Re: [asterisk-users] Unable to build sip pvt data
Hello, is there any more feedback on this thread ? On 10/20/2011 05:10 PM, Jonas Kellens wrote: On 10/20/2011 05:07 PM, Paul Belanger wrote: On 11-10-20 10:28 AM, Jonas Kellens wrote: Hello list, what does this mean ? [Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data for 'account9' (Out of memory or socket error) [Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data for 'account10' (Out of memory or socket error) [Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data for 'account11' (Out of memory or socket error) [Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data for 'account12' (Out of memory or socket error) [Oct 20 15:45:29] ERROR[4355] chan_sip.c: Unable to build sip pvt data for 'account11' (Out of memory or socket error) [Oct 20 15:45:29] ERROR[4355] chan_sip.c: Unable to build sip pvt data for 'account12' (Out of memory or socket error) [Oct 20 16:18:17] ERROR[8996] chan_sip.c: Unable to build sip pvt data for 'account7' (Out of memory or socket error) [Oct 20 16:18:17] ERROR[8996] chan_sip.c: Unable to build sip pvt data for 'account8' (Out of memory or socket error) [Oct 20 16:18:17] ERROR[8996] chan_sip.c: Unable to build sip pvt data for 'account9' (Out of memory or socket error) [Oct 20 16:18:17] ERROR[8996] chan_sip.c: Unable to build sip pvt data for 'account10' (Out of memory or socket error) [Oct 20 16:18:17] ERROR[8996] chan_sip.c: Unable to build sip pvt data for 'account11' (Out of memory or socket error) [Oct 20 16:18:17] ERROR[8996] chan_sip.c: Unable to build sip pvt data for 'account12' (Out of memory or socket error) Which version of asterisk is this? Sorry. This is Asterisk 1.6.2.20. Thanks. 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 -- _ -- 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 know RTP por of a SIP client in
OP may be able to use System through Dial plan but I'm thinking that since
tcpdump don't just give output within seconds or neither do it get
daemonized? so this system() call will hold the call to that priority. This
may even result in call failure. I think this system call should trigger a
shell script that launches an instance of tcpdump and move forward in the
dial plan.
Can anyone tell if we can extract a header value from SDP(for RTP Tx/RX
ports) within the SIP packet using the SIP_HEADER function?
How about using sipgrep: The idea is launch a sipgrep based scripts in the
background which just takes Call-ID and parse RTP port data and save it in
memcached. This memchache Key/value register will just save [Call-ID:RTP
port data] for each call entering into the Server. This script should start
separate instances of tcpdump for each call with separate file names.
On each call hangup call the h extensions will use the SIP_HEADER(call-id)
Key and trigger a stop command for the background tcpdump for this
particular call.
On Mon, Oct 24, 2011 at 4:36 AM, Bruce B bruceb...@gmail.com wrote:
Then you may use system() in dial-plan to run that shell command along with
what I suggested.
-Bruce
On Sun, Oct 23, 2011 at 5:22 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Yes, I need to know to get in in dialplan because I want to capture
traffic per call. I would like to launch $SHELL{tcpdump src port } in
the dialplan or something like this. And I want RTP traffic only of a
certain call.
Thank you!
===
Date: Fri, 21 Oct 2011 09:41:39 -0400
From: Bruce B bruceb...@gmail.com
Subject: Re: [asterisk-users] how to know RTP por of a SIP client in
the dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
CAJyE_uWLRXkhrWQ-6SvNW1ihN-nGA3HFwHt=
pu-tfr6lybi...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Do you need to know to get it in dialplan? If I not, from shell (not
Asterisk CLI) I usually use:
netstata -a | grep asterisk
By default Asterisk settings it should be something between 10k-20k
-Bruce
On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
How can I get the RTP port one SIP client is using for sending/receiving
RTP flow? Can I obtain it in from SIP_HEADER of something like that in
the
dialplan?
Thank you!
** **
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[asterisk-users] ${CALLERID(num)} after doing transfer from extension to extension
Hi All;
As I am using the ${CALLERID(num)} to be part of the filename that I am
recording it, I am facing the following problem:
If the incoming call (via PSTN) reached for an extension (which is the
reception), and then the extension transferred the call to the proper person,
and we need to do recording for the call at this proper person, the problem
that at this point the ${CALLERID(num)} will represnt the reception guy
extension and not the original caller id of the caller who called from outside
via the PSTN. How can I get this original caller id?
Regards
Bilal
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Re: [asterisk-users] ${CALLERID(num)} after doing transfer from extension to extension
Set CDR(destination) or whichever field you need to get recorded in CDRs to
get your desired stats.
