on Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
--
_
23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson:
> On 100222 1313, JT wrote:
>> When a SIP device dials another SIP device...Asterisk connects the calls and
>> displays the channel information.
>> If one of those SIP devices hangs up, Asterisk receives the hangup notice
>> and disconnects t
23 feb 2010 kl. 03.18 skrev Kevin P. Fleming:
> Kirill 'Big K' Katsnelson wrote:
>
>> The caveat here is that it is perfectly normal NOT to transmit any RTP
>> data in case of long silence. This is why the SIP timers were introduced
>> in the first place: there is no correct way to detect when t
22 feb 2010 kl. 07.23 skrev Tilghman Lesher:
>
> open audio {tcp|udp}
> close audio
If you design something now, I would strongly suggest that we stop using
"audio" as an attribute. Each call will have multiple media streams - and
already have. You need to be able to select which one, and p
21 feb 2010 kl. 16.14 skrev voipas:
> Hello,
>
>
> I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup:
> Reason: q.850;cause=17
>
No, you will have to change the code. I think there's a patch in the bug
tracker. Go search on issues.asterisk.org.
We do add a simil
19 feb 2010 kl. 11.47 skrev --[ UxBoD ]--:
> exten ==> _988XXX.,1,Dial(DAHDI/g1/${EXTEN},20)
UxBoD - you really have to read the security advisory before sending out such
examples on the mailing list. Please go to http://www.asterisk.org now.
Have a nice weekend!
Thanks,
/O
--
_
19 feb 2010 kl. 10.22 skrev Randy R:
> On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson wrote:
>> You propably have a type=friend where the user part matches before you even
>> hit the peer part, where the insecure configuration parameter matches. There
>> is a confu
17 feb 2010 kl. 19.12 skrev Joseph:
> Does the sort order matter in sip.conf file?
> I know sort order might effect:
> allow=ulaw
> allow=alaw
>
> but does it matter where I place: insecure=invite ?
>
> The reason I'm asking is that I've loaded almost two identical (sip.conf and
> extension.co
17 feb 2010 kl. 23.15 skrev Michelle Dupuis:
> Is it possible to just send an event from one Asterisk server to another?
> (Perhaps some custom event that I could define?) Or would that break the SIP
> protocol/handling in asterisk?
I think this discussion would be easier if you told us what you
17 feb 2010 kl. 19.12 skrev Joseph:
> Does the sort order matter in sip.conf file?
> I know sort order might effect:
> allow=ulaw
> allow=alaw
>
> but does it matter where I place: insecure=invite ?
>
> The reason I'm asking is that I've loaded almost two identical (sip.conf and
> extension.co
17 feb 2010 kl. 16.32 skrev Olle E. Johansson:
>
> 17 feb 2010 kl. 16.00 skrev James Northcott / Chief Systems:
>
>> Hi,
>>
>> I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
>> having a problem with Originate and chan_local.
17 feb 2010 kl. 16.00 skrev James Northcott / Chief Systems:
> Hi,
>
> I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
> having a problem with Originate and chan_local.
>
> I'm using the following Manager API action to originate a call:
>
> Action: originate
> Priority: 1
>
17 feb 2010 kl. 14.00 skrev Tzafrir Cohen:
> On Wed, Feb 17, 2010 at 12:37:33PM +0100, Håkon Nessjøen wrote:
>> Only a warning, and doesn't seem to do anything bad.
>>
>> But I can't seem to figure out what these warnings mean?
>>
>>-- Requested transfer capability: 0x00 - SPEECH
>> [Feb 17
17 feb 2010 kl. 12.37 skrev Håkon Nessjøen:
> Only a warning, and doesn't seem to do anything bad.
>
> But I can't seem to figure out what these warnings mean?
>
> -- Requested transfer capability: 0x00 - SPEECH
> [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized
17 feb 2010 kl. 11.13 skrev Mian Asif:
> Hi,
> when call is Hangup, Asterisk send "X-Asterisk-HangupCauseCode" in Bye packet.
> i want to remove Asterisk keyword from this string
> "X-Asterisk-HangupCauseCode".
> please tell how i can remove Asterisk from above string at call hangup time.
