Re: [asterisk-users] Grandstream VoIP phones

[Vladimir Mikhelson]

Bryant,

Thank you for the reply.

It looks like either I was very unlucky with the support engineer my SR
was assigned to or you were extremely lucky.  Or maybe Grandstream
singles you or your company out for some reason.

My test is plain vanilla.

 1. Enable SIPS and SRTP for an extension in Asterisk 1.8.15
 2. Sign a certificate on the Asterisk server and provision it manually
to the DP715
 3. Try calling back and forth.

My plan was to spend 30 minutes to an hour to test the above and then
move to the real-life scenarios.  So far I spent 9 days, with no help
from Grandstream whatsoever, toying with this test and making no progress.

The features they must have for real-life deployments:

  * HTTPS on the setup portal with normal set of credentials, i.e. user
name and password
  * Ability to disable HTTP/HTTPS
  * SSH vs telnet
  * Ability to send host name or other CN not equal to the phone IP in
TLS negotiation

I will probably have more after I am past my step 0 testing.

Thank you,
Vladimir



On 8/31/2012 8:55 PM, Bryant Zimmerman wrote:
> Vladimir
>
> We are testing the DP715 very aggressively. We have been please with
> the units for the most part, but we too have been working bugs with
> Grandstream. We have several in so far and a number of feature
> requests as well. I deal directly with several of the support
> engineers and they bring in the developers when necessary. I would be
> open to working with you on your issue. If I can create validation
> tests for your items and reproduce the issue I have had great success
> getting them to take note and address issues they really do want to
> address issues. In less than two weeks they have given me test builds
> address two of our issues and they are working on several others.
> Because of the cooperation of Grandsteam we are close to being able to
> offer the DP715 phones to our customers. Even then they will have more
> items to address to allow for full feature deployments but they are
> serious about the DP715 product.
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
>
> 
> *From*: "Vladimir Mikhelson" 
> *Sent*: Friday, August 31, 2012 9:07 PM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> *Subject*: Re: [asterisk-users] Grandstream VoIP phones
>
> Carlos,
>
> So far the experience with DP715 is extremely negative.
>
> It all starts with the WEB interface which is only served on port 80,
> no https, period.  There is no login name, just password.
>
> The phone worked as expected with insecure SIP and RTP.  As I started
> playing with security the phone started acting up.  It randomly took
> calls, then stopped.  It placed calls, then stopped.
>
> Following is a sample of a corrupted SIP message Asterisk receives
> from DP715 (pay attention to Call-ID: 477744485-506...@bhc.bh.bdh.hb):
>
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 0 [ 14]: SIP/2.0
> 200 OK
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 1 [ 69]: Via:
> SIP/2.0/TLS 172.17.137.11:5061;branch=z9hG4bK2f5ce157;rport=5061
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 2 [ 57]: From:
> ;tag=as50c4dc59
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 3 [ 54]: To:
> ;tag=436538044
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 4 [ 39]:
> Call-ID: 477744485-506...@bhc.bh.bdh.hb
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 5 [ 13]: CSeq:
> 102 BYE
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 6 [ 51]:
> Contact: 
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 7 [ 43]:
> Supported: replaces, path, timer, eventlist
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 8 [ 37]:
> User-Agent: Grandstream DP715 1.0.0.5
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 9 [ 80]: Allow:
> INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 10 [ 17]:
> Content-Length: 0
>
> According to RFC 3261, "Call-ID contains a globally unique identifier
> for this call, generated by the combination of a random string and the
> softphone's host name or IP address."
>
> Interestingly, the problem is intermittent. Some calls go through. 
> Asterisk must be able to process these calls from time to time.  Which
> is strange on its own.
>
> On top of everything Grandstream's support organization does not seem
> to exist for all practical purposes.  I opened the case on
> 08/22/2012.  Today, 08/31/2012, I finally received a response, "Sorry
> for missing your call yesterday. We checked the syslog you sent to us
> and seems the TLS is shut down. I just got some TLS internal test
> accounts today and will do a quick test. I'll let you know soon.  It
> took them 9 days to start looking into the issue.
>
> I will update this thread with progress.
>
> Regards,
> Vladimir
>
>
>
> On 8/17/2012 11:30 AM, Carlos Alvarez wrote:
>> On Fri, Aug 17, 2012 at 9:08 

Re: [asterisk-users] Grandstream VoIP phones

[Bryant Zimmerman]