On Mon, Oct 24, 2011 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
As I am using the ${CALLERID(num)} to be part of the filename that I am
recording it, I am facing the following problem:
If the incoming call (via PSTN) reached for an extension (which is the
reception), and then the extension transferred the call to the proper
person, and we need to do recording for the call at this proper person, the
problem that at this point the ${CALLERID(num)} will represnt the reception
guy extension and not the original caller id of the caller who called from
outside via the PSTN. How can I get this original caller id?
Regards
Bilal
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[asterisk-users] Voicemail: playing a message to give option if need to transfer for operator
Hi All;
Is it possible to be part of the voicemail to play a wave message as following:
The person you are calling is not available, press 0 if you need to call the
operator or 1 to leave voice message?
I know that I can do this as part of the extensions.conf, but I am looking if
it possible to be part of the voicemail function it self?
Actually below is the macro that I am using it for the voicemail, but really I
am facing a troubles and it is not working properly. I would like to ask about
somthing: the macro is not considered to be a context? In other words, if I
used the Background function, so it come back to the original context or it
apply the rules in the macro?
[macro-voicemail]
exten = 108,1,Dial(${ARG1},20)
exten = 108,2,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = 108,3,Goto(IncomingPSTN,t,3)
exten = s,1,Dial(${ARG1},20)
exten = s,2,Background(voicemail-opt)
exten = s,102,Background(voicemail-opt)
exten = 1,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = 1,2,Goto(IncomingPSTN,t,3)
exten = 0,1,Macro(voicemail,SIP/108)
exten = i,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = i,2,Hangup()
exten = t,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = t,2,Goto(IncomingPSTN,t,3)
exten = t,3,Hangup()
exten = a,1,VoicemailMain(${MACRO_EXTEN})
;
Regards
Bilal
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Re: [asterisk-users] Voicemail: playing a message to give option if need to transfer for operator
Yes, Macro will return to calling context BUT use GoSub instead and your
life will be easy. Forget using Macro whenever you need to get user input in
there.
On Mon, Oct 24, 2011 at 2:35 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Is it possible to be part of the voicemail to play a wave message as
following:
The person you are calling is not available, press 0 if you need to call
the operator or 1 to leave voice message?
I know that I can do this as part of the extensions.conf, but I am looking
if it possible to be part of the voicemail function it self?
Actually below is the macro that I am using it for the voicemail, but
really I am facing a troubles and it is not working properly. I would like
to ask about somthing: the macro is not considered to be a context? In other
words, if I used the Background function, so it come back to the original
context or it apply the rules in the macro?
[macro-voicemail]
exten = 108,1,Dial(${ARG1},20)
exten = 108,2,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = 108,3,Goto(IncomingPSTN,t,3)
exten = s,1,Dial(${ARG1},20)
exten = s,2,Background(voicemail-opt)
exten = s,102,Background(voicemail-opt)
exten = 1,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = 1,2,Goto(IncomingPSTN,t,3)
exten = 0,1,Macro(voicemail,SIP/108)
exten = i,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = i,2,Hangup()
exten = t,1,Voicemail(${MACRO_EXTEN}@Internal,u)
exten = t,2,Goto(IncomingPSTN,t,3)
exten = t,3,Hangup()
exten = a,1,VoicemailMain(${MACRO_EXTEN})
;
Regards
Bilal
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Re: [asterisk-users] ${CALLERID(num)} after doing transfer from extension to extension
On Monday 24 October 2011, bilal ghayyad wrote:
If the incoming call (via PSTN) reached for an extension (which is the
reception), and then the extension transferred the call to the proper
person, and we need to do recording for the call at this proper person,
the problem that at this point the ${CALLERID(num)} will represnt the
reception guy extension and not the original caller id of the caller who
called from outside via the PSTN. How can I get this original caller id?
As soon as the incoming call lands in a context, store the caller's number in
a variable; for instance,
Set(ORIG_NUM=${CALLERID(num)})
and then when building up the call filename, just use ${ORIG_NUM} instead of
=${CALLERID(num)}
--
AJS
Answers come *after* questions.
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[asterisk-users] Storing a variable at a context and using it in another context
Hi All;
Is it possible to store a variable at context and using it in another context
or in the MACRO? For example, how I can store the ${CALLERID(num)} in a
variable and use it in another context or in a MACRO?
