Re: [asterisk-users] checking if a phone number is UP
Brilliant! `log the call and busy out` is the thing I was missing. thank you so much On Thu, Feb 9, 2012 at 4:59 PM, Bryant Zimmerman wrote: > We designed our solution the following way. > > We have several land line numbers hooked to an asterisk testing server. > The testing server places one call every X seconds per line to a number we > want to test . We cycle through each number in our testing pool. Each > number on average is tested once every 30 min this can be adjusted by the > dial rate and the number of test lines in the outbound calling pool. When > a call comes from one of our test numbers our inbound dial plans log the > call and busy's out. So the test call is not answered and no call charge is > assessed per carrier. To verify that a test succeeded the testing server > checks the database after it gets a busy. By design if a call comes in it > is checked before any line counts are tested so this method never effects > the customers line counts. We also have a full audio/dtmf test that is > run once a day per number. This means that the first test call of the day > is actually answered and a DTMF and audio hand shake is done. Both ends > log the result in a database. > > We catch vendor issues with these methods and it allows us to open tickets > and resolve issues before a customer knows there might be an issue. Our > vendors hate the system as we tend to catch any hiccup they may be having > as well. Several of them are mistified how we can open tickets on issues > consistently before they know they have an issue. > Bryant > > > -- > *From*: "Aurimas Skirgaila" > *Sent*: Thursday, February 09, 2012 9:34 AM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > > *Subject*: [asterisk-users] checking if a phone number is UP > > > hi, > > We have a phone number from third party provider which is used for > inbound calls. How could I monitor if this phone number is reachable? > > the initial idea doesn't sound elegant: > - on my SIP server I set couple seconds of ringing before Answer(). > - the monitoring server calls to that phone number for few seconds, checks > if it "hears" the ringing and hangs up the call. > > ** > I use Nagios to check if my services are UP using check_sip, but it this > situation I'm more concerned about my DID provider than my server. It's > just like pinging a phone number. > > > > Thank you, > Aurimas > > > -- > _ > -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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
[asterisk-users] checking if a phone number is UP
hi, We have a phone number from third party provider which is used for inbound calls. How could I monitor if this *phone number* is reachable? the initial idea doesn't sound elegant: - on my SIP server I set couple seconds of ringing before Answer(). - the monitoring server calls to that phone number for few seconds, checks if it "hears" the ringing and hangs up the call. ** I use Nagios to check if my services are UP using check_sip, but it this situation I'm more concerned about my DID provider than my server. It's just like pinging a phone number. Thank you, Aurimas -- _ -- 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] amd detect answering machine
Hi,
do noop(${AMDCAUSE}) after exten => 1,1,AMD() , run some test calls and
find out why the call was detected as Answering Machine and adjust amd.conf
accordingly. if I recall correctly, you can also see the AMD flow in
Asterisk in verbose mode.
I'd suspect low silence_threshold . I usually set it 384, but it's
very dependent on carrier.
On Thu, Feb 2, 2012 at 5:51 PM, Etann wrote:
> **
> Hi,
> I have IVR and when I press 1, asterisk calls my mobile phone.
> If my mobile phone is offline, asterisk transfers to asterisk voicemail.
> I'd like asterisk detects my mobile voicemail and if my mobile voicemail
> answers, asterisk transfers to asterisk voicemail.
> For that, I used AMD.
> So I have problems ! Asterisk detects answering machine everytime!
> How do I do please ?