You
While we continue discussing all possible solutions to this and build an
expanding knowledgebase, I would like to repeat myself and kindly ask everyone
that blogs, twitters, talks and teaches about Asterisk to please spread the
word and the links. Later today, there will be an official Asterisk
16 feb 2010 kl. 10.40 skrev Alexandru Oniciuc:
> Hello list,
>
> debugging SIP, I found many empty lines like:
>
> <--- SIP read from UDP://XXX.XXX.XXX.XXX:5060 --->
> <->
>
> The IP address above corresponds to one of my accounts, which
> is beh
16 feb 2010 kl. 09.43 skrev Tzafrir Cohen:
> On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
>> On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri wrote:
>>
>>> Yes but in any case you can enter all of the strings that reasonably match
>>> - even if you have variable-length numbers, y
16 feb 2010 kl. 08.54 skrev Olivier:
> Hi,
>
> Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support
> TR-069 (see http://en.wikipedia.org/wiki/TR-069).
>
> Has someone experienced with TR-069 ?
> What do you think of this protocol set ?
>
And the SIP forum is about to rele
15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere:
>
> Playing around with the Grandstream GXV3140.
>
> I'm interested in having the video voicemail clips emailed in a format
> that might be opened by Windows Media Player or even Quicktime. Have been
> googling around a lot and have tried various
15 feb 2010 kl. 17.36 skrev Steve Edwards:
> On Mon, 15 Feb 2010, Amit Patkar | Avhan Technologies Pvt. Ltd. wrote:
>
>> I want at least 480 concurrent PSTN-IP calls.
>
> 0) Cross-posting is a no-no.
>
> 1) Not a -dev question. If you ever have any doubt a question belonging on
> -dev, it doe
15 feb 2010 kl. 10.00 skrev Randy R:
> On Mon, Feb 15, 2010 at 9:51 AM, Olle E. Johansson wrote:
>>> To avoid extensive rewriting and fix the current issue.
>> That works in countries where you have fixed-length numbers. Unfortunately,
>> not every dialplan works that
15 feb 2010 kl. 09.33 skrev Lenz Emilitri:
> Or one could simply rewrite to:
>
> [incoming-from-voip]
> exten => XXX,1,Dial(${ext...@incoming-from-voip-old)
> exten => ,1,Dial(${ext...@incoming-from-voip-old)
> exten => X,1,Dial(${ext...@incoming-from-voip-old)
> exten => XXX
14 feb 2010 kl. 21.04 skrev Steve Edwards:
> On Sun, 14 Feb 2010, Kyle Kienapfel wrote:
>
>> strip_ampersands(${EXTEN})?
>
> (sip.conf)
>
> [general]
> allow-characters= all
> disallow-characters = "&"
>
> [example-did-provider]
> allow-characters
ose too.
> However your article is very
> informative about how to filter them.
> The fix for this - at least at the moment - is education. I doubt it
> will take too long to see script kiddies exploiting this.
I can not agree more!
Thank you for the feedback.
Regards,
/Olle
>
Friends,
Last week, Hans Petter Selansky alerted us of a potential security issue in all
releases of Asterisk. In fact, it doesn't involve the code, but the most common
way to construct dialplans. If you have something like this in your Asterisk,
you need to update your dialplans:
[incoming-fr
13 feb 2010 kl. 16.57 skrev JR Richardson:
> Hi All,
>
> I read some discussions about the new SIP authentication methods for
> 1.6.X branches and possible addition of new type of user, type=trunk.
> I'm wondering about the disposition about this. Will it be added?
Not to the 1.6 branches, but
12 feb 2010 kl. 16.43 skrev Klaus Darilion:
>
>
> Am 11.02.2010 21:09, schrieb Olle E. Johansson:
>>
>> 11 feb 2010 kl. 13.30 skrev Klaus Darilion:
>>
>>> Am 11.02.2010 11:21, schrieb Armin Schindler:
>>>> Hello,
>>>>
>>&
11 feb 2010 kl. 13.30 skrev Klaus Darilion:
> Am 11.02.2010 11:21, schrieb Armin Schindler:
>> Hello,
>>
>> using Asterisk 1.4.28, I encountered a problem with SIP
>> RTP port allocation.
