[asterisk-users] Inbound configuration

[chima s]

Hi All,

I am trying to configure asterisk only for inbound calls and using
freepbx to configure it

I have created a inbound route and tried to call the number, i am
getting the following error message.

Extension '' in context 'default' from '' does not exist.  Rejecting call on channel 0/1, span 2

Can any one help me figure out this issue.

Regards
Chima

-- 
_
-- 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] dnd not working correctly

[Alyed]

I'm not an Amportal expert so all I can say from:

> -- Executing [...@from-internal:8] Playback("SIP/117-01f6",
"do-not-disturb&activated") in new stack
> -- Executing [...@from-internal:9] Macro("SIP/117-01f6",
"hangupcall,") in new stack

is that Asterisk is playing the "do-not-disturb&activated" file (apparently
without errors) and then the next instruction is to hangup the call, hence
Asterisk hangs it up.

Just to be sure play this sound file independently.

Sorry but other than this there's little I can do, maybe someone else has
experience with this.

Alyed


2010/3/29 Ott Rose 

>
> i posted this on the freepbx site. here is the response
>
>
> "from the trace, everything is working. Check your asterisk log for file
> errors playing back the audio, could be your sound files are not installed
> or messed up."
>
>
>
> so i checked /etc/log/asterisk/full
>
> and in vi full i did /error   and  /117 (my ext) and /activate didn't
> really find anything
>
> i didn't see anything but i might be over looking it. I did grep error full
> and it returned some errors but not related to dnd as far as i can tell. is
> there some place else to look, a better way to search that file, or can
> someone tell me what i am looking for?
>
>
>
>
> --
> Date: Fri, 26 Mar 2010 18:34:46 -0600
> From: al...@vivoxie.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] dnd not working correctly
>
> Seems like an Amportal configration problem not and Asterisk issue. Maybe
> you should try in one of the FreePBX users list.
>
> Alyed
>
>
>
> 2010/3/26 Ott Rose 
>
>  i have posted this question couple of times and never really got any hits
> i wasn't able to provide any debug info
>
> Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid =
> 3309)
> Verbosity is at least 4
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>   == Using SIP VRTP TOS bits 136
>   == Using SIP VRTP CoS mark 6
>   == Extension Changed 117[ext-local] new state InUse for Notify User 102
>   == Extension Changed 117[ext-local] new state InUse for Notify User 103
>   == Extension Changed 117[ext-local] new state InUse for Notify User 114
> -- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in
> new stack
> -- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in
> new stack
> -- Executing [...@from-internal:3] Macro("SIP/117-01f6",
> "user-callerid,") in new stack
> -- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6",
> "1?Set(REALCALLERIDNUM=117)") in new stack
> -- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6",
> "AMPUSERCIDNAME=My Name") in new stack
> -- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6",
> "AMPUSERCID=117") in new stack
> -- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6",
> "CALLERID(all)="My Name" <117>") in new stack
> -- Executing [...@macro-user-callerid:9] GotoIf("SIP/117-01f6",
> "0?continue") in new stack
> -- Executing [...@macro-user-callerid:10] Set("SIP/117-01f6",
> "__TTL=64") in new stack
> -- Executing [...@macro-user-callerid:11] GotoIf("SIP/117-01f6",
> "1?continue") in new stack
> -- Goto (macro-user-callerid,s,18)
> -- Executing [...@macro-user-callerid:18] NoOp("SIP/117-01f6",
> "Using CallerID "My Name" <117>") in new stack
> -- Executing [...@from-internal:4] GotoIf("SIP/117-01f6",
> "1?activate:deactivate") in new stack
> -- Goto (from-internal,*76,5)
> -- Executing [...@from-internal:5] Set("SIP/117-01f6",
> "DB(DND/117)=YES") in new stack
> -- Executing [...@from-internal:6] Set("SIP/117-01f6",
> "STATE=BUSY") in new stack
> -- Executing [...@from-internal:7] Gosub("SIP/117-01f6",
> "app-dnd-toggle,sstate,1") in new stack
> -- Executing [sst...@app-dnd-toggle:1] Set("SIP/117-01f6",
> "DEVICE_STATE(Custom:DND117)=BUSY") in new stack
> -- Executing [sst...@app-dnd-toggle:2] Set("SIP/117-01f6",
> "DEVICES=117") in new stack
> -- Executing [sst...@app-dnd-toggle:3] GotoIf("SIP/117-01f6",
> "0?return") in new stack
>   == Extension Changed 117[ext-local] new state Busy for Notify User 102
> -- Executing [sst...@app-dnd-toggle:4] Set("SIP/117-01f6",
> "LOOPCNT=1") in new stack
> -- Executing [sst...@app-dnd-toggle:5] Set("SIP/117-01f6",
> "ITER=1") in new stack
> -- Executing [sst...@app-dnd-toggle:6] Set("SIP/117-01f6",
> "DEVICE_STATE(Custom:DEVDND117)=BUSY") in new stack
>   == Extension Changed 117[ext-l

Re: [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)

[James Lamanna]

On Mon, Mar 29, 2010 at 9:23 PM, James Lamanna  wrote:
> On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson  wrote:
>> Dell server by any chance?
>> I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
>> several other people having issues with digium cards in dell servers as
>> well.
>> I've actually done something similar to what you have done - isolated the
>> TE220B onto its own IRQ and set processor affinity for all the IRQs to
>> particular cores... so far I haven't had kernel pancs since doing this, but
>> its still a little too early to say if it has fixed the issue 100% or not.
>
> Interesting. It is actually a Dell SC1425 - Dual, dual-core Xeon Processors.
> I'm hopefully going to be able to stress test this machine to see if I
> can make it panic again with the PRI card IRQ isolated to CPU0. If so,
> I'll see if it does the same thing on the other cores...

As a data point, I tried stress testing this box this evening. Moving
the interrupt to each core, the results did not change.
The test was as follows:
Originate() a call that goes out to the PSTN and comes back in. Both
sides used Milliwatt() to make sure audio flowed both ways.
I generated 30 calls this way (to use 60 PRI channels), however, I was
never able to simultaneously keep 60 channels alive. During the test,
there would always be a D-Channel down/up, which would drop all calls
on that PRI span.
I do not know if this is a Zaptel issue (1.4.12), PRI card issue
(TE401P first-gen), or something more subtle...

Any help would be appricated!

Thanks.

-- James

>
> -- James
>
>> --
>> Matt
>>
>> On Mon, Mar 29, 2010 at 8:30 PM, James Lamanna  wrote:
>>>
>>> Hi,
>>> I'm trying to figure out the cause of a soft lockup I experienced:
>>>
>>> Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s!
>>> [asterisk:32029]
>>> Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm:             asterisk
>>> Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[] CPU: 0
>>> Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c
>>> Mar 29 09:38:24 pstn1 kernel:  EFLAGS: 0286    Tainted: GF
>>> (2.6.18-128.1.10.el5 #1)
>>> Mar 29 09:38:24 pstn1 kernel: EAX: 0029 EBX: f7ff9380 ECX:
>>> f7fff880 EDX: c11ff9a0
>>> Mar 29 09:38:24 pstn1 kernel: ESI: 0286 EDI: cffcda00 EBP:
>>> e5e10c80 DS: 007b ES: 007b
>>> Mar 29 09:38:24 pstn1 kernel: CR0: 80050033 CR2: b7ce39e0 CR3:
>>> 0f911000 CR4: 06d0
>>> Mar 29 09:38:24 pstn1 kernel:  [] kfree_skbmem+0x8/0x61
>>> Mar 29 09:38:24 pstn1 kernel:  [] __udp_queue_rcv_skb+0x4a/0x51
>>> Mar 29 09:38:24 pstn1 kernel:  [] release_sock+0x44/0x91
>>> Mar 29 09:38:24 pstn1 kernel:  [] udp_sendmsg+0x44e/0x514
>>> Mar 29 09:38:24 pstn1 kernel:  [] inet_sendmsg+0x35/0x3f
>>> Mar 29 09:38:24 pstn1 kernel:  [] sock_sendmsg+0xce/0xe8
>>> Mar 29 09:38:24 pstn1 kernel:  []
>>> autoremove_wake_function+0x0/0x2d
>>> Mar 29 09:38:24 pstn1 kernel:  [] copy_from_user+0x17/0x5d
>>> Mar 29 09:38:24 pstn1 kernel:  [] copy_to_user+0x31/0x48
>>> Mar 29 09:38:24 pstn1 kernel:  [] zt_chan_read+0x1e0/0x20b
>>> [zaptel]
>>> Mar 29 09:38:24 pstn1 kernel:  [] copy_from_user+0x31/0x5d
>>> Mar 29 09:38:24 pstn1 kernel:  [] sys_sendto+0x116/0x140
>>> Mar 29 09:38:24 pstn1 kernel:  [] flush_tlb_page+0x74/0x77
>>> Mar 29 09:38:24 pstn1 kernel:  [] do_wp_page+0x3bf/0x40a
>>> Mar 29 09:38:24 pstn1 kernel:  [] current_fs_time+0x4a/0x55
>>> Mar 29 09:38:24 pstn1 kernel:  [] touch_atime+0x60/0x91
>>> Mar 29 09:38:24 pstn1 kernel:  [] pipe_readv+0x315/0x321
>>> Mar 29 09:38:24 pstn1 kernel:  [] sys_socketcall+0x106/0x19e
>>> Mar 29 09:38:24 pstn1 kernel:  [] syscall_call+0x7/0xb
>>> Mar 29 09:38:24 pstn1 kernel:  ===
>>>
>>>
>>> This occurred during a "high load" period (52 calls across 3 PRI spans).
>>>
>>> A couple days ago I moved the interrupts for my PRI card to CPU0 from
>>> CPU3, because CPU3 was handling everything else:
>>>           CPU0       CPU1       CPU2       CPU3
>>>  0:        306          0          0 3684057379    IO-APIC-edge  timer
>>>  1:          0          0          0      13468    IO-APIC-edge  i8042
>>>  8:          0          0          0          3    IO-APIC-edge  rtc
>>>  9:          0          0          0          0   IO-APIC-level  acpi
>>>  12:          0          0          0          4    IO-APIC-edge  i8042
>>> 169:          0          0          0          0   IO-APIC-level
>>>  uhci_hcd:usb2
>>> 177:          0          0          0   18392593   IO-APIC-level  ata_piix
>>> 185:          0          0          0          1   IO-APIC-level
>>>  ehci_hcd:usb1
>>> 193:          0          0          0          0   IO-APIC-level
>>>  uhci_hcd:usb3
>>> 201:          0          0          0 2090021759   IO-APIC-level  eth0
>>> 209:  149621223          0          0 3534419461   IO-APIC-level  wct4xxp
>>>
>>>
>>> (The CPU3 number for wct4xxp is not increasing any more).
>>>
>>> What is the interrupt distribution of other people's systems?
>>> Before I made this change I was having a

Re: [asterisk-users] Asynchronous play music

[Steve Edwards]

On Tue, 30 Mar 2010, Pham Quy wrote:

> Is there anyway to catch DTMF keypress while a music file is playing 
> without stop the music?

Have you tried externivr(). I've never used it, but it looks interesting.

-- 
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] a2billing wont pass the number

[Juan E. Rodríguez]

When you say 'a2billing' won't pass the number, you mean you are calling to an 
IVR or something like that.

And when did you dial you destination number twice???

Saludos,
Juan E. Rodríguez


-Original Message-
From: Nathanial Allan 
Date: Tue, 30 Mar 2010 13:08:24 
To: 
Subject: [asterisk-users] a2billing wont pass the number

I am running into an issue with A2Billing. I will explain  first of all that 
everything else works! the system is 90% complete its just this one small 
problem I am running into. 

So my problem is that when I place a call, 
1. I dial my number that I want and A2Billing gets activated 
2. it asks for my pin, upon successful entry of my pin A2Billing then
3. prompts me for my phone number then 
4. The call goes out (and actually connects for the record)

So I am entering my destination phone number twice which is not the worst thing 
that can happen, though it is a little annoying

Any light that you can shine on this problem would be greatly appreciated as I 
have been working on it for too long now and I want to get a product!