Regards
Bilal
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Re: [asterisk-users] Storing a variable at a context and using it in another context
Try using variables between macros and contexts without doing anything. It
works fine for me in asterisk 1.6.13+. If not then use _ before variable
name.
On Mon, Oct 24, 2011 at 2:46 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Is it possible to store a variable at context and using it in another
context or in the MACRO? For example, how I can store the ${CALLERID(num)}
in a variable and use it in another context or in a MACRO?
Regards
Bilal
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Re: [asterisk-users] Storing a variable at a context and using it in another context
Declare the global variable
On Mon, Oct 24, 2011 at 15:16, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Is it possible to store a variable at context and using it in another
context or in the MACRO? For example, how I can store the ${CALLERID(num)}
in a variable and use it in another context or in a MACRO?
Regards
Bilal
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Amit Anand
+91 9818559898
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[asterisk-users] device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
Hello Have a setup of asterisk with realtime SIP devices. Trying to organise monitoring of my SIP devices. Once device registered, its state becomes NOT_INUSE (result of DEVICE_STATE(SIP/device) function). Simulating of device breakage - powerdown it. Waiting for a while (minute or two), retrieving DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer device load no result. What I did not configured? My sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;tos_sip = none ;tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = alaw type = friend host=dynamic context = noop-context dtmfmode=rfc2833 nat=no rtcachefriends=yes qualify=1 deny=0.0.0.0/0.0.0.0 permit=172.30.8.0/255.255.255.0 regards, Yaroslav -- _ -- 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] Storing a variable at a context and using it in another context
On Monday 24 October 2011, bilal ghayyad wrote:
Hi All;
Is it possible to store a variable at context and using it in another
context or in the MACRO? For example, how I can store the ${CALLERID(num)}
in a variable and use it in another context or in a MACRO?
Variables are local to channels (to all intents and purposes, call legs),
not contexts. If you call a macro or GoTo() another context, any variables
you defined in the calling context should be preserved. They will only be
forgotten when the channel is destroyed (to all intents and purposes, when
the call is hung up).
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Strange behavior over Zap chennels
Hi Doug, Thanks for the reply. Unfortunately I can't get my telco do do anything because I can't provide proof that is a problem with their lines. 2011/10/22 Doug Lytle supp...@drdos.info Richard Reina wrote: I have a server that is hooked to a channel bank (Adit 600). It has eight lines coming from a T1 through this adit's FXO card. I have a very similar setup, but my Adit 600 is used to provide FAX only. Voice calls are PRI. And, my system hasn't been rebooted in over 2 years. It's been a while since I've logged into my Adit, but I believe it has logging. The usual suspects for analog lines are bad cable, water in the telco equipment, Thunder storms. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- 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-**usershttp://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-dev] SIP, NAT, security concerns, oh my!
On Sun, Oct 23, 2011 at 10:06 PM, Andrew Latham lath...@gmail.com wrote: OpenVPN is the solution to all NAT issues. With at least the SNOM 370 supporting it and the phone can be setup as a OpenVPN gateway as well, for very small offices, it is a great phone. Linux based and you install their OpenVPN firmware and then setup the PC port on the phone to bridge (that is what I do anyways) traffic, plug it into a switch, configure whatever, no split tunnel, and completely secure site to site VPN and no NAT issues, I would do this with something like a five workstation office at most. Alot of bang for the Buck with the SNOM 370. If you already know OpenVPN it is a breeze, and there are tons of howtos specific to the SNOM, documentation is good too. This could also be done with any number of other solutions from the WRT54GS whatever, or just a little boxen for VoIP over the VPN tunnel, and other traffic out the default gateway. I just like to secure small remote sites so I can monitor, administer, and enforce network usage policy. That is coming from a Private Military Company background. I don't want any data not going through a voice or data tunnel to Equinix. Then some small Top Secret installation in a remote area doesn't wind up infecting their little LAN. Set it up and put it in a fly-away quarter sized rugged rack with casters. This approach has saved days and days of troubleshooting with people who cannot understand me by language or technology or whatever. It took a bit of work to plan the whole thing out, mesh the systems to route over the tunnel with fault tolerance, but certainly a worth the time. Short, OpenVPN can get you around