Re: [asterisk-users] How is Queue avg holdtime and avg talktime calculated
On 11 May 2016 at 10:59, Ishfaq Malik <i...@pack-net.co.uk> wrote: > > > On 11 May 2016 at 10:24, Israel Gottlieb <isr...@gmail.com> wrote: > >> >> Hi all >> >> How is avg hold time and avg talktime calculated and over long a period >> of time? >> >> Thanks, >> Israel >> >> > Hi Israel > > If you are referring to the output of the queue show command > then this is the response I received when asking this question previously: > > "Welcome to business logic embedded into app_queue. The issue with the > queue show command rendering stats, is what timeframe are the stats > aggregated over? IIRC, the calculations are using a moving > average[1]. > > [1] http://en.wikipedia.org/wiki/Moving_average; > > If you want to find an average over a fixed period of time, your best bet is > analysing the queue log. We had to do this ourselves when implementing a > Dashboard with figures for the day. > > We found the figures outputted by the queue show command to be > misleading. > > Regards > > > Ish > > > > You can find my previous query and responses here: http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/282395 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 is Queue avg holdtime and avg talktime calculated
On 11 May 2016 at 10:24, Israel Gottlieb <isr...@gmail.com> wrote: > > Hi all > > How is avg hold time and avg talktime calculated and over long a period of > time? > > Thanks, > Israel > > Hi Israel If you are referring to the output of the queue show command then this is the response I received when asking this question previously: "Welcome to business logic embedded into app_queue. The issue with the queue show command rendering stats, is what timeframe are the stats aggregated over? IIRC, the calculations are using a moving average[1]. [1] http://en.wikipedia.org/wiki/Moving_average; If you want to find an average over a fixed period of time, your best bet is analysing the queue log. We had to do this ourselves when implementing a Dashboard with figures for the day. We found the figures outputted by the queue show command to be misleading. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] relative-periodic-announce default value
I've just spotted this line in apps/app_queue.c unsigned int relativeperiodicannounce:1; So I'm going to assume the default is yes. Please let me know if that assumption is wrong. On 12 April 2016 at 16:10, Ishfaq Malik <i...@pack-net.co.uk> wrote: > Hi > > Using asterisk 1.8.23.1 on CentOS6 > > If I do not explicitly set a value for relative-periodic-announce, what > default value will all the queues inherit? > > Regards > > Ish > > -- > > Ishfaq Malik > Department: VOIP Support > Company: Packnet Limited > t: +44 (0)161 660 2350 > f: +44 (0)161 660 9825 > e: i...@pack-net.co.uk > w: http://www.pack-net.co.uk > > Registered Address: PACKNET LIMITED, Duplex 2, Ducie House > 37 Ducie Street > Manchester, M1 2JW > COMPANY REG NO. 04920552 > > -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] relative-periodic-announce default value
Hi Using asterisk 1.8.23.1 on CentOS6 If I do not explicitly set a value for relative-periodic-announce, what default value will all the queues inherit? Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] One phone, many names / Was: Loss of devices registration (pjsip)
On 22 March 2016 at 08:55, A J Stiles <asterisk_l...@earthshod.co.uk> wrote:
> On Monday 21 Mar 2016, somsad khan wrote:
> > Hello guys,
> >
> > I need some help.
> >
> > I have a client coming who wants to assign 5 different numbers to one
> > virtual employee SIP phone at his desk or softphone (Zoiper).
> >
> > which I can assign for the incoming or outgoing both.
> >
> > but the problem is which I might not understanding enough, that,
> >
> > e.g. when line 1 calls the virtual employee will answer “hello this is
> xyz
> > company how can I help you”
> >
> > when line 2 calls the virtual employee will answer “hello this is abc
> > company how can I help you”
> >
> > So it is important the employee can recognize which line is calling as
> they
> > cannot say the wrong company name by mistake!
> >
> > please let me know if there is any possible ways.
>
> Dead easy! Done this before, in a very similar situation (agent has to
> answer with a different name, depending on the number the customer
> dialled).
>
> All you need to do -- as long as the phone you are using is modern enough
> to
> support it -- is have in your dialplan, before the Dial() instruction to
> the
> agent's phone, an instruction like
> Set(CALLERID(name)=something)
> where "something" depends on ${EXTEN}.
>
> For example, if the numbers for the virtual companies are 731615, 701289
> and
> 718182, and the extension to ring is 301, you might do
>
> [from_pstn]
> ; 731615 is company ABC
> exten => 731615,1,NoOp(Call to 731615)
> exten => 731615,n,Set(CALLERID(name)=Company ABC)
> exten => 731615,n,Dial(301)
> exten => 731615,n,HangUp()
>
> ; 701289 is company XYZ
> exten => 701289,1,NoOp(Call to 701289)
> exten => 701289,n,Set(CALLERID(name)=Company XYZ)
> exten => 701289,n,Dial(301)
> exten => 701289,n,HangUp()
>
> ; 718182 is company PQR
> exten => 718182,1,NoOp(Call to 718182)
> exten => 718182,n,Set(CALLERID(name)=Company PQR)
> exten => 718182,n,Dial(301)
> exten => 718182,n,HangUp()
>
>
> For the agent to be able to dial out presenting different caller ID
> numbers,
> use prefixes such as 16, 17, 18 to indicate dialling out as different
> companies;
> strip out the prefix using ${EXTEN:2} to recover the number by skipping two
> digits from the beginning, and Set(CALLERID(num)=) as appropriate.
>
>
>
>
You can also use the A option in the Dial application to play an audio file
to the callee before the channels are bridged.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Loss of devices registration (pjsip)
On 21 March 2016 at 20:32, George Joseph <george.jos...@fairview5.com> wrote: > > > On Mon, Mar 21, 2016 at 11:58 AM, Dmitriy Serov <serov@gmail.com> > wrote: > >> Good day. >> >> Asterisk 13.7.2, res_pjsip. >> There is a problem of loss of registration of several devices. This >> happens not on all devices, but problem devices a lot. >> Below is the log of registration of a contact of one device. >> >> Is suspect two things: >> 1. delete a contact after the contact is added. But, like, it's a feature >> of code that may already be fixed. >> 2. deleting a contact much earlier than the 90 seconds specified during >> the registration >> >> Would be grateful for any clues. >> >> Dmitriy Serov. >> >> expiration settings: >> [common-aor](!) >> type=aor >> qualify_frequency=60 >> default_expiration=120 >> maximum_expiration=600 >> minimum_expiration=90 >> >> log: >> [2016-03-21 20:39:58] VERBOSE[30251] res_pjsip_registrar.c: Added contact >> 'sip:17367@46.39.229.18:37910' to AOR '17367' with expiration of 90 >> seconds >> > The client just registered > > >> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367@46.39.229.18:37910 has been created >> > We added a new contact > > >> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367@46.39.229.18:27143 has been deleted >> > We deleted the old contact > > >> [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367@46.39.229.18:37910 is now Reachable. RTT: >> 41.882 msec >> > We qualified the contact successfully > > >> [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367@46.39.229.18:37910 is now Unreachable. RTT: >> 0.000 msec >> > At the next qualify, we couldn't reach the contact > > [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact ' >> sip:17367@46.39.229.18:60105' to AOR '17367' with expiration of 90 >> seconds >> > The client just registered > > (again) > >> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367@46.39.229.18:60105 has been created >> > We added a new contact > > [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: > Contact 17367/sip:17367@46.39.229.18:37910 has been deleted > We deleted the old contact > > >> [2016-03-21 20:41:06] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367@46.39.229.18:60105 is now Reachable. RTT: >> 44.031 msec >> > We qualified the contact successfully > > >> [2016-03-21 20:42:09] VERBOSE[28019] res_pjsip/pjsip_configuration.c: >> Contact 17367/sip:17367@46.39.229.18:60105 is now Unreachable. RTT: >> 0.000 msec >> > At the next qualify, we couldn't reach the contact > > This looks like a client that's going to sleep or a firewall that's > timing out connections. Asterisk is only deleting the contact on the next > successful register because it's replacing it. You need to figure out why > the qualify is failing and why the client keeps registering. > > > > > Check if the router or firewall has a UDP port timeout option and increase it by a lot (I usually up it to an hour). -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Queues - periodic announce while ringing members
Hi I'm using asterisk 1.8.32.3 on CentOS 6 I've noticed when using queues that the members of the queue stop ringing for the duration of any set periodic announce. Is this the only behaviour possible or is there a way to set the members to continue ringing while the periodic announce plays? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Blocking transfer by SIP REFER on a call by call basis
Hi We are using asterisk 1.8.23.1 on CentOS 6 Is there a way that transferring by SIP REFER can be blocked on a call by call basis? Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Signaling ringing on other extension
On 30 December 2015 at 10:41, Luca Bertoncello <lucab...@lucabert.de> wrote: > Ishfaq Malik <i...@pack-net.co.uk> schrieb: > > > BLF is an interaction between the phones and the server. You need to > > configure function buttons on the phones to display the presence state of > > individual peers that have been set up on the server. > > > > This command in the asterisk cli will help you: > > > > core show hints > > > > If you see an entry for the peer then the server is set up correctly and > if > > the Watchers column > 0 then you have set up the phone correctly. > > Unfortunately the Watchers are 0... > And I didn't find any option on my phone (Thomson ST2022) to enable the > BLF... > > Any other idea? > I wrote a little expect-Script to send the phone an advice and having an > LED > blinking, but I think it is a little bit exaggerated... > > Thanks > Luca Bertoncello > (lucab...@lucabert.de) > > -- > _ > -- 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 > Do you have a link to the user guide for your exact phone model? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Signaling ringing on other extension
Hi Look into Busy Lamp Field/Presence Here's a starting point: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Regards Ish On 29 December 2015 at 15:27, Luca Bertoncello <lucab...@lucabert.de> wrote: > Hi again! > > With the "call pickup"-function I can now pickup a call directed to another > phone in my Asterisk. Very nice. > My problem, now, is that I can't see on my phone, that the other phone (in > another room) rings. > > Is it possible to signal the incoming call on other extension? I use two > phones "Thomson ST2022". > > Thanks a lot > Luca Bertoncello > (lucab...@lucabert.de) > > -- > _ > -- 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 > -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Signaling ringing on other extension
On 30 December 2015 at 10:03, Luca Bertoncello <lucab...@lucabert.de> wrote:
> Ishfaq Malik <i...@pack-net.co.uk> schrieb:
>
> Hi Ishfaq
>
> > Look into Busy Lamp Field/Presence
> >
> > Here's a starting point:
> >
> >
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html
>
> Thanks a lot, but it does not seems to work...