Vladimir

We are testing the DP715 very aggressively. We have been please with the 
units for the most part, but we too have been working bugs with 
Grandstream. We have several in so far and a number of feature requests as 
well. I deal directly with several of the support engineers and they bring 
in the developers when necessary. I would be open to working with you on 
your issue. If I can create validation tests for your items and reproduce 
the issue I have had great success getting them to take note and address 
issues they really do want to address issues. In less than two weeks they 
have given me test builds address two of our issues and they are working on 
several others. Because of the cooperation of Grandsteam we are close to 
being able to offer the DP715 phones to our customers. Even then they will 
have more items to address to allow for full feature deployments but they 
are serious about the DP715 product. 

Thanks

Bryant Zimmerman (ZK Tech Inc.)  


 From: "Vladimir Mikhelson" 
Sent: Friday, August 31, 2012 9:07 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Grandstream VoIP phones

Carlos,

So far the experience with DP715 is extremely negative.

It all starts with the WEB interface which is only served on port 80, no 
https, period.  There is no login name, just password.

The phone worked as expected with insecure SIP and RTP.  As I started 
playing with security the phone started acting up.  It randomly took calls, 
then stopped.  It placed calls, then stopped.

Following is a sample of a corrupted SIP message Asterisk receives from 
DP715 (pay attention to Call-ID: 477744485-506...@bhc.bh.bdh.hb):

[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 
OK
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 1 [ 69]: Via: 
SIP/2.0/TLS 172.17.137.11:5061;branch=z9hG4bK2f5ce157;rport=5061
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 2 [ 57]: From: 
;tag=as50c4dc59
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 3 [ 54]: To: 
;tag=436538044
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 4 [ 39]: Call-ID: 
477744485-506...@bhc.bh.bdh.hb
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 5 [ 13]: CSeq: 102 
BYE
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 6 [ 51]: Contact: 

[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 7 [ 43]: Supported: 
replaces, path, timer, eventlist
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 8 [ 37]: User-Agent: 
Grandstream DP715 1.0.0.5
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 9 [ 80]: Allow: 
INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 10 [ 17]: 
Content-Length: 0

According to RFC 3261, "Call-ID contains a globally unique identifier for 
this call, generated by the combination of a random string and the 
softphone's host name or IP address."

Interestingly, the problem is intermittent. Some calls go through.  
Asterisk must be able to process these calls from time to time.  Which is 
strange on its own.

On top of everything Grandstream's support organization does not seem to 
exist for all practical purposes.  I opened the case on 08/22/2012.  Today, 
08/31/2012, I finally received a response, "Sorry for missing your call 
yesterday. We checked the syslog you sent to us and seems the TLS is shut 
down. I just got some TLS internal test accounts today and will do a quick 
test. I'll let you know soon.  It took them 9 days to start looking into 
the issue.

I will update this thread with progress.

Regards,
Vladimir

On 8/17/2012 11:30 AM, Carlos Alvarez wrote:
  On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson  
wrote:
 My primary interest is security.  Grandstream claims their intermediate 
and higher-end models support TLS and SRTP.  I am really tired of trying to 
make Cisco phones to communicate securely with Asterisk.  Cisco has a great 
security model but one has to have their provisioning server for it to 
function.

 We've never had customers ask for this, but if doing so is fairly easy we 
would look at it as just another feature we push.  Do let me know how it 
works out for you. 
  -- 
Carlos Alvarez TelEvolve 602-889-3003 

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Re: [asterisk-users] Grandstream VoIP phones

[Vladimir Mikhelson]

Carlos,

So far the experience with DP715 is extremely negative.

It all starts with the WEB interface which is only served on port 80, no
https, period.  There is no login name, just password.

The phone worked as expected with insecure SIP and RTP.  As I started
playing with security the phone started acting up.  It randomly took
calls, then stopped.  It placed calls, then stopped.

Following is a sample of a corrupted SIP message Asterisk receives from
DP715 (pay attention to Call-ID: 477744485-506...@bhc.bh.bdh.hb):

[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 0 [ 14]: SIP/2.0
200 OK
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 1 [ 69]: Via:
SIP/2.0/TLS 172.17.137.11:5061;branch=z9hG4bK2f5ce157;rport=5061
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 2 [ 57]: From:
;tag=as50c4dc59
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 3 [ 54]: To:
;tag=436538044
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 4 [ 39]: Call-ID:
477744485-506...@bhc.bh.bdh.hb
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 6 [ 51]: Contact:

[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 7 [ 43]:
Supported: replaces, path, timer, eventlist
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 8 [ 37]:
User-Agent: Grandstream DP715 1.0.0.5
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 9 [ 80]: Allow:
INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 10 [ 17]:
Content-Length: 0

According to RFC 3261, "Call-ID contains a globally unique identifier
for this call, generated by the combination of a random string and the
softphone's host name or IP address."