>
>
> extensions.conf
> [ServeurPro]
> exten => s,1,Ringing()
> exten => s,2,Wait(2)
> exten => s,3,Answer()
> exten => s,4,Set(NbInvalide=0)
> exten => s,5,Set(NbEssai=0)
> exten => s,6,background(${ChmAudio}/ServeurProBienvenu)
> exten => s,7,WaitExten(2)
>
> exten => 1,1,AMD()
> exten => 1,2,GotoIf($["${AMDSTATUS}"="MACHINE"]?1,4)
> exten => 1,3,Dial(SIP/@ippi_outgoing2,40,r)
> exten => 1,4,Voicemail(801@FloriePro,us)
> exten => i,1,Set(NbInvalide=$[${NbInvalide}+1]})
> exten => i,2,Gotoif($["${NbInvalide}" < "3"]?:6)
> exten => i,3,Playback(${ChmAudio}/ErreurSaisie)
> exten => i,4,Playback(${ChmAudio}/RetourMenu)
> exten => i,5,Goto(s,6)
> exten => i,6,Playback(${ChmAudio}/ErreurSaisie)
> exten => i,7,Playback(${ChmAudio}/Aurevoir)
> exten => i,8,Hangup()
> exten => t,1,Set(NbEssai=$[${NbEssai}+1])
> exten => t,2,Gotoif($["${NbEssai}" < "3"]?:5)
> exten => t,3,Playback(${ChmAudio}/DemandeIncomprise)
> exten => t,4,Goto(s,6)
> exten => t,5,PlayBack(${ChmAudio}/Aurevoir)
> exten => t,6,Hangup()
> exten => h,1,noOp("Statut AMD : "${AMDSTATUS})
>
> amd.conf
> [general]
> initial_silence = 2500 ; Maximum silence duration before the greeting.
> ; If exceeded then MACHINE.
> greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE.
> after_greeting_silence = 500 ; Silence after detecting a greeting.
> ; If exceeded then HUMAN
> total_analysis_time = 5000 ; Maximum time allowed for the algorithm to
> decide
> ; on a HUMAN or MACHINE
> min_word_length = 120 ; Minimum duration of Voice to considered as a word
> between_words_silence = 50 ; Minimum duration of silence after a word to
> consider
> ; the audio what follows as a new word
> maximum_number_of_words = 3 ; Maximum number of words in the greeting.
> ; If exceeded then MACHINE
> silence_threshold = 256
>
> Thank you for your reply and for help!
> --
>
> AMICALEMENT
> Manu
>
> SITES WEBS
> Mon site web Officiel (Manu-dpk.net) <http://www.manu-dpk.net/>
> Ecoutez Radio DPK <http://webradio.manu-dpk.net/?page=Radio>
>
> CONTACT
> - E-mail : manuli...@manu-dpk.net
> - Messenger (WLM) : m...@manu-dpk.net
> - Skype : manu-dpk
> --
>
> PS : Pour le respect de l'environnnement, n'imprimez ce mail qu'en cas de
> nécessité.
>
>
> --
> _
> -- 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
>
--
Mvh,
Aurimas Skirgaila
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Re: [asterisk-users] Is this doable?
t; -- > __**__**_ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- Mvh, Aurimas Skirgaila -- _ -- 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] AMD tweaking
Thank you, Alex yes, I expect the pickup time to vary within 1 second (it's just a guess). If I have to tolerate higher bias, so I would start doubting about the efficiency of this method. On Mon, May 16, 2011 at 4:00 PM, Alex Balashov wrote: > You would have to make the tolerance of variance fairly high. There are > many reasons why pickup time by a mechanical device such as an answering > machine or a fax machine may vary quite significantly. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On May 16, 2011, at 8:56 AM, Aurimas Skirgaila > wrote: > > > Hi, > > > > long time ago, I came up with an optimal configuration set for my > environment - good detection and little false positives. Unfortunately some > people are always being detected as Answering Machines. > > > > I'm not up to re-adjust my precious balance of > initial_silence/max_words/... , so I'm thinking about to check if the pickup > time is equal to the pickup time when the same phone number was previously > detected as AM - if the pickup time is different from the last time, - it's > HUMAN, else proceed standard AMD(). > > > > has anyone done this before,so I wouldn't be reinventing bicycle? > > > > > > -- > > Mvh, > > Aurimas Skirgaila > > -- > > _ > > -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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