Thank You

NallaN
-- 
_
-- 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] How are your PRI interrupts balanced? (+ Soft lockup BUG)

[James Lamanna]

On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson  wrote:
> Dell server by any chance?
> I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
> several other people having issues with digium cards in dell servers as
> well.
> I've actually done something similar to what you have done - isolated the
> TE220B onto its own IRQ and set processor affinity for all the IRQs to
> particular cores... so far I haven't had kernel pancs since doing this, but
> its still a little too early to say if it has fixed the issue 100% or not.

Interesting. It is actually a Dell SC1425 - Dual, dual-core Xeon Processors.
I'm hopefully going to be able to stress test this machine to see if I
can make it panic again with the PRI card IRQ isolated to CPU0. If so,
I'll see if it does the same thing on the other cores...

-- James

> --
> Matt
>
> On Mon, Mar 29, 2010 at 8:30 PM, James Lamanna  wrote:
>>
>> Hi,
>> I'm trying to figure out the cause of a soft lockup I experienced:
>>
>> Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s!
>> [asterisk:32029]
>> Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm:             asterisk
>> Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[] CPU: 0
>> Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c
>> Mar 29 09:38:24 pstn1 kernel:  EFLAGS: 0286    Tainted: GF
>> (2.6.18-128.1.10.el5 #1)
>> Mar 29 09:38:24 pstn1 kernel: EAX: 0029 EBX: f7ff9380 ECX:
>> f7fff880 EDX: c11ff9a0
>> Mar 29 09:38:24 pstn1 kernel: ESI: 0286 EDI: cffcda00 EBP:
>> e5e10c80 DS: 007b ES: 007b
>> Mar 29 09:38:24 pstn1 kernel: CR0: 80050033 CR2: b7ce39e0 CR3:
>> 0f911000 CR4: 06d0
>> Mar 29 09:38:24 pstn1 kernel:  [] kfree_skbmem+0x8/0x61
>> Mar 29 09:38:24 pstn1 kernel:  [] __udp_queue_rcv_skb+0x4a/0x51
>> Mar 29 09:38:24 pstn1 kernel:  [] release_sock+0x44/0x91
>> Mar 29 09:38:24 pstn1 kernel:  [] udp_sendmsg+0x44e/0x514
>> Mar 29 09:38:24 pstn1 kernel:  [] inet_sendmsg+0x35/0x3f
>> Mar 29 09:38:24 pstn1 kernel:  [] sock_sendmsg+0xce/0xe8
>> Mar 29 09:38:24 pstn1 kernel:  []
>> autoremove_wake_function+0x0/0x2d
>> Mar 29 09:38:24 pstn1 kernel:  [] copy_from_user+0x17/0x5d
>> Mar 29 09:38:24 pstn1 kernel:  [] copy_to_user+0x31/0x48
>> Mar 29 09:38:24 pstn1 kernel:  [] zt_chan_read+0x1e0/0x20b
>> [zaptel]
>> Mar 29 09:38:24 pstn1 kernel:  [] copy_from_user+0x31/0x5d
>> Mar 29 09:38:24 pstn1 kernel:  [] sys_sendto+0x116/0x140
>> Mar 29 09:38:24 pstn1 kernel:  [] flush_tlb_page+0x74/0x77
>> Mar 29 09:38:24 pstn1 kernel:  [] do_wp_page+0x3bf/0x40a
>> Mar 29 09:38:24 pstn1 kernel:  [] current_fs_time+0x4a/0x55
>> Mar 29 09:38:24 pstn1 kernel:  [] touch_atime+0x60/0x91
>> Mar 29 09:38:24 pstn1 kernel:  [] pipe_readv+0x315/0x321
>> Mar 29 09:38:24 pstn1 kernel:  [] sys_socketcall+0x106/0x19e
>> Mar 29 09:38:24 pstn1 kernel:  [] syscall_call+0x7/0xb
>> Mar 29 09:38:24 pstn1 kernel:  ===
>>
>>
>> This occurred during a "high load" period (52 calls across 3 PRI spans).
>>
>> A couple days ago I moved the interrupts for my PRI card to CPU0 from
>> CPU3, because CPU3 was handling everything else:
>>           CPU0       CPU1       CPU2       CPU3
>>  0:        306          0          0 3684057379    IO-APIC-edge  timer
>>  1:          0          0          0      13468    IO-APIC-edge  i8042
>>  8:          0          0          0          3    IO-APIC-edge  rtc
>>  9:          0          0          0          0   IO-APIC-level  acpi
>>  12:          0          0          0          4    IO-APIC-edge  i8042
>> 169:          0          0          0          0   IO-APIC-level
>>  uhci_hcd:usb2
>> 177:          0          0          0   18392593   IO-APIC-level  ata_piix
>> 185:          0          0          0          1   IO-APIC-level
>>  ehci_hcd:usb1
>> 193:          0          0          0          0   IO-APIC-level
>>  uhci_hcd:usb3
>> 201:          0          0          0 2090021759   IO-APIC-level  eth0
>> 209:  149621223          0          0 3534419461   IO-APIC-level  wct4xxp
>>
>>
>> (The CPU3 number for wct4xxp is not increasing any more).
>>
>> What is the interrupt distribution of other people's systems?
>> Before I made this change I was having a problem with D-channels
>> dropping occasionally, so I thought it might be an interrupt/load
>> issue.
>>
>> Thank you.
>>
>> -- 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
>
>
> --
> _
> -- 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
>

Re: [asterisk-users] How are your PRI interrupts balanced? (+ Soft lockup BUG)

[Matt Watson]

Dell server by any chance?

I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
several other people having issues with digium cards in dell servers as
well.

I've actually done something similar to what you have done - isolated the
TE220B onto its own IRQ and set processor affinity for all the IRQs to
particular cores... so far I haven't had kernel pancs since doing this, but
its still a little too early to say if it has fixed the issue 100% or not.

--
Matt

On Mon, Mar 29, 2010 at 8:30 PM, James Lamanna  wrote:

> Hi,
> I'm trying to figure out the cause of a soft lockup I experienced:
>
> Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s!
> [asterisk:32029]
> Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk
> Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[] CPU: 0
> Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c
> Mar 29 09:38:24 pstn1 kernel:  EFLAGS: 0286Tainted: GF
> (2.6.18-128.1.10.el5 #1)
> Mar 29 09:38:24 pstn1 kernel: EAX: 0029 EBX: f7ff9380 ECX:
> f7fff880 EDX: c11ff9a0
> Mar 29 09:38:24 pstn1 kernel: ESI: 0286 EDI: cffcda00 EBP:
> e5e10c80 DS: 007b ES: 007b
> Mar 29 09:38:24 pstn1 kernel: CR0: 80050033 CR2: b7ce39e0 CR3:
> 0f911000 CR4: 06d0
> Mar 29 09:38:24 pstn1 kernel:  [] kfree_skbmem+0x8/0x61
> Mar 29 09:38:24 pstn1 kernel:  [] __udp_queue_rcv_skb+0x4a/0x51
> Mar 29 09:38:24 pstn1 kernel:  [] release_sock+0x44/0x91
> Mar 29 09:38:24 pstn1 kernel:  [] udp_sendmsg+0x44e/0x514
> Mar 29 09:38:24 pstn1 kernel:  [] inet_sendmsg+0x35/0x3f
> Mar 29 09:38:24 pstn1 kernel:  [] sock_sendmsg+0xce/0xe8
> Mar 29 09:38:24 pstn1 kernel:  []
> autoremove_wake_function+0x0/0x2d
> Mar 29 09:38:24 pstn1 kernel:  [] copy_from_user+0x17/0x5d
> Mar 29 09:38:24 pstn1 kernel:  [] copy_to_user+0x31/0x48
> Mar 29 09:38:24 pstn1 kernel:  [] zt_chan_read+0x1e0/0x20b
> [zaptel]
> Mar 29 09:38:24 pstn1 kernel:  [] copy_from_user+0x31/0x5d
> Mar 29 09:38:24 pstn1 kernel:  [] sys_sendto+0x116/0x140
> Mar 29 09:38:24 pstn1 kernel:  [] flush_tlb_page+0x74/0x77
> Mar 29 09:38:24 pstn1 kernel:  [] do_wp_page+0x3bf/0x40a
> Mar 29 09:38:24 pstn1 kernel:  [] current_fs_time+0x4a/0x55
> Mar 29 09:38:24 pstn1 kernel:  [] touch_atime+0x60/0x91
> Mar 29 09:38:24 pstn1 kernel:  [] pipe_readv+0x315/0x321
> Mar 29 09:38:24 pstn1 kernel:  [] sys_socketcall+0x106/0x19e
> Mar 29 09:38:24 pstn1 kernel:  [] syscall_call+0x7/0xb
> Mar 29 09:38:24 pstn1 kernel:  ===
>
>
> This occurred during a "high load" period (52 calls across 3 PRI spans).
>
> A couple days ago I moved the interrupts for my PRI card to CPU0 from
> CPU3, because CPU3 was handling everything else:
>   CPU0   CPU1   CPU2   CPU3
>  0:306  0  0 3684057379IO-APIC-edge  timer
>  1:  0  0  0  13468IO-APIC-edge  i8042
>  8:  0  0  0  3IO-APIC-edge  rtc
>  9:  0  0  0  0   IO-APIC-level  acpi
>  12:  0  0  0  4IO-APIC-edge  i8042
> 169:  0  0  0  0   IO-APIC-level
>  uhci_hcd:usb2
> 177:  0  0  0   18392593   IO-APIC-level  ata_piix
> 185:  0  0  0  1   IO-APIC-level
>  ehci_hcd:usb1
> 193:  0  0  0  0   IO-APIC-level
>  uhci_hcd:usb3
> 201:  0  0  0 2090021759   IO-APIC-level  eth0
> 209:  149621223  0  0 3534419461   IO-APIC-level  wct4xxp
>
>
> (The CPU3 number for wct4xxp is not increasing any more).
>
> What is the interrupt distribution of other people's systems?
> Before I made this change I was having a problem with D-channels
> dropping occasionally, so I thought it might be an interrupt/load
> issue.
>
> Thank you.
>
> -- 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
>
-- 
_
-- 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] Asynchronous play music

[Pham Quy]

Hi all, 

Is there anyway to catch DTMF keypress while a music file is playing
without stop the music?

Thanks,

Quyps 


-- 
_
-- 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] Trying to configure xorcom on Suse 11

[Tim Panton]

On 29 Mar 2010, at 08:13, Tzafrir Cohen wrote:

> On Sun, Mar 28, 2010 at 09:16:48PM -0500, Tim Panton wrote:
>> 
>> On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote:
>> 
>>> On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
 I'm having trouble getting a xorcom set up.
 
 A large part of the problem is that the box is a _long_ way away and 
 I can't get to/at it easily, so while I could probably fix this in a few
 hours if the machine were in front of me, I'm struggling over a slow
 unreliable laggy link. 
 
 Ok, enough whining from me.
 
 I have a new Xorcom plugged into the usb of a Suse 11 machine
 I built Dahdi from trunk (last thursday) 
 
 # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux
 # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools
 
 dahdihardware -v sees the box but no spans.
>>> 
>>> Generally '/etc/init.d/dahdi start' . Or more specifically,
>>> 'dahdi_registration on' .
>>> 
>>> See also:
>>> 
>>> http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios
>> 
>> 
>> I've must be missing something here - this is what I see now.
>> 
>> sh-4.0# dahdi_hardware -v
>> usb:001/020  xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
>> LABEL=[usb:X1037246]   connect...@usb-:00:1a.7-4 
>> 
>> Shouldn't I see spans ??? I think the box (I've never seen it, but I know 
>> what I asked for) 
>> has 8fxs+8fxo+2E1 . 
> 
> Yes, you should. Any relevant kernel messages?