Interestingly, the problem is intermittent. Some calls go through. 
Asterisk must be able to process these calls from time to time.  Which
is strange on its own.

On top of everything Grandstream's support organization does not seem to
exist for all practical purposes.  I opened the case on 08/22/2012. 
Today, 08/31/2012, I finally received a response, "Sorry for missing
your call yesterday. We checked the syslog you sent to us and seems the
TLS is shut down. I just got some TLS internal test accounts today and
will do a quick test. I'll let you know soon.  It took them 9 days to
start looking into the issue.

I will update this thread with progress.

Regards,
Vladimir



On 8/17/2012 11:30 AM, Carlos Alvarez wrote:
> On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson
> mailto:v...@mikhelson.com>> wrote:
>
> My primary interest is security.  Grandstream claims their
> intermediate and higher-end models support TLS and SRTP.  I am
> really tired of trying to make Cisco phones to communicate
> securely with Asterisk.  Cisco has a great security model but one
> has to have their provisioning server for it to function.
>
>
> We've never had customers ask for this, but if doing so is fairly easy
> we would look at it as just another feature we push.  Do let me know
> how it works out for you.
>
> -- 
> Carlos Alvarez
> TelEvolve
> 602-889-3003
>
>
>
>
> --
> _
> -- 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

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[asterisk-users] Record calls as BLOB into MySQL?

[Antoine Megalla]

Why not use use voicemail odbc storage engine and it will store your voicemail 
files is MySQL as BLOBs

Sent from my iPhone

On Aug 31, 2012, at 7:00 PM, asterisk-users-requ...@lists.digium.com wrote:

> [asterisk-users] Record calls as BLOB into MySQL?

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Re: [asterisk-users] Receiving and processing unsolicited XMPP messages with Asterisk 11

[Anthony Messina]

On Friday, August 31, 2012 06:48:46 PM Noah Engelberth wrote:
> I’m trying to set up a way that our users can send an XMPP message to
> Asterisk (unsolicited) to request information, such as voicemail status or
> the like.  No matter what I set for the dialplan, I’m only seeing Asterisk
> execute the s,1 priority in the context defined in xmpp.conf for incoming
> messages, and then the “call” hangs up without executing further
> instructions.  Anything I’ve tried to accomplish in that first priority has
> worked, but it never continues to an additional priority.

This might be a separate, but related issue, as I am not using XMPP messaging
yet, but I found that at least with SIP messaging in Asterisk 11, if I had a
Hangup() in the dialplan for message routing, every message sent AFTER the
first would fail just as you describe, since the first message routed through
the dialplan hung up the channel.

This did not happen to me in Asterisk 10.  After removing the traditional
Hangup() at the end, and restarting Asterisk, the messages route properly for
me.  -A

--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] FAX detection in chan_dahdi 1.8.15

[Jeff LaCoursiere]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
> LaCoursiere
> Sent: Tuesday, August 28, 2012 3:24 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] FAX detection in chan_dahdi 1.8.15
> 
> Hi,
> 
> I recently replaced a site that was using 1.4.[mumble] with
> hylafax/iaxmodem.  They have an RBS T1 and were using about half of their 50
> DID numbers for "fax to email".  This all broke with the new system :(
> 
> The original chan_dahdi.conf had no mention of "faxdetect", so I assume it
> was operating with whatever is the default.  Off?
> 
> The new box originally had "faxdetect=no", and I found that all my test
> faxes failed with negotiation errors.  When I finally tried
> "faxdetect=incoming" test faxes from another machine running hylafax went
> through fine, and I thought I was done.
> 
> The following week the customer reported that inbound faxes weren't working.
> When I looked at the log, I saw lots of these:
> 
> chan_dahdi.c: -- Redirecting DAHDI/24-1 to fax extension
> 
> Which in my FreePBX setup eventually goes to a "no service" message and
> hangs up.  I've never defined a "fax" extension and don't really know what
> that is about.  Turns out that any fax machine that calls ends up following
> this path.  If my other hylafax server calls, it follows the normal path and
> gets answered by my pool of iaxmodems. I don't really understand the
> difference between the two types of calls, first of all.
> 
> So it seems from this experience and a recent thread on -users that enabling
> faxdetection in chan_dahdi sets up some additional buffering that at least
> in my case, in 1.8, seems to be required (without it all inbound faxes fail
> from my hylafax server with negotiation problems). 
> Unfortunately for me, this also seems to bypass normal DID handling and
> sends calls to an undefined "fax extension".
> 
> Can anyone shed some light?
> 
> Thanks,
> 
> j
> 
> On Tue, 2012-08-28 at 15:28 -0500, Danny Nicholas wrote:
> IIRC correctly this is sort of like the "s" extension; you set up your
fax
> handler in [default,fax,1].  Not sure how that is done in FreePBX.
> 
> 