[asterisk-users] AMD tweaking
Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was previously detected as AM - if the pickup time is different from the last time, - it's HUMAN, else proceed standard AMD(). has anyone done this before,so I wouldn't be reinventing bicycle? -- Mvh, Aurimas Skirgaila -- _ -- 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] malformed SIP / routing issue
Surely. B responds "404 Not Found.", as it's not configured to receive these SIP packets. provider P sends to correct IP, and moreover B has no external IP. On Mon, Dec 27, 2010 at 3:54 PM, voipas wrote: > > Hi, > > Have you checked SIP messages on B server? Maybe your provider P > sends traffic to incorrect IP. > > -- > Best Regards, > Giedrius > > -- > _ > -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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
[asterisk-users] malformed SIP / routing issue
Hi, I wonder what conditions might lead, that SIP packets from provider P destined to my external SIP server A, are reaching my internal SIP server B? the fun factor is that internal B server is used for outbound calls via the same provider P. I found no routing issues. Is it possible to build SIP header specifying the final destination - the internal IP address? -- Mvh, Aurimas Skirgaila -- _ -- 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] load balance with 2 wan connections
some providers do serve inbound by sending the traffic to exact IP, some do accept the registers from any IP. in second case for Inbound failover, you might just to "register => " using another interface/IP address. here a new question arose: how to "sip-ping" some phone number to see if it's alive? On Mon, Dec 27, 2010 at 11:51 AM, Sherwood McGowan < sherwood.mcgo...@gmail.com> wrote: > The biggest issue with any solution to use two different providers for > your IP service that will be used by your VOIP provider to deliver > calls to your Asterisk server, is that each internet service will have > a separate address. Therefore, for INBOUND calls, your VOIP provider > will have to do the load balancing. For outbound calls, it won't be > that hard as long as your provider allows you to send calls from both > IP addresses. > > -- > _ > -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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] Licensing of Default MOH
Thank you guys for making me sure about this question and pointing to useful resourses. * yes, I might be the first one because googling didn't give me any certain answer On Fri, Oct 22, 2010 at 3:06 PM, Tzafrir Cohen wrote: > On Fri, Oct 22, 2010 at 12:44:18PM +0300, Aurimas Skirgaila wrote: > > Hi, > > > > I wonder if I may freely use the default soundfiles that came with > asterisk > > (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? > > > > Are there any official sources of royalty free music? > > http://downloads.asterisk.org/pub/telephony/sounds/ > > Specifically the "opsound" music-on-hold files are all licensed under a > the terms of the CC-BY-SA 3.0. > http://creativecommons.org/licenses/by-sa/3.0/ > > This explicitly allows "public performance" and such. I suspect it > should be good for you. > > (Did I mention I'm not a lawyer?) > > -- > Tzafrir Cohen > icq#16849755 > jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > -- > _ > -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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
[asterisk-users] Licensing of Default MOH
Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -- _ -- 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] E1 check with nagios, how to?