2010-03-29T02:50:36.515445-11:00 pbx kernel: [467270.047734] FIRMWARE: : 
XPD=0-0  (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 
2010-03-29T02:50:36.515511-11:00 pbx kernel: [467270.047734] NOTICE-xpp: 
XBUS-00(00): FIRMWARE ERROR_CODE: category=2 errorbits=0x02 
(rate_limit=1669611216)
2010-03-29T02:50:36.515577-11:00 pbx kernel: [467270.047734] FIRMWARE: : 
XPD=0-0  (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 
2010-03-29T02:50:36.515645-11:00 pbx kernel: [467270.047734] NOTICE-xpp: 
XBUS-00(00): FIRMWARE ERROR_CODE: category=2 errorbits=0x02 
(rate_limit=1669611217)
2010-03-29T02:50:36.515711-11:00 pbx kernel: [467270.047734] FIRMWARE: : 
XPD=0-0  (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 
2010-03-29T02:50:36.515789-11:00 pbx kernel: [467270.047734] NOTICE-xpp: 
XBUS-00(00): FIRMWARE ERROR_CODE: category=2 errorbits=0x02 
(rate_limit=1669611218)
2010-03-29T02:50:36.515859-11:00 pbx kernel: [467270.047734] FIRMWARE: : 
XPD=0-0  (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 
2010-03-29T02:50:36.515926-11:00 pbx kernel: [467270.047734] NOTICE-xpp: 
XBUS-00(00): FIRMWARE ERROR_CODE: category=2 errorbits=0x02 
(rate_limit=1669611219)
2010-03-29T02:50:36.515993-11:00 pbx kernel: [467270.047734] FIRMWARE: : 
XPD=0-0  (0x0) OP=0x22 LEN=14 BYTES: 0E 00 22 00 02 02 08 00 09 00 04 00 00 00 
2010-03-29T02:50:36.516060-11:00 pbx kernel: [467270.047734] 


> 
> If not: try:
> 
>  rmmod xpp_usb
>  modprobe xpp_usb
> 
> What new messages do you then see in /var/log/messages ?



Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




-- 
_
-- 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] a2billing wont pass the number

[Nathanial Allan]

I am running into an issue with A2Billing. I will explain  first of all that 
everything else works! the system is 90% complete its just this one small 
problem I am running into. 

So my problem is that when I place a call, 
1. I dial my number that I want and A2Billing gets activated 
2. it asks for my pin, upon successful entry of my pin A2Billing then
3. prompts me for my phone number then 
4. The call goes out (and actually connects for the record)

So I am entering my destination phone number twice which is not the worst thing 
that can happen, though it is a little annoying

Any light that you can shine on this problem would be greatly appreciated as I 
have been working on it for too long now and I want to get a product!


Thank You

NallaN
-- 
_
-- 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] Diameter for Asterisk, Traffix Diameter stack ?

[mara greenberg]

Hi,

I need a Diameter protocol stack (DDA/Ro) for Asterisk.
I came across Traffix Systems Diameter stack. according to the site they got 
Asterisk ready Diameter stack and also Diameter Gateway that can interface to 
Asterisk, has anyone tried those out ?


thanks
Maria


  

-- 
_
-- 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-biz] Asterisk system for church call center

[Mark Phillips]

They say confession is good for the soul. Perhaps they are offering a 
phone in confessional service?

Unfortunately the "business" of the church often flies in the face of 
the business of the Church.



On 03/29/2010 07:48 PM, Alex Balashov wrote:
> Sounds like the church has strayed from its core competencies and
> invited the money-changers into the temple.
>
> Being the official asterisk-biz harbinger of God's wrath, I suggest an
> intensely commercial platform, for the meek shall inherit the Earth,
> not the 700 Club.  Fight the power.
>

-- 
_
-- 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] How are your PRI interrupts balanced? (+ Soft lockup BUG)

[James Lamanna]

Hi,
I'm trying to figure out the cause of a soft lockup I experienced:

Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s!
[asterisk:32029]
Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk
Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[] CPU: 0
Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c
Mar 29 09:38:24 pstn1 kernel:  EFLAGS: 0286Tainted: GF
(2.6.18-128.1.10.el5 #1)
Mar 29 09:38:24 pstn1 kernel: EAX: 0029 EBX: f7ff9380 ECX:
f7fff880 EDX: c11ff9a0
Mar 29 09:38:24 pstn1 kernel: ESI: 0286 EDI: cffcda00 EBP:
e5e10c80 DS: 007b ES: 007b
Mar 29 09:38:24 pstn1 kernel: CR0: 80050033 CR2: b7ce39e0 CR3:
0f911000 CR4: 06d0
Mar 29 09:38:24 pstn1 kernel:  [] kfree_skbmem+0x8/0x61
Mar 29 09:38:24 pstn1 kernel:  [] __udp_queue_rcv_skb+0x4a/0x51
Mar 29 09:38:24 pstn1 kernel:  [] release_sock+0x44/0x91
Mar 29 09:38:24 pstn1 kernel:  [] udp_sendmsg+0x44e/0x514
Mar 29 09:38:24 pstn1 kernel:  [] inet_sendmsg+0x35/0x3f
Mar 29 09:38:24 pstn1 kernel:  [] sock_sendmsg+0xce/0xe8
Mar 29 09:38:24 pstn1 kernel:  [] autoremove_wake_function+0x0/0x2d
Mar 29 09:38:24 pstn1 kernel:  [] copy_from_user+0x17/0x5d
Mar 29 09:38:24 pstn1 kernel:  [] copy_to_user+0x31/0x48
Mar 29 09:38:24 pstn1 kernel:  [] zt_chan_read+0x1e0/0x20b [zaptel]
Mar 29 09:38:24 pstn1 kernel:  [] copy_from_user+0x31/0x5d
Mar 29 09:38:24 pstn1 kernel:  [] sys_sendto+0x116/0x140
Mar 29 09:38:24 pstn1 kernel:  [] flush_tlb_page+0x74/0x77
Mar 29 09:38:24 pstn1 kernel:  [] do_wp_page+0x3bf/0x40a
Mar 29 09:38:24 pstn1 kernel:  [] current_fs_time+0x4a/0x55
Mar 29 09:38:24 pstn1 kernel:  [] touch_atime+0x60/0x91
Mar 29 09:38:24 pstn1 kernel:  [] pipe_readv+0x315/0x321
Mar 29 09:38:24 pstn1 kernel:  [] sys_socketcall+0x106/0x19e
Mar 29 09:38:24 pstn1 kernel:  [] syscall_call+0x7/0xb
Mar 29 09:38:24 pstn1 kernel:  ===


This occurred during a "high load" period (52 calls across 3 PRI spans).

A couple days ago I moved the interrupts for my PRI card to CPU0 from
CPU3, because CPU3 was handling everything else:
   CPU0   CPU1   CPU2   CPU3
  0:306  0  0 3684057379IO-APIC-edge  timer
  1:  0  0  0  13468IO-APIC-edge  i8042
  8:  0  0  0  3IO-APIC-edge  rtc
  9:  0  0  0  0   IO-APIC-level  acpi
 12:  0  0  0  4IO-APIC-edge  i8042
169:  0  0  0  0   IO-APIC-level  uhci_hcd:usb2
177:  0  0  0   18392593   IO-APIC-level  ata_piix
185:  0  0  0  1   IO-APIC-level  ehci_hcd:usb1
193:  0  0  0  0   IO-APIC-level  uhci_hcd:usb3
201:  0  0  0 2090021759   IO-APIC-level  eth0
209:  149621223  0  0 3534419461   IO-APIC-level  wct4xxp


(The CPU3 number for wct4xxp is not increasing any more).

What is the interrupt distribution of other people's systems?
Before I made this change I was having a problem with D-channels
dropping occasionally, so I thought it might be an interrupt/load
issue.

Thank you.

-- 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

[asterisk-users] Asterisk and Call files

[Anthony Geoffron]

Hello,


I was planning on using a call file to test my IVR on a regular basis to
ensure it is operational


Channel: local/1...@from-internal
Application: SendDTMF
Data: ww12345678#1w1234#w1ww

But what ever I try so far the IVR does not seem to take the data input of
the application SendDTMF
However in The ASterisk logs look good...
-- Attempting call on local/1...@from-internal for application
SendDTMF(ww12345678#1w1234#w)
-- Executing [1...@from-internal:1] Answer("
Local/1...@from-internal-2a1e,2", "") in new stack
-- Executing [1...@from-internal:2] AGI("Local/1...@from-internal-2a1e,2",
"agi://localhost/url=http%3A%2F%2Flocalhost%2Fvxml%2k

Any idea? what could be wrong here?

Thanks
-- 
_
-- 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] Trying to get reason for ending of AGI call recording

[Jeff Johnson]

That worked...The help is very much appreciated.

Jeff 

http://www.neturallyspeaking.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, March 29, 2010 8:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trying to get reason for ending of AGI call 
recording

On Mon, 29 Mar 2010, Jeff Johnson wrote:

> I would appreciate any ideas of what I’m doing wrong on this.   My 
> dialplan calls an AGI which records a file.  That works, but I’m trying 
> to find a way to determine whether the caller pressed # to stop a 
> recording before the maxtime expired, or if the recording ended due to 
> reaching the max timeout.  The $fx variable in the below agi excerpt 
> always returns 0.
> 
> $res = $agi->exec('Record', "$filename.gsm||$maxtime");
> 
> $fx = $res['result'];

Try using the AGI command "record file" instead of the "application 
record()."

I get a valid 'result' :)

-- 
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] Trying to get reason for ending of AGI call recording

[Steve Edwards]

On Mon, 29 Mar 2010, Jeff Johnson wrote:

I would appreciate any ideas of what I’m doing wrong on this.   My 
dialplan calls an AGI which records a file.  That works, but I’m trying 
to find a way to determine whether the caller pressed # to stop a 
recording before the maxtime expired, or if the recording ended due to 
reaching the max timeout.  The $fx variable in the below agi excerpt 
always returns 0.


$res = $agi->exec('Record', "$filename.gsm||$maxtime");

$fx = $res['result'];


Try using the AGI command "record file" instead of the "application 
record()."


I get a valid 'result' :)

--
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] Trying to get reason for ending of AGI call recording

[Jeff Johnson]

I would appreciate any ideas of what I'm doing wrong on this.   My
dialplan calls an AGI which records a file.  That works, but I'm trying
to find a way to determine whether the caller pressed # to stop a
recording before the maxtime expired, or if the recording ended due to
reaching the max timeout.  The $fx variable in the below agi excerpt
always returns 0.

 

$res = $agi->exec('Record', "$filename.gsm||$maxtime");

$fx = $res['result'];

 

Thanks,

 

Jeff 

 

-- 
_
-- 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-biz] Asterisk system for church call center

[Alex Balashov]

Sounds like the church has strayed from its core competencies and  
invited the money-changers into the temple.

Being the official asterisk-biz harbinger of God's wrath, I suggest an  
intensely commercial platform, for the meek shall inherit the Earth,  
not the 700 Club.  Fight the power.

-- 
Sent from mobile device

On Mar 29, 2010, at 4:58 PM, Frank Church  wrote:

> I have been asked by my church to recommend a VoIP system which can do
> the following.
>
> They do internet radio shows which are sometimes broadcast on radio.
>
> They are looking for a system which does the following for about 5
> agents, exactly as they have described it.
>
> 1. Take incoming calls
>
> 2. Put them on hold if there is no one to handle the call immediately,
> or transfer them to an available agent
>
> 3. Take down their details, and number, (if this can be retrieved and
> saved from the caller id, thats better)
>
> 4. Get them to hold on after taking their details if they still want  
> to hold
>
> 5. Call them back when the backlog is cleared up.
>
> I have a fairly good grasp of the hardware and programming part of
> Asterisk, having compiled it more than a few times and implemented
> A2Billing phone card and call shop system with it.
>
> But the type of software suited to the Call Center side is where my
> knowledge gap lies.
>
> I am looking for solutions based on the usual Asterisk distributions
> like AsteriskNow, trixbox, elastix etc, whether ready packaged or
> requiring additional customization.
>
>
> The matter of whether they will use soft phones, or regular phones
> with headsets is also something to consider. Soft phones with good
> GUI's may be preferred if more cost effective for them, although my
> personal preferences are with hard phones.
>
> Any recommendations - the ease of software for the end users is the
> main thing for me, and integration with the database for taking
> customers details is the main thing for me. One of the distributions
> with SugarCRM comes to mind here.
>
> Sorry for cross-posting, but ready made and commercially supported
> systems are not ruled out, if they come within their budget.
>
> Regards
>
>
> Frank Church
>
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-biz

-- 
_
-- 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 system for church call center

[Duncan Turnbull]

Hi Frank

I have found Freepbx on top of Asterisk a good solution for the church I look 
after and the rest of my customers, the callcentre functions you need are built 
in it and if they have someone technical then they can expand what they are 
doing

It has both queues and ring groups (which are often all they need) 

I would imagine you just send them to an IVR or mailbox to ask for their name 
and details then move them into a higher priority second queue 

Elastix has Sugar in it I believe and looks okay but I like debian/ubuntu as a 
base distribution rather than Centos so haven't gone that way.