>>
>> I found some entries in mailinglist and bugtracker regarding
>> this issue, but only old ones.
>>
>>
11 feb 2010 kl. 08.49 skrev Ron Arts:
> Op 11-02-10 03:42, sean darcy schreef:
>> Kevin P. Fleming wrote:
>>> sean darcy wrote:
I found out that the [globals] section in extensions.conf is ignored if
an #include 'd file has a [globals] section. Is this intended?
In this parti
8 feb 2010 kl. 12.29 skrev Klaus Darilion:
> Hi!
>
> IIRC there was an announcement some time ago that it is possible now to
> make conferences without the need for DAHDI anymore - but I can not
> remember the name of this feature anymore, and google didn't solved my
> problem.
>
> Thus, any
8 feb 2010 kl. 11.26 skrev Tzafrir Cohen:
> On Mon, Feb 08, 2010 at 11:03:19AM +0100, Olle E. Johansson wrote:
>
>> You will have to recompile it with the DONT_OPTIMIZE variable set so
>> that the core dump actually has meaningful symbols.
>
> Doing so hurts your perfor
8 feb 2010 kl. 08.37 skrev Steve Totaro:
> On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson wrote:
>>
>> 7 feb 2010 kl. 15.09 skrev Per Jessen:
>>
>>> Thomas Winter wrote:
>>>
>>>> Hi,
>>>>
>>>> my Asterisk on
7 feb 2010 kl. 15.09 skrev Per Jessen:
> Thomas Winter wrote:
>
>> Hi,
>>
>> my Asterisk on debian lenny died after 80 days.
>>
>> server kernel: [7572666.186852] asterisk[3673]:
>> segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l
>> ibpthread-2.7.so[7f3b8e903000+16000]
>>
>> Anything
5 feb 2010 kl. 10.36 skrev Philipp von Klitzing:
> Hi!
>
OpenVPN by default uses UDP, but can be configured to use TCP.
>>
>> So what's the configuration on the Snom? Can I change it?
>
> Google is your friend:
> http://wiki.snom.com/Networking/VPN
>
So what you're saying is that you hav
5 feb 2010 kl. 10.37 skrev Randy R:
>>> Why not run a internal DNS with forwarders to your ISP ? That way Asterisk
>>> can still resolve itself and hosts internally.
>>>
>> See above:
you need a local
resolver, like a caching BIND server, on the same host.
>
> Nice, but still, it rui
5 feb 2010 kl. 09.28 skrev --[ UxBoD ]--:
> - "Randy R" wrote:
>
>> On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson
>> wrote:
>>>>> What I have seen on my asterisk box when I had a up/down adsl line
>> was
>>>>> that the as
5 feb 2010 kl. 06.49 skrev Anthony Messina:
> On Thursday 04 February 2010 23:22:27 Alex Samad wrote:
>> What I have seen on my asterisk box when I had a up/down adsl line was
>> that the asterisk box couldn't do dns resolution and would hang( well no
>> other internal calls could be made, seemed
4 feb 2010 kl. 21.54 skrev Alex Balashov:
> On 02/04/2010 03:48 PM, Doug Lytle wrote:
>
>> OpenVPN by default uses UDP, but can be configured to use TCP.
>>
>> So, under UDP, there should be no issues with retransmits.
>
> It does have a primitive built-in backward acknowledgment mechanism eve
4 feb 2010 kl. 19.42 skrev Steve Totaro:
> On Thu, Feb 4, 2010 at 11:30 AM, --[ UxBoD ]-- wrote:
>> - "Ken D'Ambrosio" wrote:
>>
>>> It's just come to my attention that newer phones from both Snom and
>>> Grandstream support OpenVPN. Is this a new trend or something?
>>> Since
>>> OpenVPN
3 feb 2010 kl. 08.11 skrev Alex Balashov:
> On 02/03/2010 02:03 AM, Olle E. Johansson wrote:
>>
>> 2 feb 2010 kl. 11.20 skrev BERGANZ Francois:
>>
>>> Hello all,
>>>
>>> Does asterisk accept uri tel: instead of sip: ?
>>>
>>>
2 feb 2010 kl. 11.20 skrev BERGANZ Francois:
> Hello all,
>
> Does asterisk accept uri tel: instead of sip: ?