what do you want to monitor? I ended up with MRTG graphing the Incoming/Ringing/Established calls. On Tue, Sep 28, 2010 at 4:22 PM, Dario Quiroz wrote: > We need to monitorate the E1 with nagios, somebody did this? any ideia? > Thanks in advance! > > -- > Atenciosamente, > > --- > > Dario Quiroz > > (71) 9275-9080 >gtalk: darioqui...@gmail.com > > --- > > -- > _ > -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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] AMD message
I took a quick&dirty solution in your case, when I wanted to pick up samples for analyzing AMD. That was full recording of all outgoing calls (application Monitor() ), and then I've selected only the phone numbers which were detected as Answering Machines. On Wed, Aug 25, 2010 at 10:14 AM, Tino wrote: > Yes, we need to record the message > > > > On Wed, Aug 25, 2010 at 12:35 PM, Matt Riddell wrote: > >> On 20/08/10 1:52 AM, Tino wrote: >> > Hello, >> > >> > Is there a way to capture the answering machine message when the dialer >> > detects the answering machine. >> >> Record? >> >> -- >> Cheers, >> >> Matt Riddell >> ___ >> >> http://www.venturevoip.com/news.php (Daily Asterisk News) >> http://www.venturevoip.com/exchange.php (Full ITSP Solution) >> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) >> >> -- >> _ >> -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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] Tweaking AMD in Asterisk
in my case it's 0.1 second and I can confirm, that on SIP channels it really helps. On Wed, Aug 4, 2010 at 8:51 PM, Danny Nicholas wrote: > *>From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tino > >*Subject:* Re: [asterisk-users] Tweaking AMD in Asterisk > > > > >Thanks Danny, What should be the length of audio file ? > > I’m supposing that 3 to 5 seconds should be ok. > > -- > _ > -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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] Tweaking AMD in Asterisk
Hi, the basic settings are pretty good ones. What I did to do improve the performance and prevent the false positives, I started to recorded every call, and analyzed every incorrect detection :) Fairly soon I came with optimal set for my environment: initial_silence= 2500 greeting = 1500 after_greeting_silence = 300 total_analysis_time= 5000 min_word_length= 120 between_words_silence = 50 maximum_number_of_words= 4 ; it's usuall to pickup saying "Jon Anderssen, hello" in here silence_threshold = 384 by the way, for outgoing SIP calls you might want to do this Background trick as it helped me a lot regarding AMD on SIP. exten => _X.,n,Background(blank_audio) exten => _X.,n,AMD On Wed, Aug 4, 2010 at 5:08 PM, Tino wrote: > Hello , > > I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My > current values are > > AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx > 25-30 % of all answering machines. > > Anybody have any suggestion to improve the accuracy of AMD. > > Thanks > > > > > -- > _ > -- 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 > -- Mvh, Aurimas Skirgaila -- _ -- 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] identify caller hangup or callee hangup?
so you can go for it.
the basic idea:
Dian(***,***,g)
Noop("Called party hung up first")
Hangup
On Mon, May 17, 2010 at 1:23 PM, Vardan wrote:
> Yes, I know about this option, You can you this also, but, how you want
> to see what leg was hangup the channel?
>
> Olivier wrote:
> > Have you looked at Dial's g option ?
> > "*g*: When the called party hangs up, exit to execute more commands in
> > the current context."
> >
>
> --
> Vardan Harutyunyan,
> Senior System Administrator
>
> Enterprise Incubator Foundation
> 123 Hovsep Emin Street,
> Yerevan 0051, Republic of Armenia
> Tel: + 374 10 219735
> Fax: + 374 10 219777
> E-mail: i...@eif.am
> www.eif-it.com
>
>
> --
> _
> -- 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
>
--
Mvh,
Aurimas Skirgaila
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] is it possible to connect Digium TE420 and Cisco card?
just FYI, to complete the topic. The problem was caused by failed PVDM module in Cisco server. > Hello, > I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to providers Cisco 2800 with VWIC-1MFT-E1 card. > the same card runs fine with another E1 provider. > TE420 led's lite green. > Message type: RELEASE COMPLETE (90) < [08 02 80 ac] < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) < Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (resource unavailable) (2) ] -- Processing IE 8 (cs0, Cause) q931.c:3760 q931_receive: call 32770 on channel 1 enters state 0 (Null) Sending Receiver Ready (31) -- Channel 0/1, span 1 got hangup, cause 44 -- Forcing restart of channel 0/1 on span 1 since channel reported in use q931.c:3000 q931_restart: call 32768 on channel 1 enters state 62 (Restart) > [zaptel.conf] span=1,1,0,ccs,hdb3,crc4#switching timing between 0/1 does not have any effect, bchan=1-15,17-31 dchan=16 > [zapata.conf] group=1 pridialplan = unknown switchtype=euroisdn context = trunk-1 signalling = pri_net channel => 1-15,17-31 > Hardware - Dell PowerEdge R200. Now moved onto barebone test server, but same errors persist. > -- Mvh, Aurimas Skirgaila -- Mvh, Aurimas Skirgaila -- _ -- 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] SIP Dialplan Failover Solution
Hi,
I do use the first solution based on DIALSTATUS variable. (
http://www.voip-info.org/wiki/view/Superdial+macro)
since it's included to a separated context named [superdial-macro], I don't
have to repeat it over and over, so the fact that it's not a oneliner
doesn't bother me at all :)
On Tue, Apr 6, 2010 at 3:37 PM, Alexandru Oniciuc <
alexandru.onic...@trivenet.it> wrote:
> Hello list,
>
>
>
> I need a hand to find the best dialplan failover solution
> when using two SIP Trunks.
