gmane.comp.telephony.pbx.asterisk.user

Re: Mailing a fax with mutt does not succeed

Mikhail Lischuk — 2013-06-19T21:58:55

 

This worked for me: 

System(date | mutt -s "FAX from
${CALLERID(num)}" -a
/var/spool/asterisk/fax/${STRFTIME(,,%Y%m%d)}-${CALLERID(num)}.tiff
mail< at >domain) 

I don't see any difference with you (beside the echo
instead of date), so I guess you should look at the maillog to find out
what is happening. Or (as I did in the end) write AGI script to send the
goddamn mail using the language and method you like (I used Perl and
MIME::Lite::TT::HTML), and make Asterisk call that script. 

jg
писал 19.06.2013 23:12: 

any entry in /var/log/maillog (or equivalent log file)? If so, mutt
basically works and the messages should give some clues.
happens if you call mutt without any attachments?
in exactly the same way and it works.
schrieb Daniel - Asterisk: 
provide STDIN, I'm sure on variable contents, please see bellow 
Hello Steve, 
output bellow. 
has access to read TIF files since I've used ls, chmod, cp & mv from
Asterisk's CLI with '!' character. 
some advice to try using Verbose instead System 
to get thi

Re: Mailing a fax with mutt does not succeed

jg — 2013-06-19T20:12:04

More things to try:

(1) Is there any entry in /var/log/maillog (or equivalent log file)? If so, mutt basically works 
and the messages should give some clues.
(2) What happens if you call mutt without any attachments?

I am using mutt in exactly the same way and it works.

jg

Am 19.06.2013 21:50, schrieb Daniel - Asterisk:

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Re: Mailing a fax with mutt does not succeed

Daniel - Asterisk — 2013-06-19T19:50:30

Hi Andre:

I added echo to provide STDIN, I'm sure on variable contents, please see
bellow


Hello Steve,

1. I've just addd echo at my sentence, please see output bellow.
2. Asterisk is executing as root, I think Asterisk has access to read TIF
files since I've used ls, chmod, cp & mv from Asterisk's CLI with '!'
character.
3. I don't get you, please give some advice to try using Verbose instead
System
4. I don't know how to get this, but I'm using /usr/bin/mutt as you can see
bellow.
5. I have redirected output of System this way : System(echo |
/usr/bin/mutt -s "New fax" earohuanca< at >gmail.com -a ${FAXDEST}/${tempfax} >
/tmp/ocurrencies.txt 2>&1), ocurrencies.txt is empty.


DIALPLAN:
[ Context 'default' created by 'pbx_config' ]
  '*95' =>          1. NoOp(trying to send a fax to an email)
                    2. Set(FAXDEST=/tmp/faxes)
                    3. Set(tempfax=${SHELL(ls /tmp/faxes/*.tif):11})
                    4. NoOp(file name is: ${tempfax})
                    5. Goto(incoming-fax,fax,7)

[

Re: Mailing a fax with mutt does not succeed

Andre Courchesne — 2013-06-19T18:38:50

Why "echo |" ?

Alsy are you sire of the content of ${FAXDEST} and ${tempfax}.

Add some NoOp before.


On 2013-06-19, at 2:29 PM, Daniel - Asterisk <earohuanca< at >gmail.com> wrote:


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Re: Mailing a fax with mutt does not succeed

Daniel - Asterisk — 2013-06-19T18:29:25

Hi Andre,

I've tried with:
System(echo | /usr/bin/mutt -s "New fax" earohuanca< at >gmail.com -a
${FAXDEST}/${tempfax})

with no success, value of SYSTEMSTATUS variable is APPERROR

Again it works from Linux shell.

Thanks in advance

Elder


On Wed, Jun 19, 2013 at 1:08 PM, Andre Courchesne <voipforces< at >gmail.com>wrote:

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Re: Mailing a fax with mutt does not succeed

Steve Edwards — 2013-06-19T18:28:22


1) Doesn't mutt expect the body on stdin? (Where's the 'echo' in the 
Asterisk command?)

2) Is Asterisk executing as root? Does the Asterisk user ID have read 
access to the TIFF?

3) If you use 'verbose()' instead of 'system()' does the command look like 
your shell command?

4) Is mutt in the Asterisk user ID's path?

5) If you redirect the output in the system() command to a file, does that 
yield any clues? I.e., system(foo >/tmp/clue 2>&1)

Re: Mailing a fax with mutt does not succeed

Andre Courchesne — 2013-06-19T18:08:40

Probably Asterisk does not know where mutt is, specify it's path in your System command.

On 2013-06-19, at 2:03 PM, Daniel - Asterisk <earohuanca< at >gmail.com> wrote:


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Mailing a fax with mutt does not succeed

Daniel - Asterisk — 2013-06-19T18:03:14

Hello everyone,

I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.

