* You are viewing the archive for May 10th, 2011

When someone helps you, at least let them know if the problem is resolved or not

I’ll keep this brief because I don’t want to come across like any more of an
a$$ than I absolutely have to, especially since I know I’ve blown my stack
before…..

Gentlemen (and Ladies, if you’re out there),

If someone gives you advice on this list, and ESPECIALLY if they give you
advice offlist, have the courtesy to (AT THE LEAST) to let them know when/if
you get your question answered or your problem solved.

As many people point out, on community supported mailing lists and forums
around the world, these user lists are comprised of people who are giving
their time freely to help others learn about the software the list is about.
Sometimes those lists are about software that is quite useful in a
commercial setting, perhaps even very much in demand, like Asterisk. Now,
you should always appreciate when you get assistance from people on user
lists, but when you’re asking for help on a list like this one, (where I’d
say 80% of the participants on the list are professionals who earn their
living by selling their knowledge of how to install, configure, and maintain
a server application like Asterisk) it would be extremely appreciated if you
show some courtesy to the individual(s) who assisted you for free. I’ve had
several individuals contact me offlist (without being given permission
first, which is first and foremost bad form) and ask for my assistance with
configuring a feature, troubleshooting an issue, and once I got an email
that said something along the lines of:
“I saw a post on the list where you said you could accomplish *
insertNiftyFeatureThatDidNotPreviouslyExistHere*…. Tell me how to do it”
I’m sure many of you have been the recipient of more than your fair share of
emails offlist asking for help, and I’m sure a great number of you try to
offer assistance. What is bothering me is the fact there seems to be a new
trend forming, wherein I don’t get a repsonse from the person I tried to
help, even when I can feel confident in saying that I know I gave them the
piece of information they needed in order to answer their question and
accomplish the goal of making Asterisk perform the way they wanted…..

Has anyone else noticed this trend?

Those of you who are making the requests, is there a reason why you don’t
feel the need to be courteous and at least say, “Hey that advice worked,
everything’s working now”?

Next time you ask for help, especially when it’s offlist (and even MORE SO
when you’re contacting someone you weren’t invited to contact offlist), I
want you to remember that the person you’re contacting usually gets paid for
their time as an Asterisk professional, and that they’re helping you for
free. Hell, if you want to get down to brass tacks about it, thatr person
who is taking the time to try and help you is increasing his or her own
professional competition……

that’s all…nothing super rude, but I had to get that one out there…. I
get annoyed when I answer about 12-13 questions (all in separate emails,
mind you) from someone, and then I never get even find out if I was
successful in helping them

iax2 Max retries exceeded to host

We have IAX2 peer between two asterisk and I am getting following error following IAX2 WARNING. IAX calling is functional

[May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3030332, seqno=211)
[May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3040332, seqno=212)
[May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register: Registration from ‘‘ failed for ‘172.30.245.85:5060′ – No matching peer found
[May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register: Registration from ‘
‘ failed for ‘172.30.245.85:5060′ – No matching peer found
[May 10 15:23:49] WARNING[2048]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 2, ts=3045385, seqno=213)
[May 10 15:23:54] WARNING[2054]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3050332, seqno=214)
[May 10 15:24:04] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3060332, seqno=215)
[May 10 15:24:10] WARNING[2048]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 2, ts=3066385, seqno=216)
[May 10 15:24:14] WARNING[2051]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3070332, seqno=217)

About X100P and TDM400P analog card in China

So does this mean no solution when used ZAP/DAHDI with PSTN line?

If I installed an E1, will that work?

Thanks.
Regards.