>
>
>
> My reasons to do failover are:
>
> a) one of the two providers could be unreachable
>
> b) both providers may be UP but one of them could return a SIP error
> message (maybe caused by DOWN E1s)
>
>
>
> Googling I found a few possible solutions:
>
>
>
> 1. Using DIALSTATUS variable.
>
>
>
> 2. Dialing in sequence:
>
>exten => _X.,1,Dial(SIP/${TRUNK1}/${EXTEN})
>
>exten => _X.,2,Dial(SIP/${TRUNK2}/${EXTEN})
>
>
>
> 3. ChanIsAvail
>
>
>
>
>
>
>
> Using the first method it’s possible to get the CONGESTION
> and CHANUNAVAIL status which pretty much solves my problem but it takes more
> than 2 lines of dialplan(I like one liners).
>
> The second solution requires less space in the dialplan but it should work
> only when the called party is busy (or maybe even when the first trunk is
> down).
>
>
>
> Is there a clean way to send the call to the second SIP provider if the
> first one is unreachable or spits out sip error messages?
>
>
>
> Thanks in advance,
>
>
>
> Alex
>
> --
> _
> -- 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
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Mvh,
Aurimas Skirgaila
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_
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Re: [asterisk-users] Restarting Asterisk using a script - Thanks to all -
Hi, 1) It would be nice to find out the root reason that forces you to restart the Asterisk. I do run Aheeva with decently high uptimes. 2) Both "a" and "b" methods of Jose P. Espinal are functional, but if I'm having a failure, I up to grab the putty and investigate what's going on there :) How often are you having them? 3) what's the another service related to Aheeva, that requires to be restarted? > > Hi there, a. You could (maybe) use PHP and send some command via POST, and (after secure/validating the command) use 'exec()' function in php, or 'system()' function. Note: that would require to have a webserver with php installed on it. And allowing the user under which the webserver runs, to restart asterisk via sudoers file. b. You could use a shellscript that sends the command via SSH. In order to avoid password prompt, you could generate a RSA (or DSA) key pair on the machine that will send the command, and copy the rsa_key.pub content on your asterisk box 'authorized_keys'. That would allow you to execute the command remotely via SSH without having to insert the password manually. Note: you could consider using a very limited user on the asterisk box, and with sudoers file allowing it just to restart Asterisk. Regards, Amine Mrichcha wrote: > Hi All, > > I do have asterisk installed for a call center and I would like to know > if it is possible to create a scipt and execute it from a PC connected > to the Network without accessing the server. This script should restart > asterisk and another service related to aheeva. > > The problem now is that each time I have to access using PUTY to the > server to start and run services manually. > > Service asterisk restart > > Any help would be appreciated, sorry if it is a newbie question. > > Regards, > > Am > -- Jose P. Espinal http://www.eSlackware.com <http://www.eslackware.com/> IRC: Khratos @ #asterisk / -doc / -bugs -- Mvh, Aurimas Skirgaila -- _ -- 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
[asterisk-users] is it possible to connect Digium TE420 and Cisco card?