>
>
No, but I think it would be a good addition.
/O
--
_
-- Bandwidth and Colocation Provided by http://www.api
30 jan 2010 kl. 23.40 skrev Michiel van Baak:
> On 14:29, Sat 30 Jan 10, Olle E. Johansson wrote:
>> Friends,
>>
>> Before the Christmas holidays, I did send this letter and did not get a lot
>> of response, but some. Since then, I've been able to get interest
Asterisk to work as we have done for the last 10 years...
With IPv6 greetings!
/Olle
Vidarebefordrat brev:
> Från: "Olle E. Johansson"
> Datum: 17 december 2009 09.39.40 CET
> Till: Asterisk Non-Commercial Discussion Users Mailing List -
>
> Ämne: [asterisk-users] As
29 jan 2010 kl. 17.20 skrev Kristian Kielhofner:
> On Fri, Jan 29, 2010 at 10:31 AM, Kevin P. Fleming
> wrote:
>>
>> Well, that's the problem, and it's the reason why 603 is so commonly
>> used. This is a situation where the current request has failed, but
>> there is no indication that repeat
rom mobile device
>
> On Jan 29, 2010, at 3:54 AM, "Olle E. Johansson" wrote:
>
>> Agree that the 603 is wrong. It hasn't caused me issues but I see
>> where it could. And it goes against what I have been teaching in my
>> classes, which is irrita
>
>
> Da: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich
> Inviato: giovedì 28 gennaio 2010 21:41
> A: asterisk-users@lists.digium.com
> Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short
>
I would ve
Agree that the 603 is wrong. It hasn't caused me issues but I see where it
could. And it goes against what I have been teaching in my classes, which is
irritating ;-)
In Asterisk, it's only used when we have no other hangup cause - and is
propably an indication that there is a code path that do
27 jan 2010 kl. 11.47 skrev Administrator TOOTAI:
> Hi,
>
> we had an attack on a server and we don't understand how it was
> possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
> network 188.161.128.0/18
>
> Hacked account had following setup:
>
> [111]
> type=friend
> use
26 jan 2010 kl. 16.48 skrev Örn Arnarson:
> Hi guys,
>
> I am wondering (and have been unable to find out thus far) whether Asterisk
> sets some special channel variables or something when a call is transfered
> with the REFER method.
> Basically, I'm trying to figure out if it is possible to
> From: ;tag=60b512cec9;epid=08fd7dc31f
> To: ;tag=as63c5f412
> Call-ID: 16a3a30998874ae98538d221a2567fe1
> CSeq: 2 ACK
> User-Agent: UCCAPI/2.0.6362.67
> Authorization: Digest username="56", realm="asterisk", algorithm=MD5,
> uri="sip:5...@trixbox1.local", n
0/TCP 192.168.1.15:52774
> Max-Forwards: 70
> From: ;tag=39be813029;epid=f918608aea
> To: ;tag=as5c7a7ed8
> Call-ID: 738a7dd4d06d4c439c29fb703e491533
> CSeq: 2 ACK
> User-Agent: UCCAPI/2.0.6362.67
> Authorization: Digest username="56", realm="asterisk", algorithm=MD5,
> uri=
Hello!
I tried using sendtext() in the Asterisk dialplan to send a SIP MESSAGE to Bria
or a recent Eyebeam on my mac. I know it used to work, but right now I get "100
trying" and nothing else from the softphone.
Anyone that knows what's going on here?
Thanks,
/O
--
___
15 jan 2010 kl. 08.23 skrev Yuji Kondo:
>
>
> I have two questions for Asterisk feature.
>
>
> 1. Can Asterisk support presence feature ?
Asterisk is a telephony PBX and supports presence subscriptions for extension
states - if a phone line is busy or not, over a few different SIP presence
13 jan 2010 kl. 09.26 skrev hadi motamedi:
>
>
> On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson wrote:
>
> 13 jan 2010 kl. 06.56 skrev hadi motamedi:
>
> > Dear All
> > I have Asterisk 1.4 installed on my Debian server . I am considering
> > upgradi
My apologies for the multiple copies.