Successful Linux command:
echo | mutt -s "New fax" earohuanca< at >gmail.com -a /tmp/faxes/201306191111.tif

Unsuccessful Asterisk Command:
same => n,System(mutt -s "New fax" elder.arohuanca< at >gmail.com -a
${FAXDEST}/${tempfax}.tif)

I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by root.

Any hint will be appreciated.

Elder D. Arohuanca
Lima - Peru
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Re: SIP Simple support on Asterisk 11

Eloi Bail — 2013-06-19T17:58:08

Hi,

Thanks a lot for this detailed answer :

- I managed to have it working disabling auth message request
: auth_message_requests = no in sip.conf
- pedantic=no does not resolve the issue
- reenabling  auth_message_requests = yes and removing pedantic option,
your patch in chan_sip resolves the issues !

As it looks like pidgin has an issue, I guess that we can use it as a
workaround.

I would like know to enable presence notification between each users. To
fulfill it, I am using
http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265377.html

Am I doing it in a good way ?

Thanks !

Eloi


On Wed, Jun 19, 2013 at 12:11 PM, Matthew J. Roth <mroth< at >imminc.com> wrote:

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Re: SIP Simple support on Asterisk 11

Anthony Messina — 2013-06-19T16:34:54

I use the following which semi-enables message broadcasting to multiple 
devices so a user who receives a message can reply from any of the devices.

http://messinet.com/trac/wiki/Asterisk/Message

-A

Re: SIP Simple support on Asterisk 11

Matthew J. Roth — 2013-06-19T16:11:17


Eloi,

The trace shows that the initial MESSAGE from Alice does not include an
Authorization header so Asterisk responds with a 401 Unauthorized.  Alice then
replies with a MESSAGE with an Authorization header, but reuses the same CSeq
header (CSeq: 6 MESSAGE) which causes Asterisk to ignore it as a retransmit:


I believe this is a bug in Pidgin because RFC 3261 [1] states:

   CSeq or Command Sequence contains an integer and a method name.  The
   CSeq number is incremented for each new request within a dialog and
   is a traditional sequence number.
   ...
   Requests within a dialog MUST contain strictly monotonically
   increasing and contiguous CSeq sequence numbers (increasing-by-one)
   in each direction (excepting ACK and CANCEL of course, whose numbers
   equal the requests being acknowledged or cancelled).

However, there is also a similar issue [2] that can be worked around by setting
"pedantic=no" in sip.conf.  If that doesn't work, you can give the following
(untested) patch to chan_sip.c a t

Re: PCI Passthrough of T1 cards

Nick Khamis — 2013-06-19T15:52:11

Hello James,

Thank you so much for your response. I should have chose my words
carefully. PCI pass-through in terms of virtualization of devices and
it's draw back are well know. I was leaning more towards near host
performance virtualization using SR-IOV.

This moves emphasis back to the production drivers of the interface
card using virtual functions etc., and can provide near host
performance. Rephrasing my question, are any of the T1 pci
manufactures providing support for virtualization using SR-IOV and
virutal functions?

Kind Regards,

Nick

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Re: PCI Passthrough of T1 cards

James Sharp — 2013-06-19T15:26:54

On Jun 16, 2013, at 4:27 PM, Nick Khamis <symack< at >gmail.com> wrote:


PCI pass through is a function of the virtual machine's host system, not with the t1 card drivers. 
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SIP Simple support on Asterisk 11

Eloi Bail — 2013-06-19T13:44:33

Hi all,


I am trying to enable SIP SIMPLE communication in my test environment.

I have the following env :

- one server (192.168.50.126) with Asterisk 11
- 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143

I successfully had a phone call between clients.

I used the following link to enable SIMPLE messaging between my clients :
http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-or-sip.html

Both users managed to register.

Adding verbose on the server, I have the following traces when I send the
message "MESSAGE FROM ALICE TO BOB" from "demo-alice" to "demo-bob"

http://paste.fedoraproject.org/19489/37158861/

As you can see I succeed to have the message sent from alice to Asterisk.

When the server is trying to transmitting, I see a 401 error message.
According to this post (http://forums.digium.com/viewtopic.php?f=1&t=72814)
the first 401 should be normal as authentication is requested.

Afterwards the server emit 202 message.

But "demo-bob" never receives a mess

Re: Handoff dial control to dialplan after AMIOriginate

Muhammad Faheem — 2013-06-19T12:50:06

Your both channels legs are identical strings. It should be like this.

Action: Originate****

Channel: Local/outbound1< at >originateDialContext****

CallerID: 00311234567****

Context: originateDialContext2****

Exten: outbound1****

Priority: 1****

Variable: recipient=0031612345678,callerid1=****00311234567

Timeout: 10000

** **

[originateDialContext]****

exten => outbound1,1,Wait(1)****

exten => outbound1,n,Set(recipient=${recipient})****

exten => outbound1,n,Dial(SIP/${recipient}< at >originateChannel)****

[originateDialContext2]****

exten => outbound1,1,Wait(1)****

exten => outbound1,n,Dial(SIP/${callerid1}< at >originateChannel)



On Wed, Jun 19, 2013 at 11:20 AM, Grant Bagdasarian <GB< at >cm.nl> wrote:

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Re: Queue Limit Callers

Shanavaz E A — 2013-06-19T12:11:51

Thanks all for the inputs... Let me work on it and come back again with some results...