Hello, I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to providers Cisco 2800 with VWIC-1MFT-E1 card. the same card runs fine with another E1 provider. TE420 led's lite green. Message type: RELEASE COMPLETE (90) < [08 02 80 ac] < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) < Ext: 1 Cause: Requested channel not available (44), class = Network Congestion (resource unavailable) (2) ] -- Processing IE 8 (cs0, Cause) q931.c:3760 q931_receive: call 32770 on channel 1 enters state 0 (Null) Sending Receiver Ready (31) -- Channel 0/1, span 1 got hangup, cause 44 -- Forcing restart of channel 0/1 on span 1 since channel reported in use q931.c:3000 q931_restart: call 32768 on channel 1 enters state 62 (Restart) [zaptel.conf] span=1,1,0,ccs,hdb3,crc4#switching timing between 0/1 does not have any effect, bchan=1-15,17-31 dchan=16 [zapata.conf] group=1 pridialplan = unknown switchtype=euroisdn context = trunk-1 signalling = pri_net channel => 1-15,17-31 Hardware - Dell PowerEdge R200. Now moved onto barebone test server, but same errors persist. -- Mvh, Aurimas Skirgaila -- _ -- 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] configure the sound for inbound calls
Hi Salah, what's the problem? For playbacks upload a soundfile to your asterisk /var/lib/asterisk/sounds/hello.wav and setup the Routing Script to "Playback(hello)"; reload asterisk and watch asterisk and aheeva logfiles. And yes there is a possibility to retrieve customer information from your CRM as long as you get customer phone number. > Hello All, > > I do have asterisk installed for a call centre with aheeva application and > i would like to know how to configure the sound for the inbound calls and > if > there is any possibility for agent to receive a file with the phone number > and name of clients: For your information there is no problem related to > the > outbound call > > An help would be appreciated > > Kind Regards > > Salah. > -- Mvh, Aurimas Skirgaila -- _ -- 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] IAX2 trunking with Older Asterisk version ?
Asterisk versions may differ. I do IAX trunk successfully even between Asterisk 1.0.2 and 1.4.xx please post your Dial command. On Fri, May 29, 2009 at 11:33 AM, Tharanga wrote: > Hi All, > > Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and > asterisk 1.2.14 ? > > i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but > it gave an error - > > 1.2.14 End - Error Msg > WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by > 147.120.203.71: No authority found > > 1.2 END , IAX.conf > > [trunk14] > type=friend > host=147.120.203.71 > secret=test123 > context=sip,sip2,sip3 > permit=0.0.0.0/0.0.0.0 > > > 1.6.1.0 End - Error Msg > NOTICE[9854]: chan_iax2.c:8782 socket_process: Rejected connect attempt > from 147.120.203.69, who was trying to reach '4567@' > > [trunk14] > type=friend > host=147.120.203.67 > secret=test123 > context=sip,sip2,sip3 > keyrotate=off > permit=0.0.0.0/0.0.0.0 > > > what could be the problem ? do i need to have the same asterisk versions > both side ? > > Thanks, > Tharanga > > > ___ > -- Bandwidth and Colocation 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 > -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation 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
Re: [asterisk-users] SIP Trunk groups
AFAIK, unfortunatelly it's not the same as with ZAP channels where you can
group multiple lines together.
I ended up using slightly modified superdial macro:
http://www.voip-info.org/wiki/view/Superdial+macro.
if you add new gateway it's not necesarry to edit the macro, just add new
line in dialing context.
[out_via_superdial]
exten =>
s,1,Macro(superdial,IAX2/voip1/${tfnumber}voip,1,yourname,8005551234,voipjet)
exten =>
s,2,Macro(superdial,IAX2/alpeh-com/${tfnumber}voip,1,yourname,8005551234,aleph)
...
exten =>
s,9,Macro(superdial,IAX2/orange/${tfnumber}voip,1,yourname,8005551234,orange)
On 5/27/09, Mariano Lecuona wrote:
>
> Hey all,
>
> I have 2 GSM to Voip gateways and probably we will grow up to 4 more
> gateways. I already created a macro to make failover happen between
> gateways, but can imagine that everytime I add a new gateway I will need to
> modify the macro. The initial intention of this macro was to failover
> between different techonolgies.
