* You are viewing the archive for July 6th, 2012

Sip.conf And Binaddr Issue

Hi there.

i am seriously stuck with an asterisk and sip problem.

the following sip.conf works with respect to some_peer:

[general]
bindaddr = x.y.z.w nat = no

[some_peer]
type=peer host=somehost secret=somesecret some other unrelated options

here x.y.z.w is the ip address of the interface pointing to the network containing somehost. more precisely its the address of tun0 and route -n prints
Destination Gateway Genmask Flags Metric Ref Use Iface
[..]
x.y.z.0 0.0.0.0 255.255.255.0 U 0 0 0 tun0
[..]

here ‘it works’ implies that i have to change and reload sip.conf after ifup tun0, or anything that forces tun0 to go down, like my dsl provider. also, the bindaddr line is suboptimal for the other peers…

the same thing — without the bindaddr part — doesnt work. more precisely it almost works. its just incoming sound that doesnt. this must have something to do with how asterisk picks up interface addresses and communicates them to the peer in question. inspecting the packages sent to somehost, gave me the impression that asterisk uses the ip adress of ppp0 (a dsl modem) instead.

how am i supposed to tell asterisk to use tun0 as the interface for
[some_peer] so i can remove the bindaddr line? i’ve found many nat-related options in the manual, but there is no nat involved here.
also, i couldnt find anything similar to “iface=tun0″, although the sip dialogue apparently relies on ip adresses and routing.

this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course i’m going to switch to whatever you might suggest.

regards and thanks felix

Trixbox Or FreePBX?

Hi All;

Based on what I have to use Trixbox or FreePBX?

Can someone advise?

Regards
Bilal

Can I Install Asterisk Normally And Then Installing The GUI

OK, is there asterisk-gui that differs that freepbx? Or Freepbx is a GUI for asterisk?

In other words, if I have asterisk and I need to add for it a GUI, is there asterisk-gui which is differs than freepbx or it is the same?

Regards
Bilal

———–

Maximum Concurrent Calls Using Call Files

I am planning on building a testing module which would spawn about 500
calls in order to test the performance of the network by transferring audio/speech files to end points at that juncture.Is it possible to spawn as many concurrent calls (or nearly concurrent calls) using just call files.Is there a limit as to the maximum number that could be spawned.?
I tried doing this for about 20 calls and found that there is autofallthrough after a point of time.Is this a problem with my dialplan or is it because of the call files (i also get a warning which states that the ast_queue_frame:Exceptionally long queue length)

Thanks,
Sathiish

DAHDI DTMF Problem?

I have an Asterisk server configured to run as voicemail with a T1 and SMDI. It has 1.6.1.6 (dahdi 2.1.0.4) and CentOS 5.6 and has worked great for a few years. I am configuring a new server with Asterisk 1.8.13 (dahdi 2.6.1) on
CentOS 5.8

The problem I am having appears to be related to DTMF detection. When the test phone number is called (2704083000) Asterisk only receives a portion of the dialed number. It varies as to what numbers are detected. Sometimes it sees a single digit, sometimes 3 or 4 of the digits of the dialed number.

When I compare this to the old server the debug below is similar but there isn’t any mention of the “sig_analog.c” lines shown below.

I am told the T1′s on the old server and the new server are configured the same. I can make outgoing calls on the T1 from Asterisk.

Can someone give me a clue as to what could be causing this?