However most of my customers still struggle with the concepts involved in 
telephony so while they are happy to look at it while I am there they forget 
quickly how to drive it or lose their nerve, especially the church which is 
faith focussed rather than tech focussed ;-)

Because of that I think you want the easiest system for you to maintain 
remotely and elastix or freepbx is pretty easy. It also allows you to say this 
is available, this is not which is useful in narrowing down requirements. 

Cheers Duncan
On 30/03/2010, at 9:58 AM, Frank Church wrote:

> I have been asked by my church to recommend a VoIP system which can do
> the following.
> 
> They do internet radio shows which are sometimes broadcast on radio.
> 
> They are looking for a system which does the following for about 5
> agents, exactly as they have described it.
> 
> 1. Take incoming calls
> 
> 2. Put them on hold if there is no one to handle the call immediately,
> or transfer them to an available agent
> 
> 3. Take down their details, and number, (if this can be retrieved and
> saved from the caller id, thats better)
> 
> 4. Get them to hold on after taking their details if they still want to hold
> 
> 5. Call them back when the backlog is cleared up.
> 
> I have a fairly good grasp of the hardware and programming part of
> Asterisk, having compiled it more than a few times and implemented
> A2Billing phone card and call shop system with it.
> 
> But the type of software suited to the Call Center side is where my
> knowledge gap lies.
> 
> I am looking for solutions based on the usual Asterisk distributions
> like AsteriskNow, trixbox, elastix etc, whether ready packaged or
> requiring additional customization.
> 
> 
> The matter of whether they will use soft phones, or regular phones
> with headsets is also something to consider. Soft phones with good
> GUI's may be preferred if more cost effective for them, although my
> personal preferences are with hard phones.
> 
> Any recommendations - the ease of software for the end users is the
> main thing for me, and integration with the database for taking
> customers details is the main thing for me. One of the distributions
> with SugarCRM comes to mind here.
> 
> Sorry for cross-posting, but ready made and commercially supported
> systems are not ruled out, if they come within their budget.
> 
> Regards
> 
> 
> Frank Church
> 
> -- 
> _
> -- 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

[asterisk-users] amr

[Hans Witvliet]

Just noticed that packman has precompiled versions of amr codec. 
Both wideband and narrowband. Can these be used for asterisk?
Heard some nice about AMR (in general)

If so, any one around with experience with either??

hw

-- 
_
-- 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 system for church call center

[Gondar Monn]

Apart from the call back function, appears to me that any asterisk
distribution interfaced with freepbx will do what you need.
I would recommend pbxinaflash (http://www.pbxinaflash.com), they have a very
active forum, and will get up and running very fast.

Gondar

On Mon, Mar 29, 2010 at 1:46 PM, Frank Church  wrote:

> I have been asked by my church to recommend a VoIP system which can do
> the following.
>
> They do internet radio shows which are sometimes broadcast on radio.
>
> They are looking for a system which does the following for about 5
> agents, exactly as they have described it.
>
> 1. Take incoming calls
>
> 2. Put them on hold if there is no one to handle the call immediately,
> or transfer them to an available agent
>
> 3. Take down their details, and number, (if this can be retrieved and
> saved from the caller id, thats better)
>
> 4. Get them to hold on after taking their details if they still want to
> hold
>
> 5. Call them back when the backlog is cleared up.
>
> I have a fairly good grasp of the hardware and programming part of
> Asterisk, having compiled it more than a few times and implemented
> A2Billing phone card and call shop system with it.
>
> But the type of software suited to the Call Center side is where my
> knowledge gap lies.
>
> I am looking for solutions based on the usual Asterisk distributions
> like AsteriskNow, trixbox, elastix etc, whether ready packaged or
> requiring additional customization.
>
>
> The matter of whether they will use soft phones, or regular phones
> with headsets is also something to consider. Soft phones with good
> GUI's may be preferred if more cost effective for them, although my
> personal preferences are with hard phones.
>
> Any recommendations - the ease of software for the end users is the
> main thing for me, and integration with the database for taking
> customers details is the main thing for me. One of the distributions
> with SugarCRM comes to mind here.
>
> Sorry for cross-posting, but ready made and commercially supported
> systems are not ruled out, if they come within their budget.
>
> Regards
>
>
> Frank Church
>
> --
> _
> -- 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

[asterisk-users] Asterisk system for church call center

[Frank Church]

I have been asked by my church to recommend a VoIP system which can do
the following.

They do internet radio shows which are sometimes broadcast on radio.

They are looking for a system which does the following for about 5
agents, exactly as they have described it.

1. Take incoming calls

2. Put them on hold if there is no one to handle the call immediately,
or transfer them to an available agent

3. Take down their details, and number, (if this can be retrieved and
saved from the caller id, thats better)

4. Get them to hold on after taking their details if they still want to hold

5. Call them back when the backlog is cleared up.

I have a fairly good grasp of the hardware and programming part of
Asterisk, having compiled it more than a few times and implemented
A2Billing phone card and call shop system with it.

But the type of software suited to the Call Center side is where my
knowledge gap lies.

I am looking for solutions based on the usual Asterisk distributions
like AsteriskNow, trixbox, elastix etc, whether ready packaged or
requiring additional customization.


The matter of whether they will use soft phones, or regular phones
with headsets is also something to consider. Soft phones with good
GUI's may be preferred if more cost effective for them, although my
personal preferences are with hard phones.

Any recommendations - the ease of software for the end users is the
main thing for me, and integration with the database for taking
customers details is the main thing for me. One of the distributions
with SugarCRM comes to mind here.

Sorry for cross-posting, but ready made and commercially supported
systems are not ruled out, if they come within their budget.

Regards


Frank Church

-- 
_
-- 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, IAX, & Sub interfaces

[Steve Edwards]

Un-top-posting...

>> - "trebaum"  wrote:

>>> So I realize the error in my question/request.  The section I was 
>>> thinking of using for the binding IP address, is, itself, the wrong 
>>> place to do such a thing.  It would need to be in the register 
>>> statement... something like the following...
>>>
>>> register => user:pass:fro...@targetpeer
>>>

> On Mar 29, 2010, at 1:18 PM, Tim Nelson wrote:
>
>> Why not add a route to your system for each of the three IPs you'll be 
>> registering to with their associated interfaces?

On Mon, 29 Mar 2010, trebaum wrote:

> I thought about that, but there are a couple issues with that. 
> Currently the physical interface has a single gateway (the only way to 
> change this is to add more physical interfaces), and that gateway is 
> what does the routing to the specific peers.  Being that all of the 
> traffic is going to a single point, and I can't specify what interface 
> the traffic is originating from (in *), when it gets to the default gw 
> all of the traffic looks the same.

I don't have the skills, but I'd bet a beer somebody who knows iptables 
well could come up with something to mangle the source IP address based on 
the destination IP address.

-- 
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] Asterisk, IAX, & Sub interfaces

[trebaum]

I thought about that, but there are a couple issues with that.  Currently the 
physical interface has a single gateway (the only way to change this is to add 
more physical interfaces), and that gateway is what does the routing to the 
specific peers.  Being that all of the traffic is going to a single point, and 
I can't specify what interface the traffic is originating from (in *), when it 
gets to the default gw all of the traffic looks the same.

~ T

On Mar 29, 2010, at 1:18 PM, Tim Nelson wrote:

> - "trebaum"  wrote:
>> So I realize the error in my question/request.  The section I was
>> thinking of using for the binding IP address, is, itself, the wrong
>> place to do such a thing.  It would need to be in the register
>> statement... something like the following...
>> 
>> register => user:pass:fro...@targetpeer
>> 
> 
> Why not add a route to your system for each of the three IPs you'll be 
> registering to with their associated interfaces?
> 
> Tim Nelson
> 
> -- 
> _
> -- 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

[asterisk-users] Asterisk system for church call center

[Frank Church]

I have been asked by my church to recommend a VoIP system which can do
the following.

They do internet radio shows which are sometimes broadcast on radio.

They are looking for a system which does the following for about 5
agents, exactly as they have described it.

1. Take incoming calls

2. Put them on hold if there is no one to handle the call immediately,
or transfer them to an available agent

3. Take down their details, and number, (if this can be retrieved and
saved from the caller id, thats better)

4. Get them to hold on after taking their details if they still want to hold

5. Call them back when the backlog is cleared up.

I have a fairly good grasp of the hardware and programming part of
Asterisk, having compiled it more than a few times and implemented
A2Billing phone card and call shop system with it.

But the type of software suited to the Call Center side is where my
knowledge gap lies.

I am looking for solutions based on the usual Asterisk distributions
like AsteriskNow, trixbox, elastix etc, whether ready packaged or
requiring additional customization.


The matter of whether they will use soft phones, or regular phones
with headsets is also something to consider. Soft phones with good
GUI's may be preferred if more cost effective for them, although my
personal preferences are with hard phones.

Any recommendations - the ease of software for the end users is the
main thing for me, and integration with the database for taking
customers details is the main thing for me. One of the distributions
with SugarCRM comes to mind here.

Sorry for cross-posting, but ready made and commercially supported
systems are not ruled out, if they come within their budget.

Regards


Frank Church

-- 
_
-- 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, IAX, & Sub interfaces

[Tim Nelson]

- "trebaum"  wrote:
> So I realize the error in my question/request.  The section I was
> thinking of using for the binding IP address, is, itself, the wrong
> place to do such a thing.  It would need to be in the register
> statement... something like the following...
> 
> register => user:pass:fro...@targetpeer
> 

Why not add a route to your system for each of the three IPs you'll be 
registering to with their associated interfaces?

Tim Nelson

-- 
_
-- 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, IAX, & Sub interfaces

[trebaum]

So I realize the error in my question/request.  The section I was thinking of 
using for the binding IP address, is, itself, the wrong place to do such a 
thing.  It would need to be in the register statement... something like the 
following...

register => user:pass:fro...@targetpeer

I do realize that this functionality doesn't currently exist.  Requests will be 
made in the appropriate places.  Thanks for the help.

:)

~ T

On Mar 29, 2010, at 12:03 PM, trebaum wrote:

> Is there anyway to get the following scenario to work...
> 
> I have 3 IAX trunks that I want to setup to peer with other * boxes.  I have 
> 1 physical interface, eth0.  I also have 2 sub interfaces, eth0:1 & eth0:2.  
> I want to setup a single IAX trunk on each of the interfaces.  All 3 
> interfaces are going to have separate publicly routable IPs, and for this 
> purpose, let's say that because of operational restrictions, I cannot just 
> setup all 3 physical interfaces.
> 
> What would really be nice, is if in the entries for each of the trunks to be 
> able to use the bindaddr setting per trunk.  This would allow me to be able 
> to connect trunk1 from eth0, trunk2 from eth0:1, trunk3 from eth0:2.
> 
> Thanks
> 
> ~ T
> 
> 
> -- 
> _
> -- 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] Libtonezone

[Tzafrir Cohen]

On Mon, Mar 29, 2010 at 04:03:12PM +, Joseph L. Casale wrote:
> >You could read the source code, but based on it's name I would say it is a 
> >library responsible for zone specific tone generation. Many parts of the 
> >world have different tone >patterns than the U.S. and Asterisk is used 
> >worldwide. A better question is, why are you concerned by it?
> 
> I was building rpm's for dahdi w/ oslec using Anthony Messina's spec file
> and he pulls in the shared object as a dep, but looking at digiums repo, it
> isn't pulled in as a dep by any of the dahdi rpms?