all SIP/NAT/Security issues, since the tunnel is on a singe port, the big idea behind IAX2 but much better, it is still SIP. You can lock down everything using OpenVPN to prevent problems and allow simple management of global networks. All traffic passes through a few devices, giving you almost total security at a few key points. Vyatta paid version in a VM or Bare Metal is my internet facing firewall. It is so powerful, cheap, and the dev team there is great. They have helped me directly a number of times. I like to have NTOP, Webmin and Asterisk on most of these boxen, but I don't want to install a bunch of extra junk beyond the Vyatta ISO and the packages I find handy. That is my approach until IPV6 ever come out, or some other variant. Thanks, Steve Totaro Thanks, Steve Totaro I use a lot of Zentyal for OpenVPN plus networking fun. I did hear from a snom engineer that they got the openvpn working with a limited functionality on the snom 300 and other models. Direct to you email because I wanted to mention your double signature... -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ Andrew, I have one client setup to add the signature and the others are manual. On occasion, there is a double signature. Thanks for pointing it out, but content of the post is issue. My signature is extremely minimal, not spammy like many. I don't know how you can have Limited Functionality with OpenVPN. Not sure who you talked to or when, but it works great. Read some of the howtos and see that you can use one phone to create a site to site VPN using bridge-utils. For real networking fun, I would not use a phone, the OpenVPN is just for a what I said, one off sites or totally mobile hard phones. Vyatta has training and 24/7 support for $1k per server. The project you pointed out looks cool from a couple of screenshots, I will load it up on VMWare if there is an image. Vyatta paid version is so cheap, a great business plan, the backing of former Cisco execs, and is very robust. GUI needs some work, that is why I put NTOP and Webmin on it, but their engineer, the main guy, an exec, and myself have had conference calls about additional functionality. They don't want to incorporate and will not support other projects and are working on their own GUI, I totally understand. I just ask for the tools to build from source and not mess anything up from Vyatta's and my viewpoint. Beyond that, I completely understand where their demarc is from stock software to whatever I build. Just to point out that Vyatta is Sand Scrit and means Open, I thought that was cool. They are very similar to Asterisk, at least as far as having a commercial and open source offering. I see Vyatta staying the course, having legs, and not going the way of vapor. Many places are using it now, I was surprised by some of the fortune 500s, and the job descriptions I get with Vyatta listed along with Cisco. Thanks, Steve Totaro -- _ -- 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
Re: [asterisk-users] Asterisk call transfers not working
Hi everibody:
Sorry, I want to relive this issue. I still have the problem, if somebody
could help me will be appreciated. Tks.
*Ramiro PAZ
MASTERLINE LOGISTICS
***
On Wed, Oct 19, 2011 at 3:25 PM, Ramiro Paz ram...@masterline-logistics.com
wrote:
Hi Danny, Warren:
This is what I found in extensions_additional.conf:
[from-internal-additional]
include = from-internal-additional-custom
include = app-dialvm
include = app-vmmain
include = app-recordings
include = app-callwaiting-cwoff
include = app-callwaiting-cwon
include = ext-group
include = grps
include = ext-queues
include = app-queue-toggle
include = app-calltrace
include = app-directory
include = app-echo-test
include = app-speakextennum
include = app-speakingclock
include = app-cf-busy-off
include = app-cf-busy-off-any
include = app-cf-busy-on
include = app-cf-off
include = app-cf-off-any
include = app-cf-on
include = app-cf-unavailable-off
include = app-cf-unavailable-on
include = app-cf-toggle
include = app-fmf-toggle
include = ext-findmefollow
include = fmgrps
include = app-userlogonoff
include = ext-local-confirm
include = findmefollow-ringallv2
include = app-pickup
include = app-zapbarge
include = app-chanspy
include = ext-test
include = ext-local
include = outbound-allroutes
exten = h,1,Hangup
; end of [from-internal-additional]
There is nothing for [from-internal-custom]. I mean extensions_custom.conf
is empty.
Just in case, Warren is right, we're using FXS/DAHDI endpoints. Thanks for
your time.
*
Ramiro PAZ
**MASTERLINE LOGISTICS*
On Wed, Oct 19, 2011 at 12:59 PM, Warren Selby wcse...@selbytech.comwrote:
On Wed, Oct 19, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.comwrote:
Or you could just add these lines to [from-internal-xfer]
Exten = _X,1,Dial(SIP/${EXTEN},30,iKkTtt)
Exten = _XX,1,Dial(SIP/${EXTEN},30,iKkTt)
** **
If you have 3 or 4 digit extensions you would need these lines
Exten = _XXX,1,Dial(SIP/${EXTEN},30,iKkTtt)
Exten = _,1,Dial(SIP/${EXTEN},30,iKkTt)
Except he's not sending to SIP endpoints, he's sending to FXS / DAHDI
endpoints. So the syntax would be a bit more specific based on which
extension was being dialed and which port it was hooked up to on the card.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
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