I've managed to hack a fix for this... in chan_dahdi.c I found two
places where an "async goto" happens right after the message
"Redirecting to fax extension".  I simply commented it out in both
places.

While looking a the source I noticed that an attempt is made to create a
new channel variable "FAXEXTEN" with a comment "save the DID number
before sending to the fax extension".  I created a "fax" context in
extensions.conf and tried to use ${FAXEXTEN} to properly route my
inbound fax to email calls, but it turns out that it is just set to "s",
which isn't useful at all.  I spent some time trying to figure out where
in the channel structure the actual DID information exists, as properly
setting the "FAXEXTEN" variable is arguably the right "fix" for my
problem.  But I'm just not familiar enough with the internals :( I'm
surprised others haven't had this issue...

Anyway commenting out the redirection did the trick for me.

Cheers,

j



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[asterisk-users] Receiving and processing unsolicited XMPP messages with Asterisk 11

[Noah Engelberth]

I'm trying to set up a way that our users can send an XMPP message to Asterisk 
(unsolicited) to request information, such as voicemail status or the like.  No 
matter what I set for the dialplan, I'm only seeing Asterisk execute the s,1 
priority in the context defined in xmpp.conf for incoming messages, and then 
the "call" hangs up without executing further instructions.  Anything I've 
tried to accomplish in that first priority has worked, but it never continues 
to an additional priority.

Debug output looks like:
[Aug 31 14:41:15] DEBUG[6964]: res_xmpp.c:2988 xmpp_pak_message: XMPP client 
'testaccount' received a message
[Aug 31 14:41:15] DEBUG[6964]: res_xmpp.c:3029 xmpp_pak_message: Deleted 1 
messages for client testaccount from JID jabberclient@my.jabber.server
[Aug 31 14:41:15] DEBUG[6954][C-]: pbx.c:4410 pbx_extension_helper: 
Launching 'Gosub'
[Aug 31 14:41:15] DEBUG[6964]: res_xmpp.c:3494 xmpp_client_receive: XML parsing 
successful
-- Executing [s@xmpp-incoming:1] Gosub("Message/ast_msg_queue", 
"xmpp-incoming,message,1") in new stack
[Aug 31 14:41:15] DEBUG[6954][C-]: app_stack.c:578 gosub_exec: Channel 
Message/ast_msg_queue has no datastore, so we're allocating one.
[Aug 31 14:41:15] DEBUG[6954][C-]: pbx.c:6065 __ast_pbx_run: Extension 
message, priority 0 returned normally even though call was hung up

The exact specifics of the debug after priority 1 varies a little based on what 
I try to do, but in every case, the next thing immediately after the priority 1 
application is "Extension s, priority 1 returned normally even though call was 
hungup" if I don't use a Goto/Gosub, or "Extension gotoextension, priority 0 
returned normally even though call was hungup" if I do.

I'm running Asterisk SVN-branch-11-r371592M on CentOS 6.3 64-bit.  Asterisk is 
able to send using JabberSend via other processing in my dialplan.

Thank you,

Noah Engelberth
MetaLINK Technologies

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Re: [asterisk-users] asterisk on arm

[Carlos Chavez]

On Fri, 2012-08-31 at 17:53 +, Giuseppe Longo wrote:
> Hi,
> has anyone tried asterisk on arm processors? how is the performance?
> have encountered problems in the compilation?
> 
> Thanks,
> Regards.
> 


I have installed Asterisk on a Raspberry Pi  and it works very well for
a small site.  Look at http://www.raspberry-asterisk.org/ for some more
info.