It is a library used by Asterisk. It is part of dahdi-tools. It is not a
dependency of it. It should be provided by it.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- 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] 24 FXS Port Voip Gateway and Asterisk

[Doug]

At 23:26 3/28/2010, James Lamanna wrote:
 >On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards
 > wrote:
 >> On Sun, 28 Mar 2010, Joseph Begumisa wrote:
 >>
 >>> Can anyone recommend a 24 fxs port voip gateway that has worked well with
 >>> asterisk?  I have a couple of analog handsets that I want to hookup to my
 >>> asterisk server?  Any tested and tried product recommendations 
are welcome.
 >>>  Thanks.
 >>
 >> Adtran channel banks are a great "trailing edge" technology. You can get
 >> them off Ebay for pennies on the original dollar and they are built like a
 >> tank.
 >>
 >> ("voip gateway" is not very specific. If you meant SIP or IAX, you might
 >> want to specify which.)
 >
 >I've actually had decent success with the GXW-4024 (FXS <-> SIP) from
 >Grandstream which is probably one of the cheapest 24 FXS port boxes
 >you'll find out there.
 >
 >-- James

Grandstream = Garbage.  You have been warned.


-- 
_
-- 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] 24 FXS Port Voip Gateway and Asterisk

[Joseph Begumisa]

Thanks for the feedback.  Btw, I meant SIP / IAX gateway.  I'll take a look
at the suggestions.

Best Regards,

Joseph


On Sun, Mar 28, 2010 at 7:28 PM, Steve Edwards wrote:

> On Sun, 28 Mar 2010, Joseph Begumisa wrote:
>
>  Can anyone recommend a 24 fxs port voip gateway that has worked well with
>> asterisk?  I have a couple of analog handsets that I want to hookup to my
>> asterisk server?  Any tested and tried product recommendations are welcome.
>>  Thanks.
>>
>
> Adtran channel banks are a great "trailing edge" technology. You can get
> them off Ebay for pennies on the original dollar and they are built like a
> tank.
>
> ("voip gateway" is not very specific. If you meant SIP or IAX, you might
> want to specify which.)
>
> --
> 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
>
-- 
_
-- 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] Foip solution

[Zeeshan Zakaria]

When it comes to FoIP, it is a good idea to stay with analog lines and
regular fax. FoIP is a pain and not recommended where fax is a regular part
of a business.

On 2010-03-29 3:20 PM, "David Backeberg"  wrote:

On Mon, Mar 29, 2010 at 2:26 PM, Mike Diehl  wrote:
> On Monday 29 March 2010 1...
The easiest recommendation:
* call the local phone provider, get a few analog lines, install a fax
machine

Done.


-- 
_
-- Bandwidth and Colocati...
-- 
_
-- 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] Slightly more advanced dialling..

[Zeeshan Zakaria]

Hi,

I have done it a few times. Just posted a small blog about it with code.
Check it at www.ilovetovoip.com/?p=322. I hope it'll help you.

--
Zeeshan A Zakaria

On 2010-03-29 11:07 AM, "Philipp von Klitzing" <
klitz...@pool.informatik.rwth-aachen.de> wrote:

Hi!


> I'm wondering if it is possible to ring X number of extensions
> simultaneously, and each answer...
You might want to explain what you are trying to do.

Dial() can handle this by using something like SIP/peer1&SIP/peer2
The first one that answers wins. Look at the Dial option M to run a macro
after the call has been answered.

Also have a look at FollowMe() since it can do parallel calling.
Or read up how to create a bunch of .call files using System() and a
script.


> I can do a huntgroup-esque way of dialling, but I want all the dialled
> numbers to be picked up
Do you mean to say: "I want all dialed numbers to keep on ringing until
they are answered, regardless if the initial callers has already been
taken care of by the first extensions that reacted"?

In the Asterisk world, and usually in the PBX world in general, pick up
has specific and different meaning (see *8 or app_pickup).

Philipp


--

_
-- Bandwidth and Colocation Pr...
-- 
_
-- 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] Foip solution

[David Backeberg]

On Mon, Mar 29, 2010 at 2:26 PM, Mike Diehl  wrote:
> On Monday 29 March 2010 10:15:50 am jon pounder wrote:
>> Mike Diehl wrote:
>> > Hi all,
>> >
>> > I've cross-posted this to the -users and -biz groups.  Hope that's OK.
>> >
>> > I have a customer who REALLY needs to be able to send/receive faxes
>> > reliably. I could probably get hylafax configured, but I'm not sure how
>> > reliable it is.
>> >
>> > If it is considered reliable, would someone let me know?
>> >
>> > Otherwise, is there a product/service they can buy that will allow them
>> > to fax to/from their computers?
>> >
>> > TIA,
>>
>> hylafax is "the standard" never had a problem with it.
>>
>> used to have the odd issue with a faxmodem on a fxs port from a channel
>> bank, now have it on a virtual iaxmodem, no problems at all. In fact we
>> have a whole bank of virtual modems and they work just fine.
>
> From what I'm hearing, I could experiment with hylafax, or I can try Fax for
> Asterisk.  If I use Fax for Asterisk, I'll need a T.38 provider since I am
> strictly using SIP trunks.  Any recommendations there?

The easiest recommendation:
* call the local phone provider, get a few analog lines, install a fax machine

Done.

-- 
_
-- 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] dnd not working correctly

[Ott Rose]

i posted this on the freepbx site. here is the response 


"from the trace, everything is working. Check your asterisk log for file
errors playing back the audio, could be your sound files are not
installed or messed up."



so i checked /etc/log/asterisk/full 

and in vi full i did /error   and  /117 (my ext) and /activate didn't really 
find anything

i didn't see anything but i might be over looking it. I did grep error full and 
it returned some errors but not related to dnd as far as i can tell. is there 
some place else to look, a better way to search that file, or can someone tell 
me what i am looking for?



Date: Fri, 26 Mar 2010 18:34:46 -0600
From: al...@vivoxie.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly

Seems like an Amportal configration problem not and Asterisk issue. Maybe you 
should try in one of the FreePBX users list.

Alyed



2010/3/26 Ott Rose 








i have posted this question couple of times and never really got any hits i 
wasn't able to provide any debug info 


Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)

Verbosity is at least 4

  == Using SIP RTP TOS bits 184

  == Using SIP RTP CoS mark 5

  == Using SIP VRTP TOS bits 136

  == Using SIP VRTP CoS mark 6

  == Extension Changed 117[ext-local] new state InUse for Notify User 102

  == Extension Changed 117[ext-local] new state InUse for Notify User 103

  == Extension Changed 117[ext-local] new state InUse for Notify User 114

-- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in new 
stack

-- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in new 
stack

-- Executing [...@from-internal:3] Macro("SIP/117-01f6", 
"user-callerid,") in new stack

-- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6", 
"AMPUSER=117") in new stack

-- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6", 
"0?report") in new stack

-- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6", 
"1?Set(REALCALLERIDNUM=117)") in new stack

-- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6", 
"AMPUSER=117") in new stack

-- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6", 
"AMPUSERCIDNAME=My Name") in new stack

-- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6", 
"0?report") in new stack

-- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6", 
"AMPUSERCID=117") in new stack

-- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6", 
"CALLERID(all)="My Name" <117>") in new stack

-- Executing [...@macro-user-callerid:9] GotoIf("SIP/117-01f6", 
"0?continue") in new stack

-- Executing [...@macro-user-callerid:10] Set("SIP/117-01f6", 
"__TTL=64") in new stack

-- Executing [...@macro-user-callerid:11] GotoIf("SIP/117-01f6", 
"1?continue") in new stack

-- Goto (macro-user-callerid,s,18)

-- Executing [...@macro-user-callerid:18] NoOp("SIP/117-01f6", "Using 
CallerID "My Name" <117>") in new stack

-- Executing [...@from-internal:4] GotoIf("SIP/117-01f6", 
"1?activate:deactivate") in new stack

-- Goto (from-internal,*76,5)

-- Executing [...@from-internal:5] Set("SIP/117-01f6", 
"DB(DND/117)=YES") in new stack

-- Executing [...@from-internal:6] Set("SIP/117-01f6", "STATE=BUSY") in 
new stack

-- Executing [...@from-internal:7] Gosub("SIP/117-01f6", 
"app-dnd-toggle,sstate,1") in new stack

-- Executing [sst...@app-dnd-toggle:1] Set("SIP/117-01f6", 
"DEVICE_STATE(Custom:DND117)=BUSY") in new stack

-- Executing [sst...@app-dnd-toggle:2] Set("SIP/117-01f6", 
"DEVICES=117") in new stack

-- Executing [sst...@app-dnd-toggle:3] GotoIf("SIP/117-01f6", 
"0?return") in new stack

  == Extension Changed 117[ext-local] new state Busy for Notify User 102

-- Executing [sst...@app-dnd-toggle:4] Set("SIP/117-01f6", "LOOPCNT=1") 
in new stack

-- Executing [sst...@app-dnd-toggle:5] Set("SIP/117-01f6", "ITER=1") in 
new stack

-- Executing [sst...@app-dnd-toggle:6] Set("SIP/117-01f6", 
"DEVICE_STATE(Custom:DEVDND117)=BUSY") in new stack

  == Extension Changed 117[ext-local] new state Busy for Notify User 103

  == Extension Changed 117[ext-local] new state Busy for Notify User 114

-- Executing [sst...@app-dnd-toggle:7] Set("SIP/117-01f6", "ITER=2") in 
new stack

-- Executing [sst...@app-dnd-toggle:8] GotoIf("SIP/117-01f6", 
"0?begin") in new stack

-- Executing [sst...@app-dnd-toggle:9] Return("SIP/117-01f6", "") in 
new stack

-- Executing [...@from-internal:8] Playback("SIP/117-01f6", 
"do-not-disturb&activated") in new stack

-- Executing [...@from-internal:9] Macro("SIP/117-01f6", "hangupcall,") 
in new stack

-- Executing [...@macro-hangupcall:1] GotoIf("SIP/117-01f6", 
"1?skiprg") in new stack

-- Goto (macro-hangupcall,s,4)

-- Executing

[asterisk-users] Asterisk, IAX, & Sub interfaces

[trebaum]

Is there anyway to get the following scenario to work...

I have 3 IAX trunks that I want to setup to peer with other * boxes.  I have 1 
physical interface, eth0.  I also have 2 sub interfaces, eth0:1 & eth0:2.  I 
want to setup a single IAX trunk on each of the interfaces.  All 3 interfaces 
are going to have separate publicly routable IPs, and for this purpose, let's 
say that because of operational restrictions, I cannot just setup all 3 
physical interfaces.

What would really be nice, is if in the entries for each of the trunks to be 
able to use the bindaddr setting per trunk.  This would allow me to be able to 
connect trunk1 from eth0, trunk2 from eth0:1, trunk3 from eth0:2.

Thanks

~ T


-- 
_
-- 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] Foip solution

[Mike Diehl]

On Monday 29 March 2010 10:15:50 am jon pounder wrote:
> Mike Diehl wrote:
> > Hi all,
> >
> > I've cross-posted this to the -users and -biz groups.  Hope that's OK.
> >
> > I have a customer who REALLY needs to be able to send/receive faxes
> > reliably. I could probably get hylafax configured, but I'm not sure how
> > reliable it is.
> >
> > If it is considered reliable, would someone let me know?
> >
> > Otherwise, is there a product/service they can buy that will allow them
> > to fax to/from their computers?
> >
> > TIA,
>
> hylafax is "the standard" never had a problem with it.
>
> used to have the odd issue with a faxmodem on a fxs port from a channel
> bank, now have it on a virtual iaxmodem, no problems at all. In fact we
> have a whole bank of virtual modems and they work just fine.

>From what I'm hearing, I could experiment with hylafax, or I can try Fax for 
Asterisk.  If I use Fax for Asterisk, I'll need a T.38 provider since I am 
strictly using SIP trunks.  Any recommendations there?

-- 

Take care and have fun,
Mike Diehl.

-- 
_
-- 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-biz] Foip solution

[Lee Howard]

Mike Diehl wrote:
> I could probably get hylafax configured, but I'm not sure how reliable it is.
>
> If it is considered reliable, would someone let me know?

It's reliable as long as you're not using FoIP (i.e. as long as you're 
faxing with PSTN lines).

Thanks,

Lee.