>
> Here my configuration:
>
> sip.conf:
>
> [general]
> allowsubscribe=yes
> subscribecontext = default
> notifyringing = yes
> notifycid = yes
> callcounter = yes
>
> extensions.conf:
>
> [anika_incoming]
> exten => _0049351222,hint,SIP/004935
> exten => _0049351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0049351222,n,Dial(local/222@anika_incoming)
> exten => _0351222,hint,SIP/004935
> exten => _0351222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _0351222,n,Dial(local/222@anika_incoming)
> exten => _222,hint,SIP/004935
> exten => _222,1,Verbose(2,Call for Anika - [${CALLERID(num)}])
> exten => _222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
> "+49" ]?0${CALLERID(num):3}:${CALLERID(num)})}) ; Damit das "+49" mit "0"
> ersetzt wird
> exten => _222,n,Set(CHANNEL(musicclass)=default)
> exten => _222,n,Dial(SIP/0049351222,19,RcxX)
> exten => _222,n,Verbose(2,Voicemail for Anika)
> exten => _222,n,Set(CALLERID(name)=)
> ; Damit in der E-Mail der AB nicht den Namen steht
> exten => _222,n,VoiceMail(0049351222,us)
> exten => _222,n,Hangup
>
> then I reloaded the core (core reload), SIP (sip reload) and Dialplan
> (dialplan reload) and I called the 0351222 from my mobile phone.
> It rings, but on the other phone (035) is nothing to see...
>
> Where is my error?
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
> --
> _
> -- 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
>
The hints have to be in the same contexts in extensions.conf as defines in
the sip.conf subscribecontext which can be set per peer.
Also, have you configured the phones as well?
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Signaling ringing on other extension
On 30 December 2015 at 10:19, Luca Bertoncello <lucab...@lucabert.de> wrote: > Ishfaq Malik <i...@pack-net.co.uk> schrieb: > > > The hints have to be in the same contexts in extensions.conf as defines > in > > the sip.conf subscribecontext which can be set per peer. > > Well, [anika_incoming] will be included in [default], of course... > But I tried to define anika_incoming in subscribecontext, too. No > changes... > > > Also, have you configured the phones as well? > > What do you mean? > > Thanks > Luca Bertoncello > (lucab...@lucabert.de) > > -- > _ > -- 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 > BLF is an interaction between the phones and the server. You need to configure function buttons on the phones to display the presence state of individual peers that have been set up on the server. This command in the asterisk cli will help you: core show hints If you see an entry for the peer then the server is set up correctly and if the Watchers column > 0 then you have set up the phone correctly. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Signaling ringing on other extension
On 30 December 2015 at 15:04, Luca Bertoncello <lucab...@lucabert.de> wrote: > Patrick Laimbock <patr...@laimbock.com> schrieb: > > > On 12/30/15 12:24, Luca Bertoncello wrote: > > > Ishfaq Malik <i...@pack-net.co.uk> schrieb: > > > > > >> Do you have a link to the user guide for your exact phone model? > > > > > > Unfortunately not... > > > I have a Thomson ST2022, but I can just find in Internet manual for the > > > ST2030... > > > > The administrator manual can be found at: > > http://www.manualslib.com/manual/909341/Thomson-St2020.html?page=5 > > > > To download click the green Download button at the top. > > Hi, Patrick! > > Thank you very much! > Unfortunately I didn't found anything about BLF... > > Regards > Luca Bertoncello > (lucab...@lucabert.de) > > -- > _ > -- 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 > Looks like your phones do not support it. And it is a very common feature. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Signaling ringing on other extension
On 30 December 2015 at 15:09, Luca Bertoncello <lucab...@lucabert.de> wrote: > Ishfaq Malik <i...@pack-net.co.uk> schrieb: > > > Looks like your phones do not support it. And it is a very common > feature. > > I think so... > Maybe I can write a little program running on my PC to receive a message > from > Asterisk if someone calls the other phone... > I'll think about that... > Or maybe is there already such a program running on Linux? > > Regards > Luca Bertoncello > (lucab...@lucabert.de) > > -- > _ > -- 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 > Look up fop2 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Signaling ringing on other extension
On 30 December 2015 at 15:16, Luca Bertoncello <lucab...@lucabert.de> wrote: > Ishfaq Malik <i...@pack-net.co.uk> schrieb: > > > Look up fop2 > > Thank you very much, but I prefer a standalone application, if it's > possibile... > Any other suggestion? > > Thanks > Luca Bertoncello > (lucab...@lucabert.de) > > -- > _ > -- 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 > If you can programme, create an application that logs into the asterisk box via the AMI, read the event stream and produce an alert which it sees a phone ringing. https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4817239 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] No sound with internal calls depending on which phones
68.128.231:51350 (type 03, len 33) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33) > Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33) > == Spawn extension (local, 301, 1) exited non-zero on > 'SIP/hsolutionspf5-0002' > > I tried many options to disable SRTP but without success : > >- canreinvite = no >- canreinvite = nonat >- srtpcapable=no >- encryption=no >- directmedia=nonat >- ...or noload => res_srtp.so in modules.conf > > > Any help would be GREATLY appreciated ! > > Denis > > P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final) > > > -- > _ > -- Bandwidth and Colocation Provided by <http://www.api-digital.com> > 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 > -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 encrypted authentication for clients
On 28 October 2015 at 22:54, Motty <motty.c...@gmail.com> wrote: > Hello, > I am searching for a solution to encrypt authentication from Asterisk > server to clients. Searching srtp seem to encrypt traffic, I just want > client authentication with encryption. Can someone point to the right > direction? has anybody used ZRTP? experience with ZRTP? > > Thanks, > _motty > > -- > _ > -- 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 > https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] 2 asterisk instances sharing 1 astDB
Hi We are using 1.8 on CentOS 6 We use asterisk servers in pairs for machine level failover. On a recent pair we pointed the astdb location of both nodes of the pair to the same location on a shared storage device. Now it would appear that if the asterisk service is restarted, and queue members added via the AMI are forgotten. Is there any issues in trying to share a single astdb over 2 machines that we are unaware of? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 set the global setting for each pjsip endpoint
On 22 September 2015 at 16:04, Thyda ENG <ength...@gmail.com> wrote: > I have many endpoints and each endpoint has some parameter in common so i > wonder is there any way to config one for all endpoints? Like in my example > I have two endpoints and I repeat the same thing, > > [100] > > type=endpoint > > aors=100 > > auth=100-auth > > allow=ulaw,alaw,gsm,g726 > > context=from-internal > > callerid=device <100> > > dtmf_mode=rfc4733 > > use_avpf=no > > ice_support=no > > media_use_received_transport=no > > trust_id_inbound=yes > > send_pai=yes > > rtp_symmetric=yes > > rewrite_contact=yes > > message_context=astsms > > > [200] > > type=endpoint > > aors=200 > > auth=200-auth > > allow=ulaw,alaw,gsm,g726 > > context=from-internal > > callerid=device <200> > > dtmf_mode=rfc4733 > > use_avpf=no > > ice_support=no > > media_use_received_transport=no > > trust_id_inbound=yes > > send_pai=yes > > rtp_symmetric=yes > > rewrite_contact=yes > > message_context=astsms > > > how could I avoid duplicate thing like this ? > > -- > > >From my brief look at pjsip.conf it uses the same template concept as the sip.conf. Here's the relevant instructions from the sip.conf in asteris13 ; ; Because you might have a large number of similar sections, it is generally ; convenient to use templates for the common parameters, and add them ; the the various sections. Examples are below, and we can even leave ; the templates uncommented as they will not harm: [basic-options](!); a template dtmfmode=rfc2833 context=from-office type=friend [natted-phone](!,basic-options) ; another template inheriting basic-options directmedia=no host=dynamic [public-phone](!,basic-options) ; another template inheriting basic-options directmedia=yes [my-codecs](!); a template for my preferred codecs disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw ; Or, more simply: ;allow=!all,ilbc,g729,gsm,g723,ulaw [ulaw-phone](!) ; and another one for ulaw-only disallow=all allow=ulaw ; Again, more simply: ;allow=!all,ulaw ; and finally instantiate a few phones ; ; [2133](natted-phone,my-codecs) ;secret = peekaboo ; [2134](natted-phone,ulaw-phone) ;secret = not_very_secret ; [2136](public-phone,ulaw-phone) ;secret = not_very_secret_either ; ... ; Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Call waiting for Queue Agents.