On Wed, May 11, 2011 at 12:57 AM, John Novack > wrote:

> Remember that ZAP/DAHDI channels don’t receive ( because most PSTN/POTS
> lines don’t provide ) answer supervision.
> This will certainly complicate what you want do do.
>
> John Novack
>
>
> Scott Zhang wrote:
>
> Hello. All.
> I am a bit new to asterisk, started from half a month ago.
> I am setting up a home asterisk server with analog card. I am using
> asterisk 1.4.27.
> At the moment, I bought a X100P card and installed it on my computer. I
> used it to connect my home phone line. For the moment, it works fine when
> dial in. Soon I noticed when I dial out through it to my mobile, it can’t
> hang up automatically after I hang up my mobile. After googled, I found the
> reason as described as below link and some solutions.
>
> http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html
> For me, none of solutions works.
> So I am rethinking should I buy another TDM400P card.
> But I am wondering because in China. The phone system looks different
> so I don’t know if TDM400P will work or not.
>
> Here is the flow when I am using X100P to dial out.
> 1. Pick up phone
> I hear tone. DA~~~
> 2. press the number
> tone: DA~~~
> 3. dialing~~~~
> No more tone. Music playing~~~~~(lalala, I love lalal)
> At the same time, on asterisk console, it prints out. “The call has been
> answered”.
> Actually it is still dialing and my mobile is ringing because I didn’t
> answer the call.. The music was played by ISP
> 4. whether I answered the call or refuse the call. No more prints on
> asterisk console.
> But on phone end, when I refuse the call, instead of busytone, I hear the
> voice “The phone you’re dialing is busy now. Please try again later.”.
> So the whole thing is, during the whole call process, only before dialing,
> we can hear the phone tone, for all other time, Dialing, refused, the ISP
> will play music/voice instead of providing the tone. I don’t understand how
> x100p identify the status, I guess should be on the tone.
> 5. I wait asterisk/x100p to hangup the call and after 5 minutes, I have to
> cut the phone line to force it hang up.
>
> So can TDM400X work with such a system without tone only with music and
> voice?
>
> Thanks.
> Regards.
> Scott
>
>
> –
> _____________________________________________________________________
> — 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
>
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>
> –
>
> Dog is my Co-pilot
>
>

Using MixMonitor()

Hello Folks;

I appreciate all of the help so far – thanks.

Another question: I am using MixMonitor() to record calls and I would
like to include the called number/extension in the filename:

In my dialplan, I am able to save the file with the caller id in the
filename. However, what I am a little unsure about is the incoming
number/called number/extension – passing that information on to part of
the filename.

Does anyone follow me?

Glen

Asterisk 1.8.4 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4 resolves several issues reported by the community.
Without your help this release would not have been possible. Thank you!

Below is a sample of the issues resolved in this release:

* Use SSLv23_client_method instead of old SSLv2 only.
(Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell
and chazzam.

* Resolve crash in ast_mutex_init()
(Patched by twilson)

* Resolution of several DTMF based attended transfer issues.
(Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
shihchuan, grecco. Patched by rmudgett)

NOTE: Be sure to read the ChangeLog for more information about these changes.

* Resolve deadlocks related to device states in chan_sip
(Closes issue #18310. Reported, patched by one47. Patched by jpeeler)

* Resolve an issue with the Asterisk manager interface leaking memory when
disabled.
(Reported internally by kmorgan. Patched by russellb)

* Support greetingsfolder as documented in voicemail.conf.sample.
(Closes issue #17870. Reported by edhorton. Patched by seanbright)

* Fix channel redirect out of MeetMe() and other issues with channel softhangup
(Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb.
Patched by russellb)

* Fix voicemail sequencing for file based storage.
(Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by
jpeeler)

* Set hangup cause in local_hangup so the proper return code of 486 instead of
503 when using Local channels when the far sides returns a busy. Also affects
CCSS in Asterisk 1.8+.
(Patched by twilson)

* Fix issues with verbose messages not being output to the console.
(Closes issue #18580. Reported by pabelanger. Patched by qwell)

* Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by
alecdavid, Irontec, ZX81, cmaj)

Includes changes per AST-2011-005 and AST-2011-006
For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4

Information about the security releases are available at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

Thank you for your continued support of Asterisk!

Plotting fxotune dump

Can anyone shed any light on how you plot the data outputted by fxotune
-d into a graph similar to the one in the link below?

http://www.voip-info.org/storage/users/65/31065/images/673/medium.jpg