-- 
_
-- 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] Foip solution

[jon pounder]

Mike Diehl wrote:
> Hi all,
>
> I've cross-posted this to the -users and -biz groups.  Hope that's OK.
>
> I have a customer who REALLY needs to be able to send/receive faxes reliably. 
>  
> I could probably get hylafax configured, but I'm not sure how reliable it is.
>
> If it is considered reliable, would someone let me know?
>
> Otherwise, is there a product/service they can buy that will allow them to 
> fax 
> to/from their computers?
>
> TIA,
>
>   
hylafax is "the standard" never had a problem with it.

used to have the odd issue with a faxmodem on a fxs port from a channel 
bank, now have it on a virtual iaxmodem, no problems at all. In fact we 
have a whole bank of virtual modems and they work just fine.





-- 
_
-- 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] Background noise

[Jeff Brower]

Khalid-

> :) all users are having the same issue, even those connected to this server
> from abroad!

Since you have an identical working system, why are you not able to debug this? 
 First swap the phone... then swap any
cards in the server, then servers, then check carefully software differences 
etc.  You should be able to isolate one
thing that makes a difference, then you can zero in on that.

So far your questions are general and you can't give a specific problem 
definition other than "background noise" which
is a general thing that happens with radios, TVs, cars, parties, movies, etc -- 
not just Asterisk :-)  So it's
difficult for group members to give you advice.

Also suggest to not "cut" text from previous posts... most of your thread is 
missing below.

-Jeff


> 2010/3/29 Philipp von Klitzing 
>
>> > i have the same model polycom phone configured with another server
>> > (asterisk 1.4), and guess what no noise at all. any guess!
>>
>> Replace the handset?


-- 
_
-- 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] Foip solution

[Mike Diehl]

Hi all,

I've cross-posted this to the -users and -biz groups.  Hope that's OK.

I have a customer who REALLY needs to be able to send/receive faxes reliably.  
I could probably get hylafax configured, but I'm not sure how reliable it is.

If it is considered reliable, would someone let me know?

Otherwise, is there a product/service they can buy that will allow them to fax 
to/from their computers?

TIA,

-- 

Take care and have fun,
Mike Diehl.

-- 
_
-- 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] Libtonezone

[Joseph L. Casale]

>You could read the source code, but based on it's name I would say it is a 
>library responsible for zone specific tone generation. Many parts of the world 
>have different tone >patterns than the U.S. and Asterisk is used worldwide. A 
>better question is, why are you concerned by it?

I was building rpm's for dahdi w/ oslec using Anthony Messina's spec file
and he pulls in the shared object as a dep, but looking at digiums repo, it
isn't pulled in as a dep by any of the dahdi rpms?

Thanks!
jlc

-- 
_
-- 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] Realtime Issue

[D Tucny]

How about...

exten => _X.,1,NoOp()

exten => _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})

exten => _X.,3,Set(NULL="${REALTIME(agents,device,${DEVICE})}")

exten => _X.,4,Set(usernamepair=${CUT(NULL,\,,1)})

exten => _X.,5,Set(username=${CUT(usernamepair,=,2)})

exten => _X.,6,NoOp(DEVICE is ${DEVICE})

exten => _X.,7,NoOp(USERNAME is ${USERNAME})

exten => _X.,8,NoOp(username is ${username})


the REALTIME function returns a delimited string, it doesn't automatically
assign variables...


Perhaps you are thinking of the realtime application which would assign
variables? That has however was deprecated in 1.4 and has been removed from
1.6 in favour of the function which only returns a string...


d

On 29 March 2010 21:42, Jason Walker  wrote:

>  It seems that my realtime is not assigning channel variables correctly.
>
>
>
> INFO
>
> Asterisk 1.6.0.26
>
>
>
> Exten.conf
>
> exten => _X.,1,NoOp()
>
> exten => _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})
>
> exten => _X.,3,Set(NULL="${REALTIME(agents,device,${DEVICE})}")
>
> exten => _X.,4,NoOp(DEVICE is ${DEVICE})
>
> exten => _X.,5,NoOp(USERNAME is ${USERNAME})
>
> exten => _X.,6,NoOp(username is ${username})
>
>
>
>
>
> CLI
>
>
>
> -- Executing [...@default:1] NoOp("SIP/1156-55ce", "") in new stack
>
> -- Executing [...@default:2] Set("SIP/1156-55ce", "DEVICE=SIP/1156")
> in new stack
>
> -- Executing [...@default:3] Set("SIP/1156-55ce",
> "NULL="username=john.smith,name=John
> Smith,department=Dept_A,routable=no,extension=1234,device=SIP/1156,voicemail=no,monitor=yes,visible=yes,date_modified=2010-02-09
> 14:12:01,"") in new stack
>
> -- Executing [...@default:4] NoOp("SIP/1156-55ce", "DEVICE is
> SIP/1156") in new stack
>
> -- Executing [...@default:5] NoOp("SIP/1156-55ce", "USERNAME is ")
> in new stack
>
> -- Executing [...@default:6] NoOp("SIP/1156-55ce", "username is ")
> in new stack
>
>
>
> So I can see it is getting info from the database in Line 3
>
>
>
> But only the direct set variable command (Line 2) and Result (Line 4) work
>
>
>
> Lines 5 and 6 do not get the john.smith assigned
>
>
>
> Help
>
>
>
>
>
> Confidentiality Statement & Notice: This email is covered by the
> Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and
> intended only for the use of the individual or entity to whom it is
> addressed.  Any review, retransmission, dissemination to unauthorized
> persons or other use of the original message and any attachments is
> strictly prohibited. If you received this electronic transmission in error,
> please reply to the above-referenced sender about the error and
> permanently delete this message. Thank you for your cooperation.
>
>
> --
> _
> -- 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] Background noise

[khalid touati]

:) all users are having the same issue, even those connected to this server
from abroad!

2010/3/29 Philipp von Klitzing 

> > i have the same model polycom phone configured with another server
> > (asterisk 1.4), and guess what no noise at all. any guess!
>
> Replace the handset?
>
>
> --
> _
> -- 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
>



-- 
Abdullah
-- 
_
-- 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] dnd not working correctly

[Ott Rose]

Alyed,

I figured it was a freepbx issue but I have had no response on the post i put 
on their forum. I am going to look some place else  for freepbx support. I am 
not impressed at all with the freepbx support. On the other hand i have been 
getting answers to most all my post on this mail list in a day or two at the 
most.




Date: Fri, 26 Mar 2010 18:34:46 -0600
From: al...@vivoxie.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly

Seems like an Amportal configration problem not and Asterisk issue. Maybe you 
should try in one of the FreePBX users list.

Alyed



2010/3/26 Ott Rose 








i have posted this question couple of times and never really got any hits i 
wasn't able to provide any debug info 


Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)

Verbosity is at least 4

  == Using SIP RTP TOS bits 184

  == Using SIP RTP CoS mark 5

  == Using SIP VRTP TOS bits 136

  == Using SIP VRTP CoS mark 6

  == Extension Changed 117[ext-local] new state InUse for Notify User 102

  == Extension Changed 117[ext-local] new state InUse for Notify User 103

  == Extension Changed 117[ext-local] new state InUse for Notify User 114

-- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in new 
stack

-- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in new 
stack

-- Executing [...@from-internal:3] Macro("SIP/117-01f6", 
"user-callerid,") in new stack

-- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6", 
"AMPUSER=117") in new stack

-- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6", 
"0?report") in new stack

-- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6", 
"1?Set(REALCALLERIDNUM=117)") in new stack

-- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6", 
"AMPUSER=117") in new stack

-- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6", 
"AMPUSERCIDNAME=My Name") in new stack

-- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6", 
"0?report") in new stack

-- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6", 
"AMPUSERCID=117") in new stack

-- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6", 
"CALLERID(all)="My Name" <117>") in new stack

-- Executing [...@macro-user-callerid:9] GotoIf("SIP/117-01f6", 
"0?continue") in new stack

-- Executing [...@macro-user-callerid:10] Set("SIP/117-01f6", 
"__TTL=64") in new stack

-- Executing [...@macro-user-callerid:11] GotoIf("SIP/117-01f6", 
"1?continue") in new stack

-- Goto (macro-user-callerid,s,18)

-- Executing [...@macro-user-callerid:18] NoOp("SIP/117-01f6", "Using 
CallerID "My Name" <117>") in new stack

-- Executing [...@from-internal:4] GotoIf("SIP/117-01f6", 
"1?activate:deactivate") in new stack

-- Goto (from-internal,*76,5)

-- Executing [...@from-internal:5] Set("SIP/117-01f6", 
"DB(DND/117)=YES") in new stack

-- Executing [...@from-internal:6] Set("SIP/117-01f6", "STATE=BUSY") in 
new stack

-- Executing [...@from-internal:7] Gosub("SIP/117-01f6", 
"app-dnd-toggle,sstate,1") in new stack

-- Executing [sst...@app-dnd-toggle:1] Set("SIP/117-01f6", 
"DEVICE_STATE(Custom:DND117)=BUSY") in new stack

-- Executing [sst...@app-dnd-toggle:2] Set("SIP/117-01f6", 
"DEVICES=117") in new stack

-- Executing [sst...@app-dnd-toggle:3] GotoIf("SIP/117-01f6", 
"0?return") in new stack

  == Extension Changed 117[ext-local] new state Busy for Notify User 102

-- Executing [sst...@app-dnd-toggle:4] Set("SIP/117-01f6", "LOOPCNT=1") 
in new stack

-- Executing [sst...@app-dnd-toggle:5] Set("SIP/117-01f6", "ITER=1") in 
new stack

-- Executing [sst...@app-dnd-toggle:6] Set("SIP/117-01f6", 
"DEVICE_STATE(Custom:DEVDND117)=BUSY") in new stack

  == Extension Changed 117[ext-local] new state Busy for Notify User 103

  == Extension Changed 117[ext-local] new state Busy for Notify User 114

-- Executing [sst...@app-dnd-toggle:7] Set("SIP/117-01f6", "ITER=2") in 
new stack

-- Executing [sst...@app-dnd-toggle:8] GotoIf("SIP/117-01f6", 
"0?begin") in new stack

-- Executing [sst...@app-dnd-toggle:9] Return("SIP/117-01f6", "") in 
new stack

-- Executing [...@from-internal:8] Playback("SIP/117-01f6", 
"do-not-disturb&activated") in new stack

-- Executing [...@from-internal:9] Macro("SIP/117-01f6", "hangupcall,") 
in new stack

-- Executing [...@macro-hangupcall:1] GotoIf("SIP/117-01f6", 
"1?skiprg") in new stack

-- Goto (macro-hangupcall,s,4)

-- Executing [...@macro-hangupcall:4] GotoIf("SIP/117-01f6", 
"1?skipblkvm") in new stack

-- Goto (macro-hangupcall,s,7)

-- Executing [...@macro-hangupcall:7] GotoIf("SIP/117-01f6", 
"1?theend") in new stack

-- Goto (macro-hangupcall,s,9)

-- Executing [...@macro-hangupcall:9] H

Re: [asterisk-users] Background noise

[Philipp von Klitzing]

> i have the same model polycom phone configured with another server
> (asterisk 1.4), and guess what no noise at all. any guess! 

Replace the handset?


-- 
_
-- 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] Background noise

[khalid touati]

Hi!
so after using the "sip show channels" commands i can see that most of the
the communication are under ulaw format and one or two are under gsm, do you
guys know a way to force the system to use only ulaw, is it a good idea and
is it gonna solve my static noise issue?
actually, the caller and the called hear the static noise just when handset
is used, once i use the speaker it's gone.
actual situation:
-once i use the handset to dial i can here the noise.
-i have the same model polycom phone configured with another server
(asterisk 1.4), and guess what no noise at all.
any guess!


2010/3/27 khalid touati 

> Thank you very mutch Philip, i'll use these commands and get back with the
> output.
>
> 2010/3/26 Philipp von Klitzing 
>
> Hi!
>>
>> > it should be some commands that can give me a better idea about the
>> > codecs, if anyone know them, please help!
>>
>> Use "sip show channels" and "iax show channels" and look at the Format
>> column.
>>
>> About the Polycom devices: Others will have to help you there. I have no
>> good guess why you might have the issue only on speakerphone, but not in
>> handset mode. Could it maybe be some kind of electrical grounding issue
>> (instead of something caused by transcoding)?
>>
>> Philipp
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Abdullah
>



-- 
Abdullah
-- 
_
-- 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] Slightly more advanced dialling..