On 21 September 2015 at 15:27, Aziz TestAccount <azizgst...@gmail.com> wrote: > Hi All, > > I have a question about the Queues. > > I'm using Asterisk 11.13.0 , and I want to configure the following setup : > > When there is an incoming call to the queue all agents should ring even > those that are already in call, they should receive a second call. > > Is this doable in any Asterisk version ? > > Thanks in advance. > > > In 1.8 there is a ring in use option at the queue level. I doubt this will have been removed in 11. ; If you want the queue to avoid sending calls to members whose devices are ; known to be 'in use' (via the channel driver supporting that device state) ; uncomment this option. (Note: only the SIP channel driver currently is able ; to report 'in use'.) ; ; ringinuse = no Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 return a transfered call to the transferrer?
On 15 July 2015 at 20:51, Ethy H. Brito ethy.br...@inexo.com.br wrote: Hi all Any of you guys could point me in the right direction? I need to make that a blind transfer to return to the transferrer when the transferee does not answer. Scenario: . Miss Jane Doe, our front desk attendant, picks up an external call to Mr. Smith; . Miss Doe flashes, dial Mr. Smith's extension and then hangup; . Mr Smith's phone rings until timeout; . At this point, how to return the call to the Miss Doe's extension; Cheers Ethy -- _ Do a channel dump on the transferred channel, you'll see marker channel variables showing it's a transfer and that contain the sending peer name. You can use dialplan logic to check if it's a transfer. If it is, you can send the call back to the referrer peer. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 how to setup alarm too many outgoing calls from same user
On 6 July 2015 at 15:27, Motty Cruz motty.c...@gmail.com wrote: Hello, I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts. can someone help? Thanks, You could use fail2ban for this by adding your own filter string specific for that user. It would have the advantage of blocking further calls as well as alerting you by email. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Question on permit/deny
On 1 July 2015 at 04:03, Jerry Geis ge...@pagestation.com wrote: I see in my log file this: Jun 30 21:44:26] NOTICE[42192][C-02f3] chan_sip.c: Call from '' ( 5.189.144.120:5076) to extension '011972592675431' rejected because extension not found in context 'default'. which is great its rejected - however in my sip.conf file I have deny=0.0.0.0 permit=x.y.z.z/255.255.255.255 permit=a.b.c.d/255.255.255.255 So I'm expecting to deny everything and only allow the two addresses I have listed of which the 5.189.144.120 is not one of? What is wrong with my permit/deny ? Thanks, Jerry -- _ Check your sip.conf to see if allowguest is explicitly set to no. ;context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) ; If your Asterisk is connected to the Internet ; and you have allowguest=yes ; you want to check which services you offer everyone ; out there, by enabling them in the default context (see below). Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Branch based on call volume
On 27 June 2015 at 21:34, Michelle Dupuis mdup...@ocg.ca wrote: Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 You could do a core show channels and grep it for the peer name. Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Variable variables
Hi
Can asterisk handle asterisk variable variables?
For example:
If I were to set
FOO300=BAR111
and I had something in a dialplan like:
_3XX,1,NoOp(${FOO${EXTEN}})
And the user had entered 300, it would output BAR111
We are using asterisk 1.8
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Manipulate extension state in 1.8.x
Hi Is there any way to set the presence state of a peer to in-use in asterisk 1.8? The idea is to integrate DND buttons on phones to BLF. Regards -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manipulate extension state in 1.8.x
Hi John
I needed a dialplan solution so thank you very much for the pointer!
Regards
Ish
On 9 June 2015 at 17:27, John Kiniston johnkinis...@gmail.com wrote:
You can use a custom device state to do it.
[dnd]
;DND Toggle
exten = *363,1,Answer()
same =
n,Set(CURRENT_PRESENCE=${DEVICE_STATE(Custom:DND${CHANNEL(peername)})})
same = n,GotoIf($[${CURRENT_PRESENCE}=NOT_INUSE]?*78,1:*79,1)
;DND On
exten = *78,1,NoOP(Turning DND On)
same = n,Set(DEVICE_STATE(Custom:DND${CHANNEL(peername)})=BUSY)
same = n,Playback(do-not-disturbenabled)
same = n,Hangup()
;DND Off
exten = *79,1,NoOP(Turning DND Off)
same = n,Set(DEVICE_STATE(Custom:DND${CHANNEL(peername)})=NOT_INUSE)
same = n,Playback(do-not-disturbdisabled)
same = n,Hangup()
Then you can simply hint on your device like:
exten = _70X,hint,SIP/${EXTEN}Custom:DND${EXTEN}
On Tue, Jun 9, 2015 at 9:19 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there any way to set the presence state of a peer to in-use in
asterisk 1.8?
The idea is to integrate DND buttons on phones to BLF.
Regards
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
--
_
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--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
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[asterisk-users] Asterisk 1.8.32.3 chan_sip deadlock
Hello asterisk-users,
We've been having intermittent issues with chan_sip - it stops responding
to cli requests, trying to reload chan_sip from cli doesn't seem to have
any effect, initiated calls carry on for a short period, but no new SIP
requests are processed ('sip show channels' hangs forever, server stops
responding to SIP OPTIONS, or any other SIP messages). We have updated the
build from 1.8.23.1 to the latest asterisk 1.8 (1.8.32.3), however the
problem still persists. We have gathered debugging information from 'core
show locks' and from gdb, attached to this message (with phone numbers and
extension and context names obscured). We are running realtime under CentOS
6.6, built from source and packaged using rpmbuild, with the following
menuselect options (debugging version):
menuselect/menuselect --disable BUILD_NATIVE --enable DEBUG_THREADS
--enable DONT_OPTIMIZE --disable CORE-SOUNDS-EN-GSM --disable-category
MENUSELECT_EXTRA_SOUNDS --disable MOH-OPSOUND-WAV --enable-category
MENUSELECT_ADDONS --disable format_mp3 --disable cdr_tds --disable cel_tds
--disable cdr_pgsql --disable cel_pgsql --disable res_config_pgsql
menuselect.makeopts
under kernel 2.6.32-504.el6.x86_64, and linked against the following
library versions:
/usr/lib64/libssl.so.10:symbolic link to `libssl.so.1.0.1e'
/usr/lib64/libcrypto.so.10: symbolic link to `libcrypto.so.1.0.1e'
/lib64/libc.so.6: symbolic link to `libc-2.12.so'
/usr/lib64/libxml2.so.2:symbolic link to `libxml2.so.2.7.6'
/lib64/libz.so.1: symbolic link to `libz.so.1.2.3'
/lib64/libm.so.6: symbolic link to `libm-2.12.so'
/lib64/libdl.so.2: symbolic link to `libdl-2.12.so'
/lib64/libpthread.so.0: symbolic link to `libpthread-2.12.so'
/lib64/libtinfo.so.5: symbolic link to `libtinfo.so.5.7'
/lib64/libresolv.so.2: symbolic link to `libresolv-2.12.so'
/lib64/libgssapi_krb5.so.2: symbolic link to `libgssapi_krb5.so.2.2'
/lib64/libkrb5.so.3:symbolic link to `libkrb5.so.3.3'
/lib64/libcom_err.so.2: symbolic link to `libcom_err.so.2.1'
/lib64/libk5crypto.so.3:symbolic link to `libk5crypto.so.3.1'
/lib64/libkrb5support.so.0: symbolic link to `libkrb5support.so.0.1'
/lib64/libkeyutils.so.1:symbolic link to `libkeyutils.so.1.3'
We'd appreciate any possible assistance, as we're having problems working
out what exactly triggers the deadlock and we have not been able to find
the correct sequence of steps to reproduce the issue yet, other than
waiting for it to lock up at an arbitrary time with the debugging code in
place. It does seem to happen at least once a day, however.
What is the best way of getting the core show locks output for people to
see as it appears to be too big to mail?
Ish
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Re: [asterisk-users] ringing in queues
On 13 March 2015 at 14:04, Matt Hamilton efes9...@hotmail.com wrote: We use the ringall strategy for a small queue with 4 members. When a call comes in, if one of the members is busy, all the phones except the busy phone rings (as intended). While the other phones are ringing, if this busy phone becomes available again, we would like to have it start ringing. Right now it just sits idle. Is this possible? I played with ringinuse (queues.conf) and callcounter (sip.conf) values, but wasn't able to get it going. Thanks, Matt -- _ -- 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 Reduce the timeout in the queue configuration (but not in the Queue application in the dialplan), when the timeout (and the retry) value has elapsed, all available members will be rung again. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 perform some tasks after the CDR has been closed?
