 11 22:06:21] DEBUG[4737]: devicestate.c:438 devstate_event: device
'DAHDI/1' state '2'
-- Executing [s@incoming:1] NoOp("DAHDI/1-1", "Incoming s") in new stack
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4058 pbx_substitute_variables_helper_full:
Function result is ''
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper: Launching
'Verbose'
-- Executing [s@incoming:2] Verbose("DAHDI/1-1", "CID ") in new stack
CID
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper: Launching 'Set'
-- Executing [s@incoming:3] Set("DAHDI/1-1", "SPYGROUP=queue-01") in new
stack
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper: Launching
'Answer'
-- Executing [s@incoming:4] Answer("DAHDI/1-1", "") in new stack
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:1499 analog_answer: analog_answer 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:1530 analog_answer: Took DAHDI/1-1
off hook
[Sep 11 22:06:21] DEBUG[5414]: chan_dahdi.c:4927 dahdi_enable_ec: Enabled echo
cancellation on channel 1
[Sep 11 22:06:21] DEBUG[4737]: devicestate.c:340 _ast_device_state: No provider
found, checking channel drivers for DAHDI - 1
[Sep 11 22:06:21] DEBUG[5414]: chan_dahdi.c:4946 dahdi_train_ec: No echo
training requested
[Sep 11 22:06:21] DEBUG[5414]: chan_dahdi.c:9403 dahdi_indicate: Requested
indication -1 on channel DAHDI/1-1
[Sep 11 22:06:21] DEBUG[4772]: app_queue.c:1487 handle_statechange: Device
'DAHDI/1' changed to state '2'
[asterisk-users] Asterisk HangUp not breaking incoming call for caller
Hi,
Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai.
Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
When Asterisk executes HangUp() on an incoming call, the line remains
connected for the caller.
Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to
start looking appreciated.
Sample log:
[Sep 11 22:06:18] DEBUG[4759]: chan_dahdi.c:11895 do_monitor: Monitor
doohicky got event Ring Begin on channel 1
[Sep 11 22:06:18] DEBUG[4759]: sig_analog.c:3621
analog_handle_init_event: channel (1) - signaling (5) - event
(ANALOG_EVENT_RINGBEGIN)
[Sep 11 22:06:19] DEBUG[4759]: chan_dahdi.c:11895 do_monitor: Monitor
doohicky got event Ring/Answered on channel 1
[Sep 11 22:06:19] DEBUG[4759]: sig_analog.c:3621
analog_handle_init_event: channel (1) - signaling (5) - event
(ANALOG_EVENT_RINGOFFHOOK)
[Sep 11 22:06:19] DEBUG[4759]: dsp.c:471 ast_tone_detect_init: Setup
tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Sep 11 22:06:19] DEBUG[4759]: dsp.c:471 ast_tone_detect_init: Setup
tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Sep 11 22:06:19] DEBUG[4759]: dsp.c:1576 ast_dsp_set_busy_pattern: dsp
busy pattern set to 0,0
[Sep 11 22:06:19] DEBUG[4737]: devicestate.c:340 _ast_device_state: No
provider found, checking channel drivers for DAHDI - 1
[Sep 11 22:06:19] DEBUG[5414]: sig_analog.c:1769 __analog_ss_thread:
__analog_ss_thread 1
-- Starting simple switch on 'DAHDI/1-1'
[Sep 11 22:06:19] DEBUG[4737]: devicestate.c:458 do_state_change:
Changing state for DAHDI/1 - state 2 (In use)
[Sep 11 22:06:19] DEBUG[4737]: devicestate.c:438 devstate_event: device
'DAHDI/1' state '2'
[Sep 11 22:06:19] DEBUG[5414]: sig_analog.c:2392 __analog_ss_thread:
Receiving DTMF cid on channel DAHDI/1-1
[Sep 11 22:06:19] DEBUG[4772]: app_queue.c:1487 handle_statechange:
Device 'DAHDI/1' changed to state '2' (In use) but we don't care because
they're not a member of any queue.