Had issues with a mailserver that somehow wasn't talking to DNS properly. Now
fixed. It behaved like Asterisk does sometimes, very poor when it can't connect
to DNS. Had power outage yesterday and I think that started it all...
Meanwhile, I tried to retran
13 jan 2010 kl. 06.56 skrev hadi motamedi:
> Dear All
> I have Asterisk 1.4 installed on my Debian server . I am considering
> upgrading my Asterisk to the latest version (1.6) . Can you please let me
> know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
> 1.6 ?
Plea
13 jan 2010 kl. 06.56 skrev hadi motamedi:
> Dear All
> I have Asterisk 1.4 installed on my Debian server . I am considering
> upgrading my Asterisk to the latest version (1.6) . Can you please let me
> know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
> 1.6 ?
Plea
12 jan 2010 kl. 19.47 skrev Danny Nicholas:
> Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a
> 1/2 second delay before dialing, ww1234 a 1 second delay, etc.
>
> Try it with 2 or 3 w's instead of 1...
I have no solution, but can only say this: a 'w' in a SIP dialstri
13 jan 2010 kl. 06.56 skrev hadi motamedi:
> Dear All
> I have Asterisk 1.4 installed on my Debian server . I am considering
> upgrading my Asterisk to the latest version (1.6) . Can you please let me
> know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
> 1.6 ?
Plea
12 jan 2010 kl. 20.56 skrev David Gibbons:
>
> 'w' is really only supported on channels where digit-by-digit dialing is
> the norm, which generally means analog trunks (or digital trunks using
> CAS signaling).
>
>
>
> Thanks Kevin, that's what I figured (though not quite so concisely)...
>
11 jan 2010 kl. 16.23 skrev --[ UxBoD ]--:
> Hi,
>
> why would Asterisk core dump with the following test dialplan extension ?
>
> exten => 8100,1,Answer()
> exten => 8100,n,Set(CALLERID(all)="")
> exten => 8100,n,PrivacyManager()
> exten => 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:noci
11 jan 2010 kl. 12.25 skrev ahmed magdy:
> Hello,
>
> I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know
> how can i monitor the extension status?
> when i wrote sip show peers on asterisk
> Extension Domain port Status
> 111
8 jan 2010 kl. 08.01 skrev Tilghman Lesher:
> On Thursday 07 January 2010 21:17:52 JR Richardson wrote:
>> On Thu, 7 Jan 2010, Tilghman Lesher wrote:
>>> On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson wrote:
problem I'm running into is if the DNS server is not responding, the
script hang
"Net::DNS::Async is a fire-and-forget asynchronous DNS helper. That is, the
user application adds DNS questions to the helper, and the callback will be
called at some point in the future without further intervention from the user
application. The application need not handle selects, timeouts, wa
s
> Contact:
> Content-Length: 0
>
>
> <>
>-- Stopped music on hold on SIP/1050-0a6ffa70
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-
7 jan 2010 kl. 12.00 skrev Olivier:
> Hello,
>
> I've read in Mantis that asterisk.conf's "internal timing" option could
> positively impact Asterisk behaviour during faxing
> (http://issues.asterisk.org/view.php?id=16374).
> Before using it, I would be very pleased to read a line or two about
7 jan 2010 kl. 10.21 skrev Aggio Alberto:
> Hi,
> I have occasionally experienced the same problem too, and I suspect it was
> caused by some spikes in network traffic (e.g. for an intensive file
> transfer) that delayed too much SIP OPTION response, so that Asterisk marked
> these devices as
5 jan 2010 kl. 10.08 skrev hadi motamedi:
>
>
> On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson wrote:
>
> 4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
>
> > hadi motamedi wrote:
> >
> >> Sorry . I didn't get the point clearly . In the SIP Invi
4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
> hadi motamedi wrote:
>
>> Sorry . I didn't get the point clearly . In the SIP Invite message , it
>> says "my audio endpoint is IP x.x.x.x port x, and I can use codecs
>> A,B,C". The remote endpoint responds with a 200 OK, saying "my audio
>> stream
4 jan 2010 kl. 09.34 skrev Remco Barendse:
> Is there any fix or workaround for the DNS problem (old standing bug that
> when the box starts and domain names do not resolve quickly enough from
> DNS then asterisk stops using the outgoing trunks.