On 26 February 2015 at 11:57, Daniel Gonzalez gonva...@gonvaled.com wrote: Hi, I would like to do some tasks after the CDR has been closed, and the CDR(end), CDR(billsec) and CDR(duration) fields are available. I have tried to do that on the h extension, but it seems the CDR is not yet complete in the h extension. When is the CDR closed? How can I trigger some actions after that event? It would be nice if the channel is still available, since I need access to other channel variables. An alternative would be to pass those variables via the CDR if the channel has been deleted. Thanks, Daniel Gonzalez -- _ Have you set endbeforehexten=yes in your cdr.conf ? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] SIP Jitterbuffer
Hello people What are the cons, if any, of enabling a jitterbuffer? We are currently using version 1.8 Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 11.6. SIP realtime lost peers after 'sip reload'
On 16 February 2015 at 11:49, Igor Pavlov i...@izhnet.ru wrote: Hi, list. We have a problem with loss peers after ‘sip reload’, our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do ‘sip reload’ , peers are removing from available. *Before `sip reload` :* srv-pbx2*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Description Realtime 303411/303411 172.16.1.12 D 5060 OK (77 ms) Cached RT 467577/467577 172.16.1.22 D 5060 OK (141 ms) Cached RT 561871/561871 172.16.1.32 D 5060 OK (7 ms) Cached RT sip-proxy2 172.16.1.2 5061 OK (1 ms) srv-pbx-in 172.16.1.7 5060 OK (1 ms) *After `sip reload`:* [Feb 16 14:30:20] DEBUG[1468]: res_config_mysql.c:497 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipusers WHERE name LIKE '%' AND callbackextension LIKE '%' ORDER BY name [Feb 16 14:30:20] DEBUG[1468]: config.c:1650 config_text_file_load: Parsing /etc/asterisk/sip_notify.conf == Parsing '/etc/asterisk/sip_notify.conf': Found [Feb 16 14:30:20] DEBUG[1468]: chan_sip.c:32383 reload_config: SIP reload_config done...Runtime= 0 sec [Feb 16 14:30:20] DEBUG[1468]: sched.c:546 ast_sched_dump: Asterisk Schedule Dump (12 in Q, 623646 Total, 30 Cache, 42 high-water) [Feb 16 14:30:20] DEBUG[1468]: sched.c:551 ast_sched_dump: = [Feb 16 14:30:20] DEBUG[1468]: sched.c:552 ast_sched_dump: |ID Callback Data Time (sec:ms) | [Feb 16 14:30:20] DEBUG[1468]: sched.c:553 ast_sched_dump: +-+-+-+-+ [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623634 | 0x7f2ebc5415d0 | 0x7f2ea0b95b68 | 01 : 434169 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623628 | 0x7f2ebc5451c0 | 0x7f2ea0bc5148 | 04 : 912209 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623639 | 0x7f2ebc5415d0 | 0x7f2ea08a0158 | 21 : 585476 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623635 | 0x7f2ebc5415d0 | 0x7f2ea0b6bc98 | 11 : 452094 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623632 | 0x7f2ebc5451c0 | 0x7f2ea0b9b388 | 17 : 091999 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623643 | 0x7f2ebc5451c0 | 0x2d473d8 | 55 : 803782 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623511 | 0x7f2ebc527410 | 0x7f2ea0b9b388 | 000266 : 237816 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623640 | 0x7f2ebc5415d0 | 0x7f2ea0baf088 | 22 : 472571 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623541 | 0x7f2ebc527410 | 0x7f2ea0affa28 | 000650 : 207449 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623600 | 0x7f2ebc527410 | 0x7f2ea0bc5148 | 000794 : 895787 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623638 | 0x7f2ebc5451c0 | 0x7f2ea0affa28 | 40 : 622455 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:565 ast_sched_dump: |623646 | 0x7f2ebc5451c0 | 0x2d4cee8 | 55 : 902262 | [Feb 16 14:30:20] DEBUG[1468]: sched.c:568 ast_sched_dump: = [Feb 16 14:30:20] DEBUG[1468]: chan_sip.c:33170 sip_do_reload: --- Done destroying pruned peers [Feb 16 14:30:20] DEBUG[1468]: chan_sip.c:33185 sip_do_reload: do_reload finished. peer poke/prune reg contact time = 0 sec. [Feb 16 14:30:20] DEBUG[1468]: chan_sip.c:33187 sip_do_reload: --- SIP reload done srv-pbx2*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Description Realtime sip-proxy2 172.16.1.2 5061 OK (1 ms) srv-pbx-in 172.16.1.7 5060 OK (1 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] *Is this normal behavior of SIP realtime ?* - Best regards, Igor Pavlov This will always happen. When using ARA, peers will only go into the realtime cache when one tries to register or be dialled. At that point the settings will be taken from the DB and put into the realtime cache. A SIP reload will clear the realtime cache. One way to mitigate this effect to use 'sip show peer peername load'. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED
Re: [asterisk-users] IAX port
On 10 February 2015 at 09:02, jg webaccounts...@jgoettgens.de wrote: I get an occasional similar problem, we have Mikrotik firewalls and from tcpdump monitoring on the asterisk boxes I can see that the firewall (unbidden) has changed the IAX port. Usually a firewall reset and sometimes PBX reset combination fixes it. Its odd as its only one direction, occurs rarely and with no obvious driver. So IAX is happy in one direction but not the other. And I can see packets in the unhappy point arriving on the wrong port. I couldn't fix it without kicking the router/firewall so I would say its a router problem in the Destination NAT process. Cheers Duncan Port is changed when NAT is applied from LAN to WAN. While UDP session is maintained as ESTABLISHED, that port should not change. If your peer changes constantly of session port could be UDP session is too short in NAT table on routers. You can try setting qualify=1000 (which is in ms. Default is 2000), and see if peer keeps same port. Regards. voip-info.org also has an entry about general NAT related issues, which could be relevant here I do not seem to have problems with Netgear firewalls, but other firewalls show this effect. So far it happened only on a single side, such that calls work from the other side. I already checked the open ports with nc/ncat/netcat as UDP sender and receiver on the other end. The ports are open, even when the arbitrary ports are used by Asterisk. I'll need to read a bit more and evaluate my pcap traces and possibly ask the router vendors. Thank you for your efforts. jg Some firewalls have a 'consistent NAT' option that needs to be enabled, otherwise you get the symptoms described. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] IAX port
On 10 February 2015 at 12:55, jg webaccounts...@jgoettgens.de wrote: Some firewalls have a 'consistent NAT' option that needs to be enabled, otherwise you get the symptoms described. While reading about NAT, I came across this web site: http://nattest.net.in.tum.de/ The test tool looks at various NAT related properties and prints the results related to TCP/UDP binding properties, TCP/UDP hole punching, etc. In my case a very short value was reported for the UDP timeout, such that depending on the sequence of packets, the entry in the mapping table might already have been deleted. This could explain the random nature of my connection problem. Port predictability does not seem to be a problem. Does that make any sense? jg Yes UDP timeout being too short is another thing I've experience with firewalls (admittedly limited and once removed experience). Actually, this one can be a (mild) problem on Draytek routers and can be resolved by telnetting into the router and using the portmaptime command. Also, turn of stateful packet inspection if it is an option. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Question regarding custom announcements used by several Asterisk servers
On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? Keep up the good work! cheers, Olli -- _ -- 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 Hi All of that is possible and is exactly what we do, both for customer sounds and for call recordings. Just make sure you have resilience in your shared storage device. Alternatively, you could use something like Puppet to deploy the files to all the servers. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] queue show queue-name vs queue log for calculating average hold time
Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show queue_name I get the following numbers: queue_name has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the average hold time before calls get answered by a queue members. However, if I calculate the average hold time from out queue log table using the following SQL select sum(data1)/ count(*) as ave_hold_time from queue_log where time DATE(NOW()) and queuename='queue_name' and event='CONNECT'; I get the vastly different figure of 92.4. So, is the queue show figure wrong due to a bug or am I making an incorrect assumption as to what it means? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] CALLERID(ani2) inserting
Hi
According to this:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables
It is read only.
On 22 January 2015 at 16:22, CDR vene...@gmail.com wrote:
I checked
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
But I cannot find a way to insert CALLERID(ani2), which I can read, but
when I try to set it for a new call, I get a runtime error.
This information, known as isup-oli comes embedded in the From header,like
this
sip:9087311878@64.45.157.104:5060;isup-oli=62;tag=sansay1724414rdb124
and it can be read by using
Set(var=${CALLERID(ani2)}
But how do we add that information to the outbound INVITE? This is
critical in the toll-free industry and call-from-jail industries.
Thanks for your help.