________________________________
 From: Ioan Indreias <indreias< at >gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users< at >lists.digium.com> 
Sent: Tuesday, June 18, 2013 1:43 PM
Subject: Re: [asterisk-users] Queue Limit Callers

Hello Shanavaz.,

Please find some quick thoughts:

* 2 main queues
* agents logged on one or on both main queues
* before sending a new call to one of the main queues check the number of waiting callers (QUEUE_WAITING_COUNT function) and divert (for example for 30 sec) the call on a empty members queue/parking slot/music-on-hold if the queue threshold is reached.

The threshold could be read from a database, internal astdb or could be set as a global variable updated when agents login/logout/pause/unpause or could be dynamically computed based on QUEUE_MEMBER_COUNT / QUEUE_MEMBER_LIST

After the divert period is ended the call will return and the threshold

Re: Handoff dial control to dialplan afterAMIOriginate

Grant Bagdasarian — 2013-06-19T12:08:17

I used a combination of failed and the h extension to get the dialplan to do what I want.

Thanks for the help guys!

From: asterisk-users-bounces< at >lists.digium.com [mailto:asterisk-users-bounces< at >lists.digium.com] On Behalf Of Satish Barot
Sent: Wednesday, June 19, 2013 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate

On Wed, Jun 19, 2013 at 4:00 PM, Grant Bagdasarian <GB< at >cm.nl<mailto:GB< at >cm.nl>> wrote:
Why can't I execute any more dialplan after the Dial application? The scenario is when the Dial application dials the recipient but the recipient doesn't answer. The AMI will never go into the originateDialProcessor because the call was never answered. So I expect the Dialplan to continue after the Dial application has reached its timeout.

From: asterisk-users-bounces< at >lists.digium.com<mailto:asterisk-users-bounces< at >lists.digium.com> [mailto:asterisk-users-bounces< at >lists.digium.com<mailto:asterisk-users-bounces

Re: Handoff dial control to dialplan after AMIOriginate

Satish Barot — 2013-06-19T11:24:33


You need a special extension 'failed' in a context originateDialProcessor
to catch the control when call doesn't get answered in first leg.

[originateDialProcessor]

exten => outbound1,1,Wait(1)

exten => outbound1,n,NoOp(${DIALSTATUS})
exten => outbound1,n,Hangup

exten => failed,1,NoOp(----- CALL DIDN'T GET ANSWERED IN FIRST LEG -----)

--Satish Barot
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Re: Handoff dial control to dialplanafterAMIOriginate

jg — 2013-06-19T10:39:09

Did you specify a timeout value?

jg

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Re: Handoff dial control to dialplanafterAMIOriginate

Grant Bagdasarian — 2013-06-19T10:30:35

Why can’t I execute any more dialplan after the Dial application? The scenario is when the Dial application dials the recipient but the recipient doesn’t answer. The AMI will never go into the originateDialProcessor because the call was never answered. So I expect the Dialplan to continue after the Dial application has reached its timeout.

From: asterisk-users-bounces< at >lists.digium.com [mailto:asterisk-users-bounces< at >lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: Wednesday, June 19, 2013 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI Originate

I fixed it. The problem is just as I assumed; once the call is answered the dialplan goes into what’s defined in Context/Exten/Prio of the Originate action.

I changed the Context/Exten/Prio in the action and pointed it to something else. Now it works.

Action: Originate
Channel: Local/outbound1< at >originateDialContext
CallerID: 00311234567
Context: o

Re: Handoff dial control to dialplan afterAMIOriginate

Grant Bagdasarian — 2013-06-19T09:24:06

I fixed it. The problem is just as I assumed; once the call is answered the dialplan goes into what’s defined in Context/Exten/Prio of the Originate action.

I changed the Context/Exten/Prio in the action and pointed it to something else. Now it works.

Action: Originate
Channel: Local/outbound1< at >originateDialContext
CallerID: 00311234567
Context: originateDialProcessor
Exten: outbound1
Priority: 1
Variable: recipient=0031612345678
Timeout: 10000

[originateDialContext]
exten => outbound1,1,Wait(1)
exten => outbound1,n,Set(recipient=${recipient})
exten => outbound1,n,Dial(SIP/${recipient}< at >originateChannel)

[originateDialProcessor]
exten => outbound1,1,Wait(1)
exten => outbound1,n,NoOp(${DIALSTATUS})
exten => outbound1,n,Hangup

From: asterisk-users-bounces< at >lists.digium.com [mailto:asterisk-users-bounces< at >lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, June 19, 2013 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hando