>
> I read on the list before that it is conside
3 jan 2010 kl. 17.47 skrev Steve Edwards:
>> 1 jan 2010 kl. 20.04 skrev Shariq Khan:
>>
>>> I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time
>
> On Sun, 3 Jan 2010, Olle E. Johansson wrote:
>
>> No, Asterisk only supports one port.
>
ion Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
___
t;> Dec 23 13:13:43 VERBOSE[18837] logger.c:
>>-- Executing
>> Dial("SIP/6174-b456d828", "SIP/4062|20|tTwWM(auto-blkvm)")
>> in new stack
>> Dec 23 13:13:43 NOTICE[18837] app_dial.c: Unable to create
>> channel of type
>> 'SIP&#
24 dec 2009 kl. 10.30 skrev Shelvananda, Ramananda Arkalgud:
> Hi All,
>
> Is someone implemented Tel uri support in the latest asterisk ? If yes, can
> you guys share some info on it
>
No.
But I am very interested in why you ask? Do you have devices that support Tel:
uri's? DO you have
24 dec 2009 kl. 08.18 skrev listu...@spamomania.co.uk:
> Hi,
>
> How would I go about troubleshooting this:
>
> [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum
> retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for
> seqno 101 (Critical Response) -- See doc/
23 dec 2009 kl. 16.00 skrev didier.cuffaut:
> I apologize for my poor English.
> So, i don't really understand 'how to' realize thus
>
> When you use the cmd Dial and want to get $ from caller channel to callee (or
> callee channel from caller), which way is the right way ?
>
If you
23 dec 2009 kl. 19.52 skrev Ira:
> Someone posted a message suggesting someone try sendtext() and so I
> thought I'd see if it was useful. Much searching through help at the
> CLI has failed to find any help for sendtext, but I did find that:
>
> "core show function vmcount" fails but:
>
> "
23 dec 2009 kl. 11.25 skrev David Cunningham:
> Shukun,
>
> It tells you "No such file or directory". Is the file in your modules
> directory?
Actually, to be more specific. The module cdr_radius.so exists, but can't bind
to the radius library "libradiusclient-ng.so.2".
Check LD_LIBRARY_PATH
23 dec 2009 kl. 10.16 skrev Zhang Shukun:
> hi, all
> when i run "make menuselect", it say
>
> Terminal must be at least 80 x 21.
> menuselect changes NOT saved!
>
> in the bottom message, what's wrong?
Terminal must be at least 80x21
You need a terminal window that handles at least 80 ch
23 dec 2009 kl. 08.53 skrev jonas kellens:
> Can I define the realm on a per peer basis ??
> Can I define a realm to be used for one peer and another realm for another
> peer in sip.conf ??
>
> I have an ITSP that I need to authenticate with a realm that they set. But
> this realm is not valua
23 dec 2009 kl. 06.17 skrev prasha...@digilink.in:
>
> Hi,
>
> How asterisk distinguish whether the re-invite is for codec change or for a
> session refresh? I know that it checks the session version and decides the
> same. But even if session version is different from the initial invite and
Dear Asterisk community,
Yesterday, Russell Bryant finally made up his mind and confirmed on the
asterisk-dev mailing list that the next release of Asterisk will be 1.8, which
will also be a Long Term Support (LTS) release. This also means that the 1.4 is
now officially classed as a LTS release
21 dec 2009 kl. 15.11 skrev Danny Nicholas:
> You can do virtually any command with the manager command object; the trick
> is getting the syntax down. There's a decent example on voip-info.org that I
> used to set up mine.
>
Well, that's one way, but you have to be careful, because the CLI outp
20 dec 2009 kl. 09.10 skrev Magnus Benngård:
> Hi!
>
> Probably me that cannot read the manual...
>
> I am trying to get all Keys that belongs to a certain Family
> from the manager interface. Can just get single values for example:
>
> Action: DBGet
> Family: CFIM
> Key: 0317998975
>
While d
21 dec 2009 kl. 12.00 skrev jonas kellens:
> My SIP-provider sends my a SIP-invite like this :
>
> INVITE sip:329298y...@80.xx.xx.69:5060 SIP/2.0
> Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c
> Max-Forwards: 70
> From: ;tag=f395877e02bf8eb2fd8f5a0e
> To:
> Call-ID:
21 dec 2009 kl. 09.34 skrev Olivier:
>
>
> 2009/12/21 Olle E. Johansson
>
> 21 dec 2009 kl. 00.04 skrev Olivier:
>
> > Hi,
> >
> > I'm banging my head over this.