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37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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[asterisk-users] queue reload command
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] queue reload command
That's what I would have guessed but it's not working:
[ish@??? ~]$ asterisk -rx 'queue show axon-all'
axon-all has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:2, SL:0.0% within 20s
Members:
AXON200 (realtime) (Not in use) has taken no calls yet
AXON201 (realtime) (Not in use) has taken no calls yet
AXON202 (realtime) (Not in use) has taken no calls yet
AXON203 (realtime) (Not in use) has taken no calls yet
AXON204 (realtime) (In use) has taken no calls yet
AXON205 (realtime) (Not in use) has taken no calls yet
AXON206 (realtime) (Not in use) has taken no calls yet
AXON207 (realtime) (Not in use) has taken no calls yet
AXON208 (realtime) (Unavailable) has taken no calls yet
AXON209 (realtime) (Not in use) has taken no calls yet
AXON210 (realtime) (Unavailable) has taken no calls yet
AXON211 (realtime) (Unavailable) has taken no calls yet
AXON214 (realtime) (Not in use) has taken no calls yet
AXON221 (realtime) (Not in use) has taken no calls yet
AXON222 (realtime) (Not in use) has taken no calls yet
AXON223 (realtime) (Unavailable) has taken no calls yet
AXON225 (realtime) (Not in use) has taken no calls yet
AXON231 (realtime) (Unavailable) has taken no calls yet
AXON232 (realtime) (Not in use) has taken no calls yet
AXON233 (realtime) (Not in use) has taken no calls yet
No Callers
[ish@??? ~]$ asterisk -rx 'queue reload axon-all'
No such command 'queue reload axon-all' (type 'core show help queue reload
axon-all' for other possible commands)
On 8 January 2015 at 14:23, Andrew Colin and...@convergedgroup.net wrote:
Hi
queue reload(queue name) or queue reload all
for example
queue reload reception
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ishfaq Malik
*Sent:* Thursday, January 8, 2015 2:10 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] queue reload command
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] odbc connection timeout varable
On 11 November 2014 15:27, Tech Support aster...@voipbusiness.us wrote: Unless of course the database server is not running at all for some reason. Regards; JVC Surely that should be monitored by some system designed for that purpose such as Nagios? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] queue log realtime mysql
On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores the data in 4 columns : 'data1' -- 'data 5'. All other servers store data in 1 column 'data' with the data seperated by pipe. I see no difference in my configuration of extconfig.conf and logger.conf. Maybe a hidden default value ? Can someone tell me which setting makes the mysql realtime driver store data in 1 column or in seperate columns ? Using Asterisk 1.8.12.2 Kind regards, Jonas. Are you using mysql_realtime or odbc with a mysql back end? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Call forwarding from Phones and getting the referrer IP
On 24 October 2014 16:51, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8 but I'm sure this applies to other versions. If someone puts a call divert on a handset such as a Snom phone I get this type of SIP message on receipt of an inbound call: Got SIP response 302 Moved Temporarily back from xxx.xxx.xxx.xxx:x Which then triggers a local channel to make the call. Is there any way I can access that IP address inside my dialplan? I've done a ChanDump and there's no sign of it. Regards Ish Bumping this as I originally sent it late on Friday. If anyone has any idea, please let me know. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Call forwarding from Phones and getting the referrer IP
Hi I'm using asterisk 1.8 but I'm sure this applies to other versions. If someone puts a call divert on a handset such as a Snom phone I get this type of SIP message on receipt of an inbound call: Got SIP response 302 Moved Temporarily back from xxx.xxx.xxx.xxx:x Which then triggers a local channel to make the call. Is there any way I can access that IP address inside my dialplan? I've done a ChanDump and there's no sign of it. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] SIPAddHeader from a realtime databse
On 23 September 2014 15:04, Rusty Newton rnew...@digium.com wrote: On Mon, Sep 22, 2014 at 9:43 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Guys I'm using asterisk 1.8.23.1 When I add a SIP Header from inside the extensions.conf (SIPAddHeader(Alert-Info:http://www.notused.com \;info=alert-internal\;x-line-id=0) ) it works fine. When I try to do the same thing from within a database table, all of the string apart from x-line-id=0 gets ignored. I've tried escaping the semicolon and not escaping it and the result is always the same, just the last part of the full string is expressed. Some of the ways that I have tried to enter the string are below: appdata='Alert-Info:http://www.notused.com \\;info=alert-internal\\;x-line-id=0' appdata='Alert-Info:http://www.notused.com ;info=alert-internal;x-line-id=0' appdata='Alert-Info:http://www.notused.com ;info=alert-internal;x-line-id=0' Does anyone know the correct format to store this in a DB table for it to be expressed correctly? I'm using MySQL. There is an existing report filed here: https://issues.asterisk.org/jira/browse/ASTERISK-19254 You can try Walter's suggestion on the issue and report back whether it works or not. Hi Replacing the ; with ^3B and removing the \ so column data looks like: Alert-Info:http://www.notused.com^3Binfo=alert-internal^3Bx-line-id=0 works perfectly. Thanks for the help. Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] SIPAddHeader from a realtime databse
Hi Guys I'm using asterisk 1.8.23.1 When I add a SIP Header from inside the extensions.conf (SIPAddHeader(Alert-Info:http://www.notused.com\;info=alert-internal\;x-line-id=0) ) it works fine. When I try to do the same thing from within a database table, all of the string apart from x-line-id=0 gets ignored. I've tried escaping the semicolon and not escaping it and the result is always the same, just the last part of the full string is expressed. Some of the ways that I have tried to enter the string are below: appdata='Alert-Info:http://www.notused.com \\;info=alert-internal\\;x-line-id=0' appdata='Alert-Info:http://www.notused.com ;info=alert-internal;x-line-id=0' appdata='Alert-Info:http://www.notused.com ;info=alert-internal;x-line-id=0' Does anyone know the correct format to store this in a DB table for it to be expressed correctly? I'm using MySQL. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] (no subject)
If you're using a redhat based distro, have you checked SELinux? Try disabling (will require a server reboot) Regards Ish On 3 September 2014 20:41, Steve Edwards asterisk@sedwards.com wrote: For future reference, a well chosen subject will yield more relevant replies. Better bait == better fish. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] features.conf and mixmonitor stop and start
On 28 August 2014 07:56, Leandro Dardini ldard...@gmail.com wrote:
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik i...@pack-net.co.uk:
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename into a variable and give it full
inheritance so you can resume recording to the same file (using the a
option)
On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have a recording started in the dialplan with the MixMonitor
application. I want to be able to stop it during a call and maybe restart
it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is
running?
[featuremap]
automixmon = *1
I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:
= {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
}
[applicationmap]
pauseMonitor = #1,self/both,stopMixMonitor
Any advice?
extensions.conf:
[macro-pause-recording]
exten = s,1,Verbose(Stopping Recording)
exten = s,n,StopMixMonitor()
[macro-unpause-recording]
exten = s,1,Verbose(Resuming Recording)
exten = s,n,MixMonitor(${REC_FILE_NAME},a)
features.conf
StopMixMonitor = #00,peer/both,Macro(pause-recording)
;
MixMonitor = #01,peer/both,Macro(unpause-recording)
Make sure you set REC_FILE_NAME early on with a double underscore and
remember to add Set(__DYNAMIC_FEATURES=MixMonitor#StopMixMonitor) early on
too
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
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Re: [asterisk-users] features.conf and mixmonitor stop and start
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename into a variable and give it full
inheritance so you can resume recording to the same file (using the a
option)
On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have a recording started in the dialplan with the MixMonitor
application. I want to be able to stop it during a call and maybe restart
it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is running?
[featuremap]
automixmon = *1
I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:
= {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
}
[applicationmap]
pauseMonitor = #1,self/both,stopMixMonitor
Any advice?
--
_
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--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
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Re: [asterisk-users] Asterisk on CentOS7
On 13 August 2014 17:51, Paul Greenberg p...@greenberg.pro wrote: Hi Matthew, I am using it. Works like a charm! Running it for 3 week already and have no issues. However, my system is not heavily utilized, i.e. 50-150 phone calls a day. The only thing is I was not able to get asterisk integrated with CentOS services daemon. So, I am starting asterisk manually. Best Regards, Paul Greenberg, Esq. Law Office of Paul Greenberg 530 Main Street, Suite 102 Fort Lee, NJ 07024 E-mail: p...@greenberg.pro Tel: 201-402-6777 Fax: 201-301-8876 Web: http://www.greenberg.pro From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Matthew Jordan mjor...@digium.com Sent: Wednesday, August 13, 2014 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on CentOS7 On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Is anyone using asterisk on CentOS 7? If so, is it working fine and as expected? Random data point: the Asterisk project's build agents are still on CentOS 6. Your mileage may vary. Thanks for the feedback. I think I've heard enough not to leapfrog from 5 to 7 and to go to 6 instead. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Asterisk on CentOS7
Hi Is anyone using asterisk on CentOS 7? If so, is it working fine and as expected? Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] The plain old PBX functionality
On 7 August 2014 21:06, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: back in the old analog telephony days there was digital PBX-es and digital system phonesets. This phonesets have had many individual illuminatable buttons connected with extensions. The PBX can show on the buttons if some extension is ringing (blinks) or busy (constant light), and the user can transfer the call with one touch (pressing one of this button). I search this functionality in Asterisk. What versions, and what extension functions (or other settings), and what VoIP phones can do this? Asterisk has had this functionality for a long time. The terms you want to search for are BLF (Busy Lamp Field) and Subscribe. I imagine that most sip phones have the necessary features to do BLF. I know the Polycom phones I use certainly do. The Digium branded phones do as well. Also certain models of Snom and Yealink phones. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 configuration in realtime static and realtime dynamic
Not that I know of but since you are using a database you can update multiple rows at once. Please note, if you change the settings in the database entry of a peer that is currently connected, you will need to flush the realtime cache with the following command sip prune realtime peer-name The next time the endpoint registers it will pick up the new configuration. On 25 July 2014 12:38, Robin Kipp mli...@robin-kipp.net wrote: Hi Ishfaq, Am 24.07.2014 um 09:57 schrieb Ishfaq Malik i...@pack-net.co.uk: It supplements it. In fact, you can define some peers in the sip.conf and some in the MySQL table. However, if you do add any in the sip.conf directly, you'll have to do a sip reload which will clear your realtime cache. Thanks a lot for the information! Makes a lot more sense to me now :-) What about templates though, is there any way of doing that? For example, defining templates in sip.conf and then referencing them in the MySQL database… Thanks! Robin -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 configuration in realtime static and realtime dynamic
On 23 July 2014 21:53, Robin Kipp mli...@robin-kipp.net wrote: Hi all, I’m currently in the process of familiarizing myself with Asterisk, and am trying to move certain configuration objects (such as SIP peers) into a MySQL database, accessed by Asterisk using the ODBC connector. Now, I’ve imported the sippeers MySQL table from the contrib directory of the Asterisk source, and I could add SIP users in here. However, I currently don’t understand whether this realtime dynamic configuration table is meant to replace or just supplement sip.conf. This is because the sippeers table does not offer certain fields for entries in the [general] section of my sip.conf file, such as the ‚udpbindaddr‘ variable. So, am I supposed to put all that in the database by adding appropriate table columns, or can I leave this in the sip.conf file and chan_sip.so will read both the file and MySQL table once loaded? Also, is there anyway that I could use templates, so that I don’t have to redefine everything for each SIP peer? Thanks a lot for help! Robin Hi It supplements it. In fact, you can define some peers in the sip.conf and some in the MySQL table. However, if you do add any in the sip.conf directly, you'll have to do a sip reload which will clear your realtime cache. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] TLS, STRP and ARA
Hi I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP. However, we exclusively use the asterisk realtime architecture using the mysql connector. Looking at tutorials we have to set encryption=yes and transport=tls for any peer we want encrypted traffic for. Having a look at contrib/realtime/mysql/sippeers.sql from the source code shows that the encryption column is completely absent and tls is not an option for transport. Does this mean I can't configure a peer to use TLS and SRTP if using ARA? Are there any workarounds? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] TLS, STRP and ARA
I have just answered my own questions and it's all fine.