[Philipp von Klitzing]

Hi!

> I'm wondering if it is possible to ring X number of extensions 
> simultaneously, and each answered call can be handled with some code. 

You might want to explain what you are trying to do.

Dial() can handle this by using something like SIP/peer1&SIP/peer2
The first one that answers wins. Look at the Dial option M to run a macro 
after the call has been answered.

Also have a look at FollowMe() since it can do parallel calling.
Or read up how to create a bunch of .call files using System() and a 
script.

> I can do a huntgroup-esque way of dialling, but I want all the dialled
> numbers to be picked up. 

Do you mean to say: "I want all dialed numbers to keep on ringing until 
they are answered, regardless if the initial callers has already been 
taken care of by the first extensions that reacted"?

In the Asterisk world, and usually in the PBX world in general, pick up 
has specific and different meaning (see *8 or app_pickup).

Philipp


-- 
_
-- 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] Realtime Issue

[Jason Walker]

It seems that my realtime is not assigning channel variables correctly.

 

INFO

Asterisk 1.6.0.26

 

Exten.conf

exten => _X.,1,NoOp()

exten => _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})

exten => _X.,3,Set(NULL="${REALTIME(agents,device,${DEVICE})}")

exten => _X.,4,NoOp(DEVICE is ${DEVICE})

exten => _X.,5,NoOp(USERNAME is ${USERNAME})

exten => _X.,6,NoOp(username is ${username})

 

 

CLI 

 

-- Executing [...@default:1] NoOp("SIP/1156-55ce", "") in new
stack

-- Executing [...@default:2] Set("SIP/1156-55ce",
"DEVICE=SIP/1156") in new stack

-- Executing [...@default:3] Set("SIP/1156-55ce",
"NULL="username=john.smith,name=John
Smith,department=Dept_A,routable=no,extension=1234,device=SIP/1156,voice
mail=no,monitor=yes,visible=yes,date_modified=2010-02-09 14:12:01,"") in
new stack

-- Executing [...@default:4] NoOp("SIP/1156-55ce", "DEVICE is
SIP/1156") in new stack

-- Executing [...@default:5] NoOp("SIP/1156-55ce", "USERNAME is ")
in new stack

-- Executing [...@default:6] NoOp("SIP/1156-55ce", "username is ")
in new stack

 

So I can see it is getting info from the database in Line 3

 

But only the direct set variable command (Line 2) and Result (Line 4)
work

 

Lines 5 and 6 do not get the john.smith assigned

 

Help

 

 

Confidentiality Statement & Notice: This email is covered by the 
Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and 
intended only for the use of the individual or entity to whom it is 
addressed.  Any review, retransmission, dissemination to unauthorized 
persons or other use of the original message and any attachments is 
strictly prohibited. If you received this electronic transmission in error, 
please reply to the above-referenced sender about the error and 
permanently delete this message. Thank you for your cooperation.
-- 
_
-- 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] Trying to configure xorcom on Suse 11

[Tzafrir Cohen]

On Sun, Mar 28, 2010 at 09:16:48PM -0500, Tim Panton wrote:
> 
> On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote:
> 
> > On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
> >> I'm having trouble getting a xorcom set up.
> >> 
> >> A large part of the problem is that the box is a _long_ way away and 
> >> I can't get to/at it easily, so while I could probably fix this in a few
> >> hours if the machine were in front of me, I'm struggling over a slow
> >> unreliable laggy link. 
> >> 
> >> Ok, enough whining from me.
> >> 
> >> I have a new Xorcom plugged into the usb of a Suse 11 machine
> >> I built Dahdi from trunk (last thursday) 
> >> 
> >> # svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux
> >> # svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools
> >> 
> >> dahdihardware -v sees the box but no spans.
> > 
> > Generally '/etc/init.d/dahdi start' . Or more specifically,
> > 'dahdi_registration on' .
> > 
> > See also:
> > 
> >  
> > http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html#_installation_scenarios
> 
> 
> I've must be missing something here - this is what I see now.
> 
> sh-4.0# dahdi_hardware -v
> usb:001/020  xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
>  LABEL=[usb:X1037246]   connect...@usb-:00:1a.7-4 
> 
> Shouldn't I see spans ??? I think the box (I've never seen it, but I know 
> what I asked for) 
> has 8fxs+8fxo+2E1 . 

Yes, you should. Any relevant kernel messages?

If not: try:

  rmmod xpp_usb
  modprobe xpp_usb

What new messages do you then see in /var/log/messages ?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- 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] Slightly more advanced dialling..

[Andy Dixon]

Hi Danny,
Thats excellent, thank you. I have limited knowledge on writing AGI, but I
am always up for a challenge!

Thanks

Andy

On 29 March 2010 14:08, Danny Nicholas  wrote:

>  The built-in Dial command will not satisfy this requirement (first pickup
> terminates function).  You could do an AGI to do an Asynchronous dialing of
> the X extensions simultaneously (Although the realistic limit would probably
> be 5-10 extensions at once).
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Andy Dixon
> *Sent:* Monday, March 29, 2010 8:03 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Slightly more advanced dialling..
>
>
>
> Hello,
>
>
>
> I'm wondering if it is possible to ring X number of extensions
> simultaneously, and each answered call can be handled with some code.
>
>
>
> I can do a huntgroup-esque way of dialling, but I want all the dialled
> numbers to be picked up.
>
>
>
> I hope this makes sense.. If not please say..
>
>
>
> Many thanks!
>
>
>
> Andy
>
> --
> _
> -- 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] Slightly more advanced dialling..

[Danny Nicholas]

The built-in Dial command will not satisfy this requirement (first pickup
terminates function).  You could do an AGI to do an Asynchronous dialing of
the X extensions simultaneously (Although the realistic limit would probably
be 5-10 extensions at once).

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Dixon
Sent: Monday, March 29, 2010 8:03 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Slightly more advanced dialling..

 

Hello,

 

I'm wondering if it is possible to ring X number of extensions
simultaneously, and each answered call can be handled with some code.

 

I can do a huntgroup-esque way of dialling, but I want all the dialled
numbers to be picked up.

 

I hope this makes sense.. If not please say..

 

Many thanks!

 

Andy

-- 
_
-- 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] Slightly more advanced dialling..

[Andy Dixon]

Hello,

I'm wondering if it is possible to ring X number of extensions
simultaneously, and each answered call can be handled with some code.

I can do a huntgroup-esque way of dialling, but I want all the dialled
numbers to be picked up.

I hope this makes sense.. If not please say..

Many thanks!

Andy
-- 
_
-- 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] 24 FXS Port Voip Gateway and Asterisk

[John Novack]

Adtran also has a 10 year warranty on their products. Doesn't matter if 
you bought it off eBay for 30 bucks. Also excellent support. Something 
needs fixed, get an RMA you ship it to them, they ship back paid, often 
an exchange upgraded.

also there are cheap TE110 cards available on eBay, since they are no 
longer made by Digium.

John Novack


Steve Edwards wrote:
> On Sun, 28 Mar 2010, Joseph Begumisa wrote:
>
>> Can anyone recommend a 24 fxs port voip gateway that has worked well 
>> with asterisk?  I have a couple of analog handsets that I want to 
>> hookup to my asterisk server?  Any tested and tried product 
>> recommendations are welcome.  Thanks.
>
> Adtran channel banks are a great "trailing edge" technology. You can 
> get them off Ebay for pennies on the original dollar and they are 
> built like a tank.
>
> ("voip gateway" is not very specific. If you meant SIP or IAX, you 
> might want to specify which.)
>
> 
>
>
>
> Checked by AVG - www.avg.com 
> Version: 9.0.791 / Virus Database: 271.1.1/2774 - Release Date: 03/27/10 
> 15:32:00
>
>   

-- 
Dog is my co-pilot


-- 
_
-- 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 add custom CDR fields to MySQL

[Zeeshan Zakaria]

If I remember correctly you should do CDR(flavor)="cherry" and it should
work. I have added custom fields in my CDR table in the past and didn't need
triggers.

On 2010-03-29 3:40 AM, "Robert Price"  wrote:

Hello Alex,

I'm struggling with the same problem and, not wanting to modify the CDR
backend, I just put in a workaround in the form of a MySQL trigger.
I'll describe what I did in case it helps someone, though I'm very
inexperienced at making compound procedures in MySQL.

In my extensions.conf, I can do something like this:

  exten => s,1,Set(CDR(userfield)="flavor=cherry|color=maroon"

The result is that my CDR(userfield) is a pipe-delimited list of
key=value pairs.  In my MySQL 5.0.45 database, I have altered my cdr
table with a column called 'flavor' and another called 'color'.  I have
also created a trigger thus:

DELIMITER //

CREATE TRIGGER cdr_insert BEFORE INSERT ON cdr
FOR EACH ROW
  BEGIN
SET @numidx = LENGTH(NEW.userfield)
 - LENGTH(REPLACE(NEW.userfield, '|', '')) + 1;
SET @idx = 0;
WHILE @idx + 1 <= @numidx DO
  SET @idx = @idx + 1;
  SET @param = SUBSTRING_INDEX(
  SUBSTRING_INDEX(NEW.userfield, '|', @idx), '|', -1);
  SET @pos = LOCATE('=', @param);
  IF @pos > 0 THEN
SET @key = SUBSTRING(@param, 1, @pos - 1);
SET @value = SUBSTRING(@param, @pos + 1);

CASE @key
  WHEN 'flavor' THEN SET NEW.flavor = @value;
  WHEN 'color' THEN SET NEW.color = @value;
END CASE;

  END IF;
END WHILE;
SET NEW.userfield = '';
  END
//

DELIMITER ;

You can omit SET NEW.userfield = '' if you want to retain the userfield
to prevent data loss.  Expand the CASE statement as necessary to
enumerate all the fields you want to be able to specify via userfield,
and make sure you've created the appropriate columns beforehand.  You
can even specify things like 'dst' and 'dcontext', which you wouldn't
normally be able to control.  The loop silently ignores elements of the
pipe-delimited list that it does not recognize.  In particular, if the
value of CDR(userfield) begins with a pipe, that doesn't create a problem.

So far, it appears to work.

Cheers,
Robert

 > Hi all,
 >
 > I've been trying to add a custom mysql field to my CDR's, but I
 > must be doing something wrong.
 >
 > I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I
 > add:
 >
 > exten => h,1,Set(CDR(q931)=${HANGUPCAUSE})
 >
 > This extension is executed, I can see it in the asterisk console.
 >
 > I have added a new column in my MySQL database called q931.
 > However,
 > the new field does not show up in my database or in the Master.csv
 > file.
 >
 > Any help would be greatly appreciated.
 > Regards,
 >
 > Alex

--
_
-- 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] Continue a dialplan when the client hang up the call

[Zeeshan Zakaria]

Actually h extension is for hangup context, g within the dialplan command is
used for this purpose, so you can stay within the same context when the
other party hangs up the call, and execute further commands.

On 2010-03-29 6:18 AM, "huu giang"  wrote:

Thanks Ishfaq, h extension is the answer for my question :).

--- On *Mon, 3/29/10, huu giang * wrote:


From: huu giang 


> Subject: Re: [asterisk-users] Continue a dialplan when the client hang up
the call
> To: "Asteris...
Date: Monday, March 29, 2010, 2:52 AM


>
> Hi Ishfaq
>
>
> When Asterisk continue the dialplan, can it discover that the client has
hang u...
-Inline Attachment Follows-


>
> --
> _
> -- Bandwidth and ...



--
_
-- 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] Live Audio Streaming- From Aux interface-Online resource

[nik600]

On Fri, Mar 19, 2010 at 2:30 PM, Jonathan Addleman  wrote:

>
> If that doesn't work for some reason (In my case, I needed to stream
> through a flash applet on a web page, so it needed to be an mp3 stream),
> you can use an eagi that pipes through an encoder and then to your
> streaming software. In my case, I piped the audio through ffmpeg and
> then to ezstream which sent it to icecast.
>
> --
> Jon-o Addleman - http://www.redowl.ca
>
i'm looking for that, can you kindly give me a more detailed example?

I was trying to record a call usng Mixmonitor and then convert it
using ffmpeg but the recording file is continuosly growing and ffmpeg
ends the conversion before of the call completion.