-- 
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Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] asterisk on arm

[Giuseppe Longo]

Hi,
has anyone tried asterisk on arm processors? how is the performance?
have encountered problems in the compilation?

Thanks,
Regards.

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Re: [asterisk-users] Record calls as BLOB into MySQL?

[Tony Mountifield]

In article ,
Stefan at WPF  wrote:
> 
> Thanks Danny,
> 
> do you mean better put the filename into the database?

I would normally just put the filename into the database. However, if you
had a good reason for storing the actual content in a BLOB field, you could
use the LOAD_FILE() function to copy from the file to the field. See

http://dev.mysql.com/doc/refman/5.0/en/string-functions.html#function_load-file

You would need to make sure your MySQL configuration had a suitable value
for max_allowed_packet, as mentioned in the above section.

Cheers
Tony


> 2012/8/31 Danny Nicholas 
> 
> > Probably “possible” but not recommendable.  You have no idea how long your
> > call will or will not be.  It would be better to record the call into a wav
> > or gsm file, then put that into the blob.
> >
> > ** **
> >
> > *From:* asterisk-users-boun...@lists.digium.com [mailto:
> > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Stefan at WPF
> > *Sent:* Friday, August 31, 2012 10:22 AM
> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> > *Subject:* [asterisk-users] Record calls as BLOB into MySQL?
> >
> > ** **
> >
> > Hello all,
> >
> > is it possible, to record calls directly as BLOB into a MySQL database?
> >
> >
> > Best regards
> > Stefan
> >
> > --
> > _
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Re: [asterisk-users] Record calls as BLOB into MySQL?

[A J Stiles]

On Friday 31 August 2012, Stefan at WPF wrote:
> Hello all,
> 
> is it possible, to record calls directly as BLOB into a MySQL database?
> 
> 
> Best regards
> Stefan

Yes, it's possible, with some heavy hacking  (and it ends up creating an 
unredistributable binary -- you can't comply with all the provisions of all 
the licences involved, and the only thing allowing you to use it at all is the 
fair dealing provision of copyright law);  but the real question is, *why* 
would you want to do it?

Unix already provides a perfectly good filesystem for storing files.  Just 
record your call to a file as normal, and stick the filename and whatever 
metadata you like in the database.  Job's a good 'un!

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Record calls as BLOB into MySQL?

[Stefan at WPF]

Thanks Danny,

do you mean better put the filename into the database?

2012/8/31 Danny Nicholas 

> Probably “possible” but not recommendable.  You have no idea how long your
> call will or will not be.  It would be better to record the call into a wav
> or gsm file, then put that into the blob.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Stefan at WPF
> *Sent:* Friday, August 31, 2012 10:22 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Record calls as BLOB into MySQL?
>
> ** **
>
> Hello all,
>
> is it possible, to record calls directly as BLOB into a MySQL database?
>
>
> Best regards
> Stefan
>
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Re: [asterisk-users] Record calls as BLOB into MySQL?

[Danny Nicholas]

Probably "possible" but not recommendable.  You have no idea how long your
call will or will not be.  It would be better to record the call into a wav
or gsm file, then put that into the blob.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Friday, August 31, 2012 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Record calls as BLOB into MySQL?

 

Hello all,

is it possible, to record calls directly as BLOB into a MySQL database?


Best regards
Stefan

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[asterisk-users] Record calls as BLOB into MySQL?

[Stefan at WPF]

Hello all,

is it possible, to record calls directly as BLOB into a MySQL database?


Best regards
Stefan
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Re: [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API

[Shitian Long]

For some reason, we don't want directly access database. more over, it avoid to 
develop different database connection code, and leave the database connection 
part with Asterisk 



On Aug 31, 2012, at 5:00 PM, "Danny Nicholas"  wrote:

> -Original Message-
> From: Shitian Long [mailto:longst...@gmail.com] 
> Sent: Friday, August 31, 2012 9:59 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion;
> da...@debsinc.com
> Subject: Re: [asterisk-users] Good way to query data from asterisk realtime
> with Asterisk Manager API
> 
> Do you think it is a good way to use Manager API "command action" to
> implement this feature?
> 
> 
> On Aug 31, 2012, at 4:42 PM, "Danny Nicholas"  wrote:
> 
>> There might be a specific command to do it, but you can do almost any 
>> CLI command using "command" function.
>> 
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian 
>> Long
>> Sent: Friday, August 31, 2012 9:36 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] Good way to query data from asterisk 
>> realtime with Asterisk Manager API
>> 
>> Hello.
>> 
>> I am trying to use Asterisk Manager API query data from realtime. From 
>> Asterisk CLI, we could use realtime load   
>>  query realtime it would have response like
>> 
>>  Column Name  Column Value  
>>     
>>   id  1 
>>mykey  content  
>>  myvalue  value
>> 
>> I am wondering how I could make this type of query from Manager API.
>> 
>> 
>> Thanks for your time in advance.
>> 
> I agree with Warren.  Why bother having realtime if you're going to add a
> layer you don't need?  Just query the database.
> 


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Re: [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API

[Shitian Long]

Do you think it is a good way to use Manager API "command action" to implement 
this feature?


On Aug 31, 2012, at 4:42 PM, "Danny Nicholas"  wrote:

> There might be a specific command to do it, but you can do almost any CLI
> command using "command" function.
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
> Sent: Friday, August 31, 2012 9:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Good way to query data from asterisk realtime with
> Asterisk Manager API
> 
> Hello.
> 
> I am trying to use Asterisk Manager API query data from realtime. From
> Asterisk CLI, we could use realtime loadkey value> query realtime it would have response like 
> 
>   Column Name  Column Value  
>      
>id  1 
> mykey  content  
>   myvalue  value
> 
> I am wondering how I could make this type of query from Manager API.
> 
> 
> Thanks for your time in advance.
> 
> 
> 
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Re: [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API

[Warren Selby]

On Fri, Aug 31, 2012 at 9:36 AM, Shitian Long  wrote:

> Hello.
>
> I am trying to use Asterisk Manager API query data from realtime. From
> Asterisk CLI, we could use
> realtime load   
> query realtime
> it would have response like
>
>Column Name  Column Value
>     
> id  1
>  mykey  content
>myvalue  value
>
> I am wondering how I could make this type of query from Manager API.
>
>
> Thanks for your time in advance.
>
>
Is there a specific reason you want to access the realtime data through the
Manager API and not directly from the database itself?  It seems like the
Manager API would add an extra layer to whatever you're trying to
accomplish.


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Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] Question about cli

[Shitian Long]

As far as I know there is no function add SIP extension directly from CLI, and 
actually, it is not convenience that add an extension from CLI as well. 


On Aug 31, 2012, at 4:18 PM, Giuseppe Longo  wrote:

> Hello guys,
> i would like to ask a question about cli.
> 
> Today, while i was using the cli, i thinked that there could be more features.
> IMHO, might be interesting, for example, to add a sip extensions from
> cli, or other similar functions, without having to modify the
> configuration files.
> 
> Or not? What do you think?
> 
> Regards
> 
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Re: [asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API

[Danny Nicholas]

There might be a specific command to do it, but you can do almost any CLI
command using "command" function.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
Sent: Friday, August 31, 2012 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Good way to query data from asterisk realtime with
Asterisk Manager API

Hello.

I am trying to use Asterisk Manager API query data from realtime. From
Asterisk CLI, we could use realtime loadquery realtime it would have response like 

   Column Name  Column Value  
      
id  1 
 mykey  content  
   myvalue  value

I am wondering how I could make this type of query from Manager API.


Thanks for your time in advance.



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[asterisk-users] Good way to query data from asterisk realtime with Asterisk Manager API

[Shitian Long]

Hello.

I am trying to use Asterisk Manager API query data from realtime. From Asterisk 
CLI, we could use 
realtime load
query realtime 
it would have response like 

   Column Name  Column Value  
      
id  1 
 mykey  content  
   myvalue  value

I am wondering how I could make this type of query from Manager API.


Thanks for your time in advance.



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Re: [asterisk-users] Question about cli

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo
Sent: Friday, August 31, 2012 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about cli

Hello guys,
i would like to ask a question about cli.

Today, while i was using the cli, i thinked that there could be more
features.
IMHO, might be interesting, for example, to add a sip extensions from cli,
or other similar functions, without having to modify the configuration
files.

Or not? What do you think?

Regards

Many things like that Can be done from CLI - the wisdom or danger of that
practice depends on who's at the keyboard.


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[asterisk-users] Question about cli

[Giuseppe Longo]

Hello guys,
i would like to ask a question about cli.