transport will accept a value of tls and interpret it (you'll have to alter
the column definition if you're using an enum).
encryption column can be added and interpreted, here's the column defintion
I used.
alter table sip add column encryption enum ('yes','no') default 'no';
On 21 July 2014 11:31, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP.
However, we exclusively use the asterisk realtime architecture using the
mysql connector.
Looking at tutorials we have to set encryption=yes and transport=tls for
any peer we want encrypted traffic for.
Having a look at contrib/realtime/mysql/sippeers.sql from the source code
shows that the encryption column is completely absent and tls is not an
option for transport.
Does this mean I can't configure a peer to use TLS and SRTP if using ARA?
Are there any workarounds?
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
--
_
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Re: [asterisk-users] packet2packet bridging
://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 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 monitor non-SNMP SIP devices ?
On 9 July 2014 16:19, Olivier oza.4...@gmail.com wrote: Hi, I'm seeing a trend in which SIP devices such as Yealink SIP phones (with v72 firmware), are dropping support of SNMP in favor of HTTP eventing if may call this as such : when configuring the SIP device, you can define a couple of HTTP URL which triggered when some event occur (end of boot, on hook, ...). How do deal with those devices ? Do you still try to monitor them with usual tools (Nagios, OpenNMS) or do you favor another class of software ? Regards If you set qualify on your peers you could monitor the event stream of the AMI which would show you any end point going unreachable. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] CDR dcontext not updated on FAILED and BUSY calls
Hi We're using asterisk 1.8.23.1. Our inbound calls are routed into the default context with explicit number matching. If found they are passed on to a distinct context for the number being called using the Goto application. If the call is successful or even if it has no answer, the cdr dcontext field has the correct second context. However, if the call fails or is busy, and even though we can see it is executing a step in the second context as show in the cdr lastdata field, the dcontext still shows as default. Is this a bug or expected behaviour? If it is expected behaviour, what is the reasoning behind it? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Changing recorded file storage directory.
On 26 June 2014 15:42, Anurag Rana anuragrana31...@gmail.com wrote: Hi All, In asterisk, default directory to store the call-recording files is /var/spool/asterisk/monitor. Can we change this directory? How? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. Hi You can specify the full path when doing the Monitor or MixMonitor application. You can change the spool directory in your asterisk.conf but this will move all the directories that normally live under /var/spool/asterisk Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 realtime peer registration
On 10 June 2014 05:27, ortei...@tiscali.it wrote: Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers = odbc,asterisk,sipclient sipusers = odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (https://wiki.asterisk.org/wiki/display/ ... +structure https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure ) The user registered on asterisk works fine, but not the peer. I'd like to use my voipdiscount account as a peer to do external call. Name/username Host Dyn Forcerport ACL Port Status Realtime 2000/2000 xxx.xxx.xxx.xxx D N 65476 OK (117 ms) Cached RT Mysql entry on sipclient table is below: 3 sip.voipdiscount.com 5060 \N XX \N \N \N \N sip.voipdiscount.com peer default \N \N XXX \N \N rfc2833 yes no \N \N \N \N \N port,invite \N \N \N \N \N \N 01234556678 \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N sip.voipdiscount.com X yes \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N XX \N voipdiscount_out \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N I enabled also sip debug, but I don't see any attempt towards sip.voipaccount.com What am I doing wrong? Someone can help me? Thanks in advance Pietro Try changing the type from peer to friend. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Mixing res_mysql and res_odbc
Hi Is there any harm in using res_mysql for some things and res_odbc for others? We already use res_mysql for ARA but could do with having CEL logged to MySQL. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] R: Mixing res_mysql and res_odbc
Hi Pietro That wasn't a response to you but a genuine question for myself out to the users list! Regards Ish On 10 June 2014 13:13, ortei...@tiscali.it wrote: Ok Ish, I will try with res_mysql. Still thanks *Da:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Per conto di *Ishfaq Malik *Inviato:* martedì 10 giugno 2014 12:05 *A:* Asterisk Users Mailing List - Non-Commercial Discussion *Oggetto:* [asterisk-users] Mixing res_mysql and res_odbc Hi Is there any harm in using res_mysql for some things and res_odbc for others? We already use res_mysql for ARA but could do with having CEL logged to MySQL. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Queue is not working
On 22 May 2014 12:42, omakhileshchand omakhileshch...@gmail.com wrote: Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come that call goes to Line 2(same extension used by Line 1) and 3rd call goes to 3rd line and so on. But i want to whenever 2nd call will come that call goes into different extentsion that call never hit into reserved extention. extenstion.conf [Queue_Test] exten = s,1,Answer ; Important, see notes exten = s,2,Queue(Queue_Test|tT|||300) ;dont set n option until really needed exten = s,3,Hangup() queues.conf [Queue_Test] music = default strategy = fewestcalls context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member = Sip/4001 member = Sip/4003 member = Sip/4004 member = Sip/4005 member = Sip/4006 member = Sip/4007 Regards Akhilesh In your sip.conf have you got callcounter = yes set? What stats is queue show Queue_Test showing at various times? (this will give you an indication of how many calls each member has taken) What happens when you choose rrmemory as the stratergy? Have you read and fully understood the joinempty parameter? Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Voicemail message to text
HI there I was wondering if anyone has implemented voicemail to text and if so, what package is being used to do so? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 1.8 and calendar intergration
On 15 May 2014 16:04, Ishfaq Malik i...@pack-net.co.uk wrote: On 15 May 2014 16:03, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8.25.0 on CentOS 6. I have compiled it with all the calendar modules: *CLI module show like calendar Module Description Use Count res_calendar.soAsterisk Calendar integration4 res_calendar_ews.soAsterisk MS Exchange Web Service Calenda 0 res_calendar_caldav.so Asterisk CalDAV Calendar Integration 0 res_calendar_exchange.so Asterisk MS Exchange Calendar Integratio 0 res_calendar_icalendar.so Asterisk iCalendar .ics file integration 0 5 modules loaded I'm trying to integrate this with a new calendar I've created on an existing Google account but can't get it to work. I've tried ical with the ical url from the calendar settings and I've tried caldav using https://www.google.com/calendar/dav/my-cal-id/events/ as the url but it just won't work. I've added events in the next couple of hours with reminders on the calendar that I'm referencing but the asterisk just wont pick up the events: *CLI calendar show calendar ishcal Name : ishcal Notify channel: SIP/a-sip-peer Notify context: Notify extension : Notify applicatio : Playback Notify appdata: tt-weasels Refresh time : 1 Timeframe : 3600 Autoreminder : 10 Events -- *CLI The firewall on this machine is pretty permissive but I even turned that off for a while to see if that was the problem but it had no effect. I've reconfirmed the Google credentials and I've also tried making the calendar public but I never see any events that I have added. I have set reminders on the events themselves as I know they wont show without this. Can anyone give me any pointers on where to look to debug as I'm struggling a touch right now. Thanks in Advance Ish Additionally, I've done a tcpdump on the server and can see 2 way traffic to 173.194.66.105 Got this working. I had to reboot the server. and use ical rather than caldav -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Asterisk 1.8 and calendar intergration
Hi I'm using asterisk 1.8.25.0 on CentOS 6. I have compiled it with all the calendar modules: *CLI module show like calendar Module Description Use Count res_calendar.soAsterisk Calendar integration4 res_calendar_ews.soAsterisk MS Exchange Web Service Calenda 0 res_calendar_caldav.so Asterisk CalDAV Calendar Integration 0 res_calendar_exchange.so Asterisk MS Exchange Calendar Integratio 0 res_calendar_icalendar.so Asterisk iCalendar .ics file integration 0 5 modules loaded I'm trying to integrate this with a new calendar I've created on an existing Google account but can't get it to work. I've tried ical with the ical url from the calendar settings and I've tried caldav using https://www.google.com/calendar/dav/my-cal-id/events/ as the url but it just won't work. I've added events in the next couple of hours with reminders on the calendar that I'm referencing but the asterisk just wont pick up the events: *CLI calendar show calendar ishcal Name : ishcal Notify channel: SIP/a-sip-peer Notify context: Notify extension : Notify applicatio : Playback Notify appdata: tt-weasels Refresh time : 1 Timeframe : 3600 Autoreminder : 10 Events -- *CLI The firewall on this machine is pretty permissive but I even turned that off for a while to see if that was the problem but it had no effect. I've reconfirmed the Google credentials and I've also tried making the calendar public but I never see any events that I have added. I have set reminders on the events themselves as I know they wont show without this. Can anyone give me any pointers on where to look to debug as I'm struggling a touch right now. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 1.8 and calendar intergration