If you can give me a practical example i'll appreciate it a lot.

Bye

-- 
/*/
nik600
http://www.kumbe.it

-- 
_
-- 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] queue autopause status

[Christian Gansberger]

hi all!

Does anybody know, how to get the status "autopaused" from queues.
I want to display the status to the agent.

I'm using asterisk-1.4.29.1

thanks
chris

-- 
_
-- 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] Continue a dialplan when the client hang up the call

[huu giang]

Thanks Ishfaq, h extension is the answer for my question :).

--- On Mon, 3/29/10, huu giang  wrote:

From: huu giang 
Subject: Re: [asterisk-users] Continue a dialplan when the client hang up the 
call
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Monday, March 29, 2010, 2:52 AM

Hi Ishfaq


When Asterisk continue the dialplan, can it discover that the client has hang 
up the call ?.
Is there any way ?.


--- On Mon, 3/29/10, Ishfaq Malik  wrote:

From: Ishfaq Malik 
Subject: Re: [asterisk-users] Continue a dialplan when the client hang up the 
call
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Monday, March 29, 2010, 2:34 AM

There is the h exten to deal with exactly what you want

http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension

huu giang wrote:
> Hi all,
>
> When a user make a call to Asterisk, and when user hang up the call at 
> any point of the conversation,  Asterisk will stop Diaplan 
> intermediately.
>
> At this situation,  Are there any way to make  Asterisk continue 
> execute the Diaplan ?, so Asterisk can do something like that delete 
> temporary file, .. etc.
>
> Thanks in advance,
> Giang
>
>

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

-- 
_
-- 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





  
-Inline Attachment Follows-

-- 
_
-- 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] Continue a dialplan when the client hang up the call

[huu giang]

Hi Ishfaq


When Asterisk continue the dialplan, can it discover that the client has hang 
up the call ?.
Is there any way ?.


--- On Mon, 3/29/10, Ishfaq Malik  wrote:

From: Ishfaq Malik 
Subject: Re: [asterisk-users] Continue a dialplan when the client hang up the 
call
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Monday, March 29, 2010, 2:34 AM

There is the h exten to deal with exactly what you want

http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension

huu giang wrote:
> Hi all,
>
> When a user make a call to Asterisk, and when user hang up the call at 
> any point of the conversation,  Asterisk will stop Diaplan 
> intermediately.
>
> At this situation,  Are there any way to make  Asterisk continue 
> execute the Diaplan ?, so Asterisk can do something like that delete 
> temporary file, .. etc.
>
> Thanks in advance,
> Giang
>
>

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

-- 
_
-- 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] Continue a dialplan when the client hang up the call

[Ishfaq Malik]

There is the h exten to deal with exactly what you want

http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension

huu giang wrote:
> Hi all,
>
> When a user make a call to Asterisk, and when user hang up the call at 
> any point of the conversation,  Asterisk will stop Diaplan 
> intermediately.
>
> At this situation,  Are there any way to make  Asterisk continue 
> execute the Diaplan ?, so Asterisk can do something like that delete 
> temporary file, .. etc.
>
> Thanks in advance,
> Giang
>
>

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

-- 
_
-- 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] MixMonitor and StopMixMonitor

[jonas kellens]

Hello list,

how does StopMixMonitor know which 'monitoring channel' to stop when
there are multiple conversations that are being monitored/recorded ??

I want to use StopMixMonitor in a macro, called from within
applicationmap (features.conf).


Jonas.
-- 
_
-- 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] Continue a dialplan when the client hang up the call

[huu giang]

Hi all,

When a user make a call to Asterisk, and when user hang up the call at any 
point of the conversation,  Asterisk will stop Diaplan intermediately. 


At this situation,  Are there any way to make  Asterisk continue execute the 
Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc.

Thanks in advance,
Giang




  -- 
_
-- 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 load balancing and failover

[huu giang]

I'm sorry, Anyone has experience in configuring Redfone to support 
load-balancing, please
 share with me? I can't find any guide about this feature from RedFone.

--- On Mon, 3/29/10, huu giang  wrote:

From: huu giang 
Subject: Re: [asterisk-users] Asterisk load balancing and failover
To: "Eric Wheeler" 
Cc: "asterisk-users" 
Date: Monday, March 29, 2010, 12:37 AM


Anyone has experience in configuring Redfone to support failover, please share 
with me?.



--- On Fri, 3/26/10, Eric Wheeler  wrote:

From: Eric Wheeler 
Subject: Re: [asterisk-users] Asterisk load balancing and failover
To: huugiang...@yahoo.com
Cc: "asterisk-users" 
Date: Friday, March 26, 2010, 9:12 AM


>If I have two Asterisk Servers, and each server has a TDM card and a > 
> PRI line connect to each card, how your solution can provide failover
>ability to Asterisk ? Do you need any other hardware?

Have a look at this article and how they shared a single
 T1 line across
two servers for failover:

http://www.linuxjournal.com/article/7661


(sorry about the missing in-reply-to header.  I'm not sure how to get
Evolution to add in-reply-to manually and I'm receiving messages in
digest form.)

-- 
Eric Wheeler
President
Portland Linux Support

www.PortlandLinuxSupport.com
503-330-4277
PO Box 86710
Portland, OR 97286




  
-Inline Attachment Follows-

-- 
_
-- 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 load balancing and failover

[huu giang]

Anyone has experience in configuring Redfone to support failover, please share 
with me?.



--- On Fri, 3/26/10, Eric Wheeler  wrote:

From: Eric Wheeler 
Subject: Re: [asterisk-users] Asterisk load balancing and failover
To: huugiang...@yahoo.com
Cc: "asterisk-users" 
Date: Friday, March 26, 2010, 9:12 AM


>If I have two Asterisk Servers, and each server has a TDM card and a > 
> PRI line connect to each card, how your solution can provide failover
>ability to Asterisk ? Do you need any other hardware?

Have a look at this article and how they shared a single T1 line across
two servers for failover:

http://www.linuxjournal.com/article/7661


(sorry about the missing in-reply-to header.  I'm not sure how to get
Evolution to add in-reply-to manually and I'm receiving messages in
digest form.)

-- 
Eric Wheeler
President
Portland Linux Support

www.PortlandLinuxSupport.com
503-330-4277
PO Box 86710
Portland, OR 97286




  -- 
_
-- 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] Restarting Asterisk using a script - Thanks to all -

[Aurimas Skirgaila]

Hi,

1) It would be nice to find out the root reason that forces you to
restart the Asterisk. I do run Aheeva with decently high uptimes.

2) Both "a" and "b" methods of Jose P. Espinal are functional, but if
I'm having a failure, I up to grab the putty and investigate what's
going on there :) How often are you having them?

3) what's the another service related to Aheeva, that requires to be restarted?



>
> Hi there,


a. You could (maybe) use PHP and send some command via POST, and (after

secure/validating the command) use 'exec()' function in php, or


'system()' function.


Note: that would require to have a webserver with php installed on it.

And allowing the user under which the webserver runs, to restart


asterisk via sudoers file.


b. You could use a shellscript that sends the command via SSH.


In order to avoid password prompt, you could generate a RSA (or DSA) key


pair on the machine that will send the command, and copy the rsa_key.pub

content on your asterisk box 'authorized_keys'.


That would allow you to execute the command remotely via SSH without


having to insert the password manually.


Note: you could consider using a very limited user on the asterisk box,

and with sudoers file allowing it just to restart Asterisk.




Regards,


Amine Mrichcha wrote:

> Hi All,

>


> I do have asterisk installed for a call center and I would like to know

> if it is possible to create a scipt and execute it from a PC connected

> to the Network without accessing the server. This script should restart


> asterisk and another service related to aheeva.

>

> The problem now is that each time I have to access using PUTY to the

> server to start and run services manually.


>

> Service asterisk restart

>

> Any help would be appreciated, sorry if it is a newbie question.


>

> Regards,

>

> Am

>


-- 

Jose P. Espinal


http://www.eSlackware.com 

IRC: Khratos @ #asterisk / -doc / -bugs





-- 
Mvh,
Aurimas Skirgaila
-- 
_
-- 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] is it possible to connect Digium TE420 and Cisco card?

[Aurimas Skirgaila]

Hello,

I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to
providers Cisco 2800 with  VWIC-1MFT-E1 card.

the same card runs fine with another E1 provider.

TE420 led's lite green.

 Message type: RELEASE COMPLETE (90)
< [08 02 80 ac]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
 Location: User (0)
<  Ext: 1  Cause: Requested channel not available (44),
class = Network Congestion (resource unavailable) (2) ]
-- Processing IE 8 (cs0, Cause)
q931.c:3760 q931_receive: call 32770 on channel 1 enters state 0 (Null)
Sending Receiver Ready (31)

-- Channel 0/1, span 1 got hangup, cause 44
-- Forcing restart of channel 0/1 on span 1 since channel reported in
use
q931.c:3000 q931_restart: call 32768 on channel 1 enters state 62 (Restart)

[zaptel.conf]
span=1,1,0,ccs,hdb3,crc4#switching timing between 0/1 does not have any
effect,
bchan=1-15,17-31
dchan=16

[zapata.conf]
group=1
pridialplan = unknown
switchtype=euroisdn
context = trunk-1
signalling = pri_net
channel => 1-15,17-31

Hardware - Dell PowerEdge R200. Now moved onto barebone test server, but
same errors persist.

-- 
Mvh,
Aurimas Skirgaila
-- 
_
-- 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] How to add custom CDR fields to MySQL

[Robert Price]

Hello Alex,

I'm struggling with the same problem and, not wanting to modify the CDR 
backend, I just put in a workaround in the form of a MySQL trigger. 
I'll describe what I did in case it helps someone, though I'm very 
inexperienced at making compound procedures in MySQL.

In my extensions.conf, I can do something like this:

   exten => s,1,Set(CDR(userfield)="flavor=cherry|color=maroon"

The result is that my CDR(userfield) is a pipe-delimited list of 
key=value pairs.  In my MySQL 5.0.45 database, I have altered my cdr 
table with a column called 'flavor' and another called 'color'.  I have 
also created a trigger thus:

DELIMITER //

CREATE TRIGGER cdr_insert BEFORE INSERT ON cdr
FOR EACH ROW
   BEGIN
 SET @numidx = LENGTH(NEW.userfield)
  - LENGTH(REPLACE(NEW.userfield, '|', '')) + 1;
 SET @idx = 0;
 WHILE @idx + 1 <= @numidx DO
   SET @idx = @idx + 1;
   SET @param = SUBSTRING_INDEX(
   SUBSTRING_INDEX(NEW.userfield, '|', @idx), '|', -1);
   SET @pos = LOCATE('=', @param);
   IF @pos > 0 THEN
 SET @key = SUBSTRING(@param, 1, @pos - 1);
 SET @value = SUBSTRING(@param, @pos + 1);

 CASE @key
   WHEN 'flavor' THEN SET NEW.flavor = @value;
   WHEN 'color' THEN SET NEW.color = @value;
 END CASE;

   END IF;
 END WHILE;
 SET NEW.userfield = '';
   END
//

DELIMITER ;

You can omit SET NEW.userfield = '' if you want to retain the userfield 
to prevent data loss.  Expand the CASE statement as necessary to 
enumerate all the fields you want to be able to specify via userfield, 
and make sure you've created the appropriate columns beforehand.  You 
can even specify things like 'dst' and 'dcontext', which you wouldn't 
normally be able to control.  The loop silently ignores elements of the 
pipe-delimited list that it does not recognize.  In particular, if the 
value of CDR(userfield) begins with a pipe, that doesn't create a problem.

So far, it appears to work.

Cheers,
Robert

 > Hi all,
 >
 > I've been trying to add a custom mysql field to my CDR's, but I
 > must be doing something wrong.
 >
 > I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I
 > add:
 >
 > exten => h,1,Set(CDR(q931)=${HANGUPCAUSE})
 >
 > This extension is executed, I can see it in the asterisk console.
 >
 > I have added a new column in my MySQL database called q931.
 > However,
 > the new field does not show up in my database or in the Master.csv
 > file.
 >
 > Any help would be greatly appreciated.
 > Regards,
 >
 > Alex

-- 
_
-- 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