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3509 analog_exception:
analog_exception 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3603 analog_exception:
Exception on 16, channel 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2660 __analog_handle_event:
__analog_handle_event 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2687 __analog_handle_event:
Got event ANALOG_EVENT_RINGBEGIN(12) on channel 1 (index 0)
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3509 analog_exception:
analog_exception 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3603 analog_exception:
Exception on 16, channel 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2660 __analog_handle_event:
__analog_handle_event 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2687 __analog_handle_event:
Got event ANALOG_EVENT_RINGOFFHOOK(2) on channel 1 (index 0)
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:3043 __analog_handle_event:
Ring detected
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2441 __analog_ss_thread: CID
got string ''
[Sep 11 22:06:21] DEBUG[5414]: callerid.c:207 callerid_get_dtmf: No cid
detected
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:2443 __analog_ss_thread: CID
is '', flags 8
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:3239 ast_str_retrieve_variable:
Result of 'EXTEN' is 's'
[Sep 11 22:06:21] DEBUG[4737]: devicestate.c:340 _ast_device_state: No
provider found, checking channel drivers for DAHDI - 1
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper:
Launching 'NoOp'
[Sep 11 22:06:21] DEBUG[4737]: devicestate.c:458 do_state_change:
Changing state for DAHDI/1 - state 2 (In use)
[Sep 11 22:06:21] DEBUG[4737]: devicestate.c:438 devstate_event: device
'DAHDI/1' state '2'
-- Executing [s@incoming:1] NoOp("DAHDI/1-1", "Incoming s") in new
stack
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4058
pbx_substitute_variables_helper_full: Function result is ''
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper:
Launching 'Verbose'
-- Executing [s@incoming:2] Verbose("DAHDI/1-1", "CID ") in new
stack
CID
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper:
Launching 'Set'
-- Executing [s@incoming:3] Set("DAHDI/1-1", "SPYGROUP=queue-01") in
new stack
[Sep 11 22:06:21] DEBUG[5414]: pbx.c:4230 pbx_extension_helper:
Launching 'Answer'
-- Executing [s@incoming:4] Answer("DAHDI/1-1", "") in new stack
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:1499 analog_answer:
analog_answer 1
[Sep 11 22:06:21] DEBUG[5414]: sig_analog.c:1530 analog_answer: Took
DAHDI/1-1 off hook
[Sep 11 22:06:21] DEBUG[5414]: chan_dahdi.c:4927 dahdi_enable_ec:
Enabled echo cancellation on channel 1
[Sep 11 22:06:21] DEBUG[4737]: devicestate.c:340 _ast_device_state: No
provider found, checking channel drivers for DAHDI - 1
[Sep 11 22:06:21] DEBUG[5414]: chan_dahdi.c:4946 dahdi_train_ec: No echo
training requested
[Sep 11 22:06:21] DEBUG[5414]: chan_dahdi.c:9403 dahdi_indicate:
Requested indication -1 on channel DAHDI/1-1
[Sep 11 22:06:21] DEBUG[4772]: app_queue.c:1487 handle_statechange:
Device 'DAHDI/1' changed to st
[asterisk-users] codec priorities
Hello, I am about to start playing with wideband codecs in our lab, and was hoping to get some clarification on a few things. To date I've pretty much forced the use of G.711 on all legs of all calls, and life has been grand. Now we are distributing phones with G.722 and speex capability, and I would like to use those codecs if it can somehow be determined that the end legs have the capability. There may be several asterisk servers in between, however. My first question is - is the order in which I specify "allow" in the sip.conf entry the priority order asterisk will negotiate with the phone? Assuming the above is true, will it *always* negotiate G.722 if I make it first, even if the call, within the same asterisk server, ends up going out a dahdi trunk? How about if, again in the same asterisk server, it ends up going to a SIP endpoint that *cannot* do G.722? What about a different scenario involving two asterisk servers, SIP trunked together with the ability to do G.722 or G.711. If a SIP endpoint on one asterisk server can only do G.711, will the the call to and endpoint on the other side (lets say that can also only do G.711) get transcoded "up" to G.722 to cross the trunk, then back "down" to G.711 on the other side? I vaguely recall a conversation about this some time ago but couldn't find it in the archives. I'm afraid I know the answer - that asterisk doesn't try to find the most efficient codec for the call and that if I decide to let my fancy new phones do G.722 or speex, then transcoding will often be involved. I'd love to understand the complications in this better... seems like it would be a fantastic feature to be able to negotiate end to end and pick the most efficient codec. Cheers, j -- _ -- 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] Async AGI
Hi David, For sure I will ask them but I think Asterisk should be able to handle this case because it doesn't matter if it is adhearsion or something else. If it is not present there is no way to get answer. So there must be some way to go to another priority in the dialplan. @Danny - this is the normal behaviour of async agi Regards, Pavel >Pavel, >This may be due to the asynchronous nature of Async AGI. The AGI(agi:async) command will continue to run until the channel is hung up, the 'asyncagi break' command is sent over AMI, or some sort of fatal error is >encountered. >When a call comes in, the AGI(agi:async) command sill send out a start event over AMI. But if Adhearsion isn't there to hear it, it's also not there to tell Asterisk what to do with the channel. So it will sit there, waiting for >a break or hangup. >I'm sure the Adhearsion guys have run into this sort of thing before. You may want to ask on their forum[1] for how they handle the situation. > [1]: https://groups.google.com/forum/?fromgroups=#!forum/adhearsion -- _ -- 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