On 15 May 2014 16:03, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8.25.0 on CentOS 6. I have compiled it with all the calendar modules: *CLI module show like calendar Module Description Use Count res_calendar.soAsterisk Calendar integration4 res_calendar_ews.soAsterisk MS Exchange Web Service Calenda 0 res_calendar_caldav.so Asterisk CalDAV Calendar Integration 0 res_calendar_exchange.so Asterisk MS Exchange Calendar Integratio 0 res_calendar_icalendar.so Asterisk iCalendar .ics file integration 0 5 modules loaded I'm trying to integrate this with a new calendar I've created on an existing Google account but can't get it to work. I've tried ical with the ical url from the calendar settings and I've tried caldav using https://www.google.com/calendar/dav/my-cal-id/events/ as the url but it just won't work. I've added events in the next couple of hours with reminders on the calendar that I'm referencing but the asterisk just wont pick up the events: *CLI calendar show calendar ishcal Name : ishcal Notify channel: SIP/a-sip-peer Notify context: Notify extension : Notify applicatio : Playback Notify appdata: tt-weasels Refresh time : 1 Timeframe : 3600 Autoreminder : 10 Events -- *CLI The firewall on this machine is pretty permissive but I even turned that off for a while to see if that was the problem but it had no effect. I've reconfirmed the Google credentials and I've also tried making the calendar public but I never see any events that I have added. I have set reminders on the events themselves as I know they wont show without this. Can anyone give me any pointers on where to look to debug as I'm struggling a touch right now. Thanks in Advance Ish Additionally, I've done a tcpdump on the server and can see 2 way traffic to 173.194.66.105 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Realtime peers and sendrpid
Hello all If I look at the sip peers table definition as provided with the source of asterisk-1.8.23.0/ (looking at contrib/realtime/mysql/sippeers.sql) for the sendrpid column it's an enum with 2 possible values, yes and no. However, the sip.conf allows 4 values, no, yes, rpid and pai. Is this discrepancy an oversight? Is it possible to set the system default to pai but an individual peer to rpid via a realtime table? I have tried setting the system value to pai and a single peer value to yes but it still sent pai rather than rpid. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] CDR billsec issue with calls forwarded through the Local channel
Hi I'm using asterisk 1.8.23.1 but I've seen this same issue in previous versions of 1.8. I have created some work arounds but the behaviour is incorrect. This is the scenario: Call comes in and goes to appropriate dialplan In the dialplan the call is forwarded to another number using a Local channel (and using /n ) e.g. Dial(Local/my-number@outbound-context/n,60) The number is dialled and the call is all fine. In the CDR we have 2 entries, one for the inbound leg and one for the outbound leg as is expected by the use of the /n However, the outbound leg CDR entry has a billsec of 0. The CDR for the inbound leg has the correct duration of the call in the billsec column (I'm writing CDRs to MySQL) This is causing issues in my billing module for obvious reasons. I'm having to find the inbound call by matching the channel in one leg with the dstchannel in the other leg and that is quite messy. Would others agree that this behaviour is incorrect? Has anyone else seen this or be able to replicate it? Am I just missing something obvious? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] CDR billsec issue with calls forwarded through the Local channel
A big correction to the above! This 0 billsec entry happens when someone forwards a call from their phone using an auto forward (which then uses a Local channel on the asterisk server). The phone in question is a Snom. If I use a Local channel in the dial plan, the entry has a the correct billsec. On 2 May 2014 11:23, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8.23.1 but I've seen this same issue in previous versions of 1.8. I have created some work arounds but the behaviour is incorrect. This is the scenario: Call comes in and goes to appropriate dialplan In the dialplan the call is forwarded to another number using a Local channel (and using /n ) e.g. Dial(Local/my-number@outbound-context/n,60) The number is dialled and the call is all fine. In the CDR we have 2 entries, one for the inbound leg and one for the outbound leg as is expected by the use of the /n However, the outbound leg CDR entry has a billsec of 0. The CDR for the inbound leg has the correct duration of the call in the billsec column (I'm writing CDRs to MySQL) This is causing issues in my billing module for obvious reasons. I'm having to find the inbound call by matching the channel in one leg with the dstchannel in the other leg and that is quite messy. Would others agree that this behaviour is incorrect? Has anyone else seen this or be able to replicate it? Am I just missing something obvious? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Channel names
Hi I'm using asterisk 1.8. How are channel names constructed. I always thought they were technology/peer-hex counter but I've had a lot of instances where a channel name doesn't have the correct peer as part of it. Is it unwise to use channel names to extract the peers involved in a call? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Putting a notice in the logs from the dialplan
Hi Using asterisk 1.8 NoOp and Verbose both put messages into the logs as VERBOSE, is there any way to put a message into the logs as NOTICE from within a dial plan? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Channel names
On 1 May 2014 15:19, Matthew Jordan mjor...@digium.com wrote: On Thu, May 1, 2014 at 7:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8. How are channel names constructed. I always thought they were technology/peer-hex counter but I've had a lot of instances where a channel name doesn't have the correct peer as part of it. Is it unwise to use channel names to extract the peers involved in a call? How a channel is named is a function of the channel technology. Which channel technology(ies) are you curious about? Matt SIP only Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Putting a notice in the logs from the dialplan
That works a treat, thank you. On 1 May 2014 15:28, Steven Wheeler swhee...@usinternet.com wrote: On Thu, May 1, 2014 at 8:37 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Using asterisk 1.8 NoOp and Verbose both put messages into the logs as VERBOSE, is there any way to put a message into the logs as NOTICE from within a dial plan? Thanks in advance What about the Log application? It is available on our Asterisk 1.8.26 box. Connected to Asterisk 1.8.26.0 Verbosity is at least 3 CLI core show application Log -= Info about application 'Log' =- [Synopsis] Send arbitrary text to a selected log level. [Description] Sends an arbitrary text message to a selected log level. [Syntax] Log(level,message) [Arguments] level Level must be one of 'ERROR', 'WARNING', 'NOTICE', 'DEBUG', 'VERBOSE' or 'DTMF'. message Output text message. [See Also] Not available -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Anyone used WatchGuard SIP ALG?
On 22 April 2014 16:24, Tony Mountifield t...@softins.co.uk wrote: Has anyone here used Asterisk inside a WatchGuard firewall, talking via the WatchGuard SIP Application Layer Gateway to an outside SIP service? I have a customer doing just that, and I am 100% convinced there is a bug in the ALG regarding the media port number it inserts into the SDP when it rewrites it. However, either they or WatchGuard will not accept there is a bug, despite my very detailed description of it. So if anyone else has any experience of using this product, I'd be very interested to hear from you. Thanks! Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org Just about every SIP ALG (Watchguard included) makes things worse or simply not work. Have you tried to simply disable it? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] AMI and pyst
On 14 April 2014 16:34, Matthew Jordan mjor...@digium.com wrote: On Thu, Apr 10, 2014 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Does anyone on this list use pyst for AMI purposes? If so, can you point me in the direction of some simple examples. There seems to be none anywhere online. Probably doesn't help that I'm not that experienced at python but not insurmountably so. Thanks in Advance Ish Hey Ish - This isn't directly answering your question, but I noticed no one chimed in. At Digium we don't use pyst for Python integration with Asterisk, so I don't have any experience with it. We do, however, use starpy (https://github.com/asterisk/starpy) extensively in the Asterisk Test Suite. It does lock you into using twisted (https://twistedmatrix.com/trac/) - which has both pros and cons - but it may be a viable alternative for you if pyst doesn't work out. Matt Hi Matt Thanks for the reply. I actually chose pyst as Billy Chia said that's what you guys used when I was at Astricon last year... Anyway, I overcame my initial hurdle but do want to try out the alternatives before I commit to one library -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 to Microsoft Lync2013?