Today, while i was using the cli, i thinked that there could be more features.
IMHO, might be interesting, for example, to add a sip extensions from
cli, or other similar functions, without having to modify the
configuration files.

Or not? What do you think?

Regards

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Re: [asterisk-users] Automatic ODBC reconnect?

[Stefan at WPF]

Thank you Doug, I tried that, but it doesn't have the expected effect. I
set it to 1 second and restarted, no ODBC connection according to "odbc
show". After a "reload" Asterisk is connected though.

2012/8/31 Doug Lytle 

> >> So, how to tell Asterisk to automatically retry to connect via ODBC on
> failures?
>
> /etc/asterisk/res_odbc.conf
>
> idlecheck
>
> Doug
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
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Re: [asterisk-users] Automatic ODBC reconnect?

[Doug Lytle]

>> So, how to tell Asterisk to automatically retry to connect via ODBC on 
>> failures? 

/etc/asterisk/res_odbc.conf 

idlecheck 

Doug 

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Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

[Steve Davies]

On 31 August 2012 07:49, Olle E. Johansson  wrote:
>
> 24 aug 2012 kl. 16:18 skrev Steve Davies :
>
>> Hi SIP Gurus,
>>
>> I've tried to find the relevant RFCs, but am struggling. I can find
>> the odd opinion online, but was wondering if anyone could give a
>> definitive answer.
>>
>> If a SIP call is initiated (INVITE) and receives either a "180 with
>> SDP", or a "183 with SDP", then the remote party will start to send
>> audio for the inband-ringing. Asterisk then passes this audio, and it
>> is correctly heard by the caller.
>>
>> At present, Asterisk will also start to pass back any handset audio in
>> return, in theory allowing a conversation to occur on an unanswered
>> channel if an endpoint were designed to allow this (free phonecalls
>> here we come!).
>>
>> My question:
>>
>> Should:
>> 1) Asterisk block outbound audio between the 183 Progress and the 200
>> OK packets?
>> 2) Replace any outbound audio with silence?
>> 3) Replace outbound audio with a special NULL RTP of some sort? Does that 
>> exist?
>> 4) Allow any audio to be sent regardless?
>>
>> I have implemented 1) at present on our test rig, but the lack of
>> outbound RTP causes issues with firewall state not being set-up to
>> allow the inbound audio. I am not sure how hard/easy it would be to do
>> 2) as you'd need to create silence of the correct duration to replace
>> each audio frame.
>>
>> Thoughts please?
>
> First, because of Asterisk's RTP implementation we have to send some RTP 
> packets at this point. You could mute the calling channel before calling and 
> unmute the channel at answer if needed, but normally sending audio won't 
> hurt. A normal user should not be able to send early media on a pstn-like 
> installation where you bill the calls, so there should be little risc of 
> two-way conversations before an answer.
>
> In some cases you have to let the caller send DTMF (the famous fed ex 
> example) in
> early media, so we can't block any media by default in Asterisk.
>
> Using the "r" option in dial causes a lot of issues, since you can still get 
> busy or congestion when you have early media, so that is not a good solution.
>
> /Olle
>

Excellent information as always Olle. Many thanks.

My intention is to make the early-audio prevention in SIP a little
more harsh, such that if SIP receives audio before a 183 or 200 is
received, it is dropped.

This fixes the case where "useless" early-audio is received from a
non-SIP (eg ISDN) technology, and can cause an onward node to
auto-enable early audio mode, causing silent ringing and other broken
behaviours.

Cheers,
Steve

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[asterisk-users] Automatic ODBC reconnect?

[Stefan at WPF]

I recently rebooted by Asterisk server, MySQL (via ODBC) is also installed
on the same machine. After rebooting, Asterisk didn't connect via ODBC, I
assume that MySQL wasn't yet running when Asterisk tried to connect. So,
how to tell Asterisk to automatically retry to connect via ODBC on
failures? And is there a way to start MySQL before Asterisk? Thanks :-)
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Re: [asterisk-users] change channel variable to a user chosen value during a call

[Frederic Van Espen]

On Fri, 2012-08-31 at 00:11 +, Andrew White wrote:
> Is realtime an option for you to install?

Andrew,

Realtime is not an option actually. We have a whole system built up that
generates configuration files.

The primary goal is to let the user directly change the channel variable
with his phone, while in conversation, or with a short interruption of
the call.

If that isn't possible, an AMI call will be just fine. I'd just like to
make sure it is not possible on the phone itself first.

Regards,

Frederic


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