Bill Dunn



[Jul 6 10:28:40] DEBUG[15045]: chan_dahdi.c:11895 do_monitor: Monitor doohicky got event Ring/Answered on channel 1
[Jul 6 10:28:40] DEBUG[15045]: sig_analog.c:3621 analog_handle_init_event:
channel (1) – signaling (7) – event (ANALOG_EVENT_RINGOFFHOOK)
[Jul 6 10:28:40] DEBUG[15045]: dsp.c:471 ast_tone_detect_init: Setup tone
1100 Hz, 500 ms, block_size0, hits_required!
[Jul 6 10:28:40] DEBUG[15045]: dsp.c:471 ast_tone_detect_init: Setup tone
2100 Hz, 2600 ms, block_size0, hits_required6
[Jul 6 10:28:40] DEBUG[15036]: devicestate.c:458 do_state_change: Changing state for DAHDI/1 – state 2 (In use)
[Jul 6 10:28:40] DEBUG[15081]: sig_analog.c:1769 __analog_ss_thread:
__analog_ss_thread 1
— Starting simple switch on ‘DAHDI/1-1′
[Jul 6 10:28:40] DEBUG[15036]: devicestate.c:438 devstate_event: device
‘DAHDI/1′ state ’2′
[Jul 6 10:28:40] DEBUG[15053]: 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.
[Jul 6 10:28:40] DEBUG[15081]: sig_analog.c:1602 analog_handle_dtmf: Begin
DTMF digit: 0×32 ’2′ on DAHDI/1-1
[Jul 6 10:28:40] DEBUG[15081]: chan_dahdi.c:2026 my_handle_dtmf: Begin DTMF
digit: 0×32 ’2′ on DAHDI/1-1
[Jul 6 10:28:40] DEBUG[15081]: sig_analog.c:1602 analog_handle_dtmf: End
DTMF digit: 0×32 ’2′ on DAHDI/1-1
[Jul 6 10:28:40] DEBUG[15081]: chan_dahdi.c:2026 my_handle_dtmf: End DTMF
digit: 0×32 ’2′ on DAHDI/1-1
[Jul 6 10:28:40] DEBUG[15081]: sig_analog.c:1602 analog_handle_dtmf: Begin
DTMF digit: 0×37 ’7′ on DAHDI/1-1
[Jul 6 10:28:40] DEBUG[15081]: chan_dahdi.c:2026 my_handle_dtmf: Begin DTMF
digit: 0×37 ’7′ on DAHDI/1-1
[Jul 6 10:28:40] DEBUG[15081]: sig_analog.c:1602 analog_handle_dtmf: End
DTMF digit: 0×37 ’7′ on DAHDI/1-1
[Jul 6 10:28:40] DEBUG[15081]: chan_dahdi.c:2026 my_handle_dtmf: End DTMF
digit: 0×37 ’7′ on DAHDI/1-1
[Jul 6 10:28:41] DEBUG[15081]: sig_analog.c:1602 analog_handle_dtmf: Begin
DTMF digit: 0×38 ’8′ on DAHDI/1-1
[Jul 6 10:28:41] DEBUG[15081]: chan_dahdi.c:2026 my_handle_dtmf: Begin DTMF
digit: 0×38 ’8′ on DAHDI/1-1
[Jul 6 10:28:41] DEBUG[15081]: sig_analog.c:1602 analog_handle_dtmf: End
DTMF digit: 0×38 ’8′ on DAHDI/1-1
[Jul 6 10:28:41] DEBUG[15081]: chan_dahdi.c:2026 my_handle_dtmf: End DTMF
digit: 0×38 ’8′ on DAHDI/1-1
[Jul 6 10:28:41] DEBUG[15081]: chan_dahdi.c:4927 dahdi_enable_ec: Enabled echo cancellation on channel 1
— Unknown extension ’278′ in context ‘default’ requested




Here is another call.

[Jul 6 10:48:34] DEBUG[15138]: chan_dahdi.c:11895 do_monitor: Monitor doohicky got event Ring/Answered on channel 1
[Jul 6 10:48:34] DEBUG[15138]: sig_analog.c:3621 analog_handle_init_event:
channel (1) – signaling (7) – event (ANALOG_EVENT_RINGOFFHOOK)
[Jul 6 10:48:34] DEBUG[15138]: dsp.c:471 ast_tone_detect_init: Setup tone
1100 Hz, 500 ms, block_size0, hits_required!
[Jul 6 10:48:34] DEBUG[15138]: dsp.c:471 ast_tone_detect_init: Setup tone
2100 Hz, 2600 ms, block_size0, hits_required6
[Jul 6 10:48:34] DEBUG[15150]: sig_analog.c:1769 __analog_ss_thread:
__analog_ss_thread 1
[Jul 6 10:48:34] DEBUG[15129]: devicestate.c:458 do_state_change: Changing state for DAHDI/1 – state 2 (In use)
[Jul 6 10:48:34] DEBUG[15129]: devicestate.c:438 devstate_event: device
‘DAHDI/1′ state ’2′
— Starting simple switch on ‘DAHDI/1-1′
[Jul 6 10:48:34] DEBUG[15145]: 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.
[Jul 6 10:48:34] DEBUG[15150]: sig_analog.c:1602 analog_handle_dtmf: Begin
DTMF digit: 0×37 ’7′ on DAHDI/1-1
[Jul 6 10:48:34] DEBUG[15150]: chan_dahdi.c:2026 my_handle_dtmf: Begin DTMF
digit: 0×37 ’7′ on DAHDI/1-1
[Jul 6 10:48:35] DEBUG[15150]: sig_analog.c:1602 analog_handle_dtmf: End
DTMF digit: 0×37 ’7′ on DAHDI/1-1
[Jul 6 10:48:35] DEBUG[15150]: chan_dahdi.c:2026 my_handle_dtmf: End DTMF
digit: 0×37 ’7′ on DAHDI/1-1
[Jul 6 10:48:35] DEBUG[15150]: chan_dahdi.c:4927 dahdi_enable_ec: Enabled echo cancellation on channel 1
— Unknown extension ’7′ in context ‘default’ requested

Asterisk Trying To Call A Queue With No Members

Hi,

I am trying to configure some static queues in asterisk, it’s almost working, the problem is that asterisk is not verifying if the queue has logged members. For example, if I create queue called test, which has no members logged in, and try to place a call using Queue(test) I get into the queue, even if all phones are turned off, I tried to verify it with the QUEUE_MEMBER function, using the “ready” parameter, it shows me that all members are logged in. Here is my queues.conf:

[general]
persistentmembers = yes monitor-type = MixMonitor

[default-queue](!)
musicclass