On 11 April 2014 11:34, Tony Mountifield t...@softins.co.uk wrote: Are they any gotchas to be aware of in getting Asterisk and Lync 2013 talking to each other using SIP? Or is Lync a pretty standard implementation of SIP? Cheers Tony You have to use TCP for transport and you need to define the host and port address in your peer config and then secure it with ACL. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] AMI and pyst
Does anyone on this list use pyst for AMI purposes? If so, can you point me in the direction of some simple examples. There seems to be none anywhere online. Probably doesn't help that I'm not that experienced at python but not insurmountably so. Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 1.6
On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 1.6
Well in that case fail2ban gets my vote. On 4 April 2014 16:15, motty cruz motty.c...@gmail.com wrote: Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote: thank you all for your support. I am using Linux, I only have about 7 users outside our home network. I will learn fail2ban and will use it accordingly. again Thanks for your support. Do the 7 users outside of your home network always connect from the same IP addresses? If so, you can just lock down your SIP port to those 7 IPs explicitly in your IPTables configuration. Another option would be to change which port you're running SIP on. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Numbers hackers call
Hi The 11 bit is them thinking there's some prefix which will cause your PBX to become an open relay. The number (97259) is a Palestine Mobile number. These's a lot of hacking attempts coming from Palestine and this type of number probably has some revenue generation properties to it. Regards Ish On 26 March 2014 15:05, Michelle Dupuis mdup...@ocg.ca wrote: I see a lot of attempts by hackers to call 00972595301123 or 011972595115207 or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XX is unclear... -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] AMI Proxy
Hi people Just having a quick check to see if anyone is using any AMI proxies and which are the most popular. For our purposes it must be able to connect to multiple asterisk instances. Thanks for the help. Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] WebRTC and Asterisk 12
On 20 March 2014 19:24, Dan Cropp d...@amtelco.com wrote: Anyone know of a tutorial for configuring WebRTC on Asterisk 12 using PJSIP? Some useful stuff here, it's video's from last Astricon: https://www.youtube.com/playlist?list=PLighc-2vlRgSwgJCxEh6NZwC8lE6XogaP This particular session might be helpful https://www.youtube.com/watch?v=GHFduPTNE1Qindex=9list=PLighc-2vlRgSwgJCxEh6NZwC8lE6XogaP Not sure it's as detailed as you'd like though. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 Authentication
On 11 March 2014 11:39, Jim Boykin boykin...@gmail.com wrote: Hi, I am trying to setup asterisk so that anyone from any IP can call using any callerid as long they have an account - also no registration is required. However, it seems like asterisk tries to find peer based on either the IP address or from header. What I really want is asterisk to find account/peer based on username passed as part of the authentication and NOT from the IP address or the from header. Any idea how to achieve this. Thanks It has to be either fixed IP address or username and password with a dynamic host. This is no in between to the best of my knowledge. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] chan_sip.c:3115 __sip_xmit of 0x108d33c0 (len 523) to xxx.xxx.xxx.xxx:0 returned -1: Invalid argument
Hi Re raising this issue as it's still affecting me. Where is the asterisk server getting port 0 from? We use ARA and port 0 is neither in the full contact not in the port field of the sip table. Nor is port 0 in the realtime cache for any peer registering from the IP address generating the error. On 16 March 2011 18:19, Tilghman Lesher tilgh...@meg.abyt.es wrote: On Wednesday 16 March 2011 06:09:33 Ishfaq Malik wrote: Does anyone know what this error is about? I've had 0 success in trying to find any reference to it on the internet Well, the most obvious problem is that you cannot send (or bind, or do anything, really) to port 0. -- Tilghman -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] h extension isn't processed after call file finishes.
What version of asterisk are you using? Ish On 17 February 2014 20:49, Mike Diehl mdiehlena...@gmail.com wrote: Hi all, I'm trying to build a fax relay mechanism where faxes come in and get relayed out to their final destination. I'm using the h extension to store various results from both legs. This data is being saved correctly for the first (receiving) leg. The second leg isn't calling the h extension when it's finished. The second leg is being initiated by a .call file like: Channel: local/1505xxx@context Application: sendfax Data: /tmp/voice11-voice11-1392668806.182025.tiff,zfds WaitTime: 90 MaxRetries: 2 Account: vFax CallerID: Fax 505xxx The h extension calls an agi scrip that logs a bunch of information about the fax attempt. Works just fine when I receive a fax. But there is no sign of it in the logs for the sending leg of the fax. Is there something I need to do in order to get the h extension to get called? Mike. -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Retaining P-Asserted Info
HI Have you tried: sendrpid = pai ; Use the P-Asserted-Identity header ; to send the identity of the remote party in the sip.conf? Regards Ish On 16 February 2014 20:29, Nick Cameo sym...@gmail.com wrote: Hello Markus, Thank you so much for your response. Our switch is already generating the needed P-Asserted header: P-Asserted-Identity: John Doe sip:14167493...@toronto.location.com; user=phone; nat=yes. I really did not want to have to rebuild it using `SIPAddHeader` however, if I have no choice, can someone please provide an extension rule that will include the exiting inbound leg line above in the outbound leg. Kind Regards, Nick. -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Looking for some guidance with the Asterisk 12 ARI/API
We run a multi node, multi tenanted hosted VoIP service using centralised databases for sip/extensions/voicemail configuration allowing resellers and end users to make updates to their walled garden themselves. We're using asterisk 1.8 but Realtime is no different on asterisk 12 (with the exception of PJSIP). Not done anything with the ARI. On 6 February 2014 15:10, James Wystead szilvertho...@gmail.com wrote: Hi - I figured this was probably the best place to ask this question Is there anyone that has done anything practical with the API and/or Real Time Database config? If so, I would like to pick your brains if I may. Thanks - G -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Change the preferred audio playback format
Hi Is there any way to change the preferred audio playback format in asterisk (I'm using 1.8.25.0) i.e. first check for gsm, if doesn't exits then check for slin? Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 ignoring nat settings
Is directmedia set to no? On 15 January 2014 23:11, Leandro Dardini ldard...@gmail.com wrote: Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the sip show peer extension, I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Yes Asterisk is behind a nat the the externip and localnet has been configured. The local net on the asterisk network is different from the local net on phone. What else could I check? Leandro -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 API
On 10 January 2014 17:12, James Wystead szilvertho...@gmail.com wrote: Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something available? Thanks - Glen You could use asterisk realtime architecture and use your favourite database to hold peer/voicemail/dialplan configuration. https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] CTI
Hi people I'm just mailing to see what people are using for CTI solutions with asterisk. Aslos, has anyone managed to integrate asterisk with Salesforce? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 NAT friendly settings
On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net wrote: I'm asking about this scenario: Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia Have you added directmedia=no? Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Reading DTMF sent by callee during a SIP call
On 20 December 2013 16:13, Alex ralie...@gmail.com wrote: Hi everyone, I am looking for advice about the design of a SIP-based intercom. I count on your help, as my current attempts are not fruitful (yet). This will be a pretty long message, so here's my fundamental question: Is there a way to interpret DTMF tones sent by the calee (not the caller) while a voice call is in progress? Here's the desired scenario: - there is a box with speakers and a mic - Asterisk is running on a computer inside that box - the box is embedded in a door - There are two user accounts, UserA and userB - UserA is a client that runs on the server* - UserA calls UserB and they are having a voice conversation Throughout the call, Asterisk must react to DTMF tones sent by userB; such that an action is executed when a specific key is pressed. The idea is to build an intercom that would enable me to open a door remotely, by relying entirely on SIP, so there would be no need to have some additional communication channel to send the open door signal. I have previously implemented IVRs using `Background` and jumped to specific extensions, when a button was pressed. But in that case, the extensions are dialed by the caller; whereas now the input must from the person who answered the call. If I use `Dial` and `Read` - the latter is only executed after `Dial` terminates - so this is not suitable. `Background` behaves like I need - but it plays back a predefined file, so it is not suitable for an interactive conversation. * Having a SIP client on the same machine as the Asterisk server itself is not possible, because both won't be able to bind to port 5060. My guess is that the solution is to originate a call from the CLI; but I haven't gotten to that part yet. Thank you for your patience, I am looking forward to your feedback, Alex You could create your own feature in features.conf that executes a Macro/Gosub defined in sip.conf... Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Lync and Asterisk Realtime Architecture
Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer peer-name load. Has anyone got any experience of connecting to Lync using ARA? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Lync and Asterisk Realtime Architecture
Hi Eric, thanks for that. I hadn't been specifying a port, I'll give it a go now. On 5 December 2013 15:39, Eric Wieling ewiel...@nyigc.com wrote: If the device is not registering then you have to specify the port as well as the ip in the database entry for the peer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Thursday, December 05, 2013 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Lync and Asterisk Realtime Architecture Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer peer-name load. Has anyone got any experience of connecting to Lync using ARA? Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Unix connections not always disconnecting
in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] Unix connections not always disconnecting
On 7 November 2013 15:26, Gareth Blades mailinglist+aster...@dns99.co.ukwrote: On 07/11/13 11:20, Ishfaq Malik wrote: Hi We are using asterisk 1.8.23.1 We have a script that runs on a minute cron which polls the asterisk server for 3 bits of information by using asterisk -rx 'command' which then gets pushed to a graphite server we have 99% of this runs smoothly. Out of interest what are you trying to monitor? We tend to use cacti for graphing and snmp provides all the information we require. Active calls, sip peers connected, sip peers disconnected and then breaking all of those down by customer as we run a multi tenanted set up. SNMP would give us totals but I don't think it would do the breakdown by customer. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] No matching peers message has gone (1.8.23.1)
Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? Thanks in Advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] No matching peers message has gone (1.8.23.1)
Hi Thanks for the quick response. I'll read all the change logs from now on, I promise! Ish On 4 November 2013 15:29, Joshua Colp jc...@digium.com wrote: Ishfaq Malik wrote: Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? This is on purpose. To fix some exposure issues the code was changed to have an internal peer (albeit one that can never successfully be authenticated against) that gets used if no real peer is found. This reduces the chance (by a lot) of the code exposing information in some off nominal cases. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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