> So I was hoping to create a Sip Trunk group using the same idea as
> truckgroup under dahdi but for sip trunks.
>
> Is that possible?, have you ever done this before?
>
> My Idea is:
>
> sip_trunk1 = SIP/gateway1
> sip_trunk2 = SIP/gateway2
> sip_trunk3 = SIP/gateway3
>
> gsm_trunkgoup = sip_trunk1 ; sip_trunk2 ; sip_trunk3
>
>
> [user]
>
> exten = _0.,1,wait()
> exten = _0.,n,Dial(gsm_trunkgoup/${ exten:1},30)
> exten = _0.,n,Hangup
>
> Thanks,
>
> --
> --
> *Mariano Lecuona*
>
>
> ___
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>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Mvh,
Aurimas Skirgaila
___
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Re: [asterisk-users] SIP over VPN
run tcpdump, while trying to connect to asterisk to see what ports are requiered. default SIP port is UDP 5060, but as mentioned before all your traffic should go over VPN so port openening shouldn't be a problem On Wed, May 27, 2009 at 8:40 AM, Marco Sambo wrote: > Ok, > but if I want to open only SIP port on firewall, which ones? I have the > following situation: > > computer A (softphone) firewall computer B (asterisk) > > and I dont' want to open any ports, only SIP and voice. > > > > > 2009/5/26 David Gibbons > >> Assuming you mean the firewall in front of the client, you don’t need >> to open any ports as long as the VPN client is tunneling all traffic to and >> from the Asterisk server. >> >> >> >> I set NAT=yes in the config file for the extensions behind a VPN. >> >> >> >> -Dave >> >> >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo >> *Sent:* Tuesday, May 26, 2009 11:21 AM >> *To:* asterisk-users@lists.digium.com >> *Subject:* [asterisk-users] SIP over VPN >> >> >> >> Hi all, >> I have a question. I have a VPN and I want to use a SIP softphone on my >> notebook using with asterisk. But I have some problem with firewall and >> port. >> Someone knows which ports I should open on my firewall??? I can't connect >> ... >> >> Thanks all. >> >> Marco >> >> ___ >> -- Bandwidth and Colocation 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 >> > > > ___ > -- Bandwidth and Colocation 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 > -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation 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
Re: [asterisk-users] QoS & VPN
On Fri, May 8, 2009 at 3:45 PM, Jeff LaCoursiere wrote: > > On Fri, 8 May 2009, Aurimas Skirgaila wrote: > > > Despite the VPN overhead, running VOIP through VPN is good idea because > VPN > > reorders encapsulated UDP packets in correct order. Security matters as > > well. > > Reorders? How so? I think it will maintain the order, only if they have > arrived in the "correct" order. UDP doesn't guarantee that over long way packets arrive in correct order, while TCP based VPN would sort them correctly ;) well, I'm not sure if all kinds of VPN are SSL/TCP based. The author mentioned remote offices so this might be useful for him. > > > > > > I'd suggest to route VNC packets rather over internet than VPN (so do I), > as > > VPN usually has the highest priority. > > > > Unless QoS is implemented packets are first come first served. There is > no "usually has the highest priority". Routing one over the Internet > versus over the VPN won't change that priority. > ok. probably I've misread somewhere about switches which QoS enabled is by default. By the way we do ask our ISP to prioritize VPN packets and they do. > > j > > > On Thu, May 7, 2009 at 11:33 PM, Roberto Piola >wrote: > > > >> I do not have examples, but if you are using the 1700 series router in > >> order to originate the ipsec vpn, you may use command qos pre-classify > >> (please search for it on cco.cisco.com) > >> > >> On Thu, May 7, 2009 at 9:54 PM, Brent Davidson < > >> br...