> >
> > Usually, I'm using a SIP hardphone feature called "Call
21 dec 2009 kl. 00.04 skrev Olivier:
> Hi,
>
> I'm banging my head over this.
>
> Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to
> enhance BLF with Directed Call Pickup :
> basically, SIP hardphone (here a Thomson ST2030S) is configured to send an
> INVITE message
Friends,
At the first Astricon I was very happy to see Marc Blanchet as one of the
attendees. I knew he was one of the IPv6 gurus and wanted someone to show some
interest in Asterisk and IPv6.
Well, he did not only get interested in it, but started coding on it. The
results have been availabl
11 dec 2009 kl. 23.21 skrev John Taylor:
> I have multiple trunks to the same ITSP. Incoming calls to any trunk
> go to the last "incoming" label defined in those trunks' contexts in
> sip.conf.
>
> My ITSP insists on insecure=very in the trunk context; is this the cause?
>
This is an effect of
_send_all_registers() that is a good starting point for exploration.
Cheers,
/O
---
* Olle E. Johansson - o...@edvina.net
* Asterisk Training http://edvina.net/training/
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asteri
Hello,
Several folks working with Kamailio, SIP Router, SER, OpenIMSCore,
SEMS and Asterisk are in Berlin next week, so we think of having a
dinner (or beer) meeting Thursday, 19:00, Nov 12, 2009. If happens
that you are around and want to join, please send me an email to make
sure you get
14 sep 2009 kl. 12.05 skrev Stanisław Pitucha:
> 2009/9/9 Stanisław Pitucha :
>> I've got different customers that may use the same asterisk. Each
>> user
>> can blind-transfer a call to whatever place they want. But of course
>> the transferring side should be billed for it.
>> What can I do t
You can also use our jabber/xmpp integration and send an Instant
message to the user/desktop before you place the call with dial(). Or
do it in the dial() macro as soon as someone answers.
/O
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10 sep 2009 kl. 17.35 skrev Alex Balashov:
> Andrew Stewart wrote:
>
>> Figured out the problem. There is an "inspect sip" command in our
>> global policy map on our Cisco ASA firewall. That was "fixing" the
>> CALL-ID. Took it out and all is working now.
>
> Ah, yes. Those ALGs (or other >=
10 sep 2009 kl. 12.33 skrev DHAVAL INDRODIYA:
> hello
>
> I would like to take value RTPAUDIOQOS channel variable on DAHDI /
> IAX Channel...
DAHDI doesn't use the Realtime Transport Protocol, RTP.
/O
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8 sep 2009 kl. 15.40 skrev Benny Amorsen:
> I see a lot of these on an otherwise idle Asterisk 1.6.0.15:
>
> Extension Changed 773[Hints] new state Ringing for Notify User
> 792-00041327d17e-1. Then a little while later it changes to InUse or
> Idle, completely randomly. It happens for many diffe
8 sep 2009 kl. 10.17 skrev jonas kellens:
> Erik,
>
> I have placed everything in features.conf in comment ( ; ). Still
> when I run show features, I get this :
>
>> clarkconnect*CLI> show features
>> Builtin Feature Default Current
>> --- --- ---
>> Pick
5 sep 2009 kl. 20.02 skrev Jens Wolf:
> Hi,
> i have try to send IM from Client A (Ekiga) to Client B (Ekiga).
Realtime text is sent in the RTP stream. What you're sending is a SIP
message. Those are two different things.
Asterisk does not support SIP messages outside of a call - so test
whi
183 Session
> > Progress
> > 4 0.046546 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK,
>
>
> On Fri, Sep 4, 2009 at 11:55 PM, Olle E. Johansson
> wrote:
>
> 5 sep 2009 kl. 04.58 skrev Jai Rangi:
>
> > Hello,
> >
> > I have a issue betwe
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