@texascountrytitle.com> wrote: > >> > >>> I've got multiple satellite office all linked back to the main office > >>> via VPN. Each office has their own asterisk server which registers > back > >>> to the main office's Asterisk server. Each office also has a 1Mb > >>> downstream / 384k - 768k upstream connection. The branches are using > >>> Speex for their connections back to the main office. The issue I'm > >>> having is that there are times that I need to VNC in to machines at the > >>> various offices for tech support while the user is also on the phone. > >>> Unfortunately the VNC connection apparently takes priority and makes it > >>> impossible for me to understand anything the person on the phone is > >>> saying, although they can still hear me fine. > >>> > >>> Our Main office uses a Cisco PIX 506 for the main firewall and VPN > >>> concentrator. Each branch office used a Cisco 1700 series router with > >>> IPSec enabled in the IOS. Is there any sort of QoS I can turn on on > the > >>> main router or the branch routers to make sure the voice quality takes > >>> precedence over the VNC? (Any example configs would be greatly > >>> appreciated) > >>> > >>> Would I be better off routing the voice packets over the internet > rather > >>> than the VPN, and could I safely do that without exposing the asterisk > >>> boxes to unnecessary security risks? (At present all of our asterisk > >>> boxes are behind the firewalls and only talk to each other over the > >>> VPN. All PSTN connection is done through TDM boards so they have no > >>> direct exposure to the internet.) > >>> > >>> > >> _______ > >> -- Bandwidth and Colocation 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 > >> > > > > > > > > -- > > Mvh, > > Aurimas Skirgaila > > > > ___ > -- Bandwidth and Colocation 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 > -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation 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
Re: [asterisk-users] QoS & VPN
Despite the VPN overhead, running VOIP through VPN is good idea because VPN reorders encapsulated UDP packets in correct order. Security matters as well. I'd suggest to route VNC packets rather over internet than VPN (so do I), as VPN usually has the highest priority. On Thu, May 7, 2009 at 11:33 PM, Roberto Piola wrote: > I do not have examples, but if you are using the 1700 series router in > order to originate the ipsec vpn, you may use command qos pre-classify > (please search for it on cco.cisco.com) > > On Thu, May 7, 2009 at 9:54 PM, Brent Davidson < > br...@texascountrytitle.com> wrote: > >> I've got multiple satellite office all linked back to the main office >> via VPN. Each office has their own asterisk server which registers back >> to the main office's Asterisk server. Each office also has a 1Mb >> downstream / 384k - 768k upstream connection. The branches are using >> Speex for their connections back to the main office. The issue I'm >> having is that there are times that I need to VNC in to machines at the >> various offices for tech support while the user is also on the phone. >> Unfortunately the VNC connection apparently takes priority and makes it >> impossible for me to understand anything the person on the phone is >> saying, although they can still hear me fine. >> >> Our Main office uses a Cisco PIX 506 for the main firewall and VPN >> concentrator. Each branch office used a Cisco 1700 series router with >> IPSec enabled in the IOS. Is there any sort of QoS I can turn on on the >> main router or the branch routers to make sure the voice quality takes >> precedence over the VNC? (Any example configs would be greatly >> appreciated) >> >> Would I be better off routing the voice packets over the internet rather >> than the VPN, and could I safely do that without exposing the asterisk >> boxes to unnecessary security risks? (At present all of our asterisk >> boxes are behind the firewalls and only talk to each other over the >> VPN. All PSTN connection is done through TDM boards so they have no >> direct exposure to the internet.) >> >> > ___ > -- Bandwidth and Colocation 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 > -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation 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
[asterisk-users] Difference between Transfer and Dial applications
Hi, I wonder what is the difference between Transfer and Dial applications? Could somebody give me an example of Transfer usage? (documentation and voip-info looks poor a bit). I'm using Asterisk 1.2.5 if it matters. -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation 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
