Re: [asterisk-users] New DAHDI complete
https://issues.asterisk.org/jira/browse/DAHLIN-379 Le mer. 20 janv. 2021 à 19:50, Jerry Geis a écrit : > When might there be a new dahdi complete to support the 5.4 kernel? > Thanks, > > Jerry > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 16.2.1: P-Asserted-Id set to s when Dialing inside a certain Gosub
May I add that, to me, I would expect Asterisk to use CALLERID vlaues (name
and num) to set P-Asserted-Id.
Maybe in a couple of days, I'll report my findings here if can find some
time to experiment with Asterisk 17 or Asterisk 18 and compare behaviours..
Le mar. 8 déc. 2020 à 16:41, Olivier a écrit :
> Hello,
>
> With Debian Buster's Asterisk 16.2.1, please consider the following
> dialplan
> ;Case A
> ;exten = 29,1,Dial(PJSIP/${EXTEN})
> ;Case B
> ;exten = 29,1,Gosub(foo,${EXTEN},1)
> ;Case C
> exten = 29,1,Gosub(bar,s,1(${EXTEN}))
>
> [foo]
> exten = _X.,1,Dial(PJSIP/${EXTEN})
> same = n,Return()
>
> [bar]
> exten = s,1,Set(TGT=${ARG1})
> same = n,Dial(PJSIP/${TGT})
> same = n,Return()
>
> When running case A et B, dialing phone gets a Ringing signal from
> Asterisk with a P-Asserted-Id
> header set to 29: this is expected.
>
> When running case C, dialing phone gets a Ringing signal from Asterisk
> with a P-Asserted-Id
> header set to s: this is not what I expected.
>
>
> Thoughts ?
> Is there a way to work around this without using pattern B above ?
>
> Best regards
>
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[asterisk-users] Asterisk 16.2.1: P-Asserted-Id set to s when Dialing inside a certain Gosub
Hello,
With Debian Buster's Asterisk 16.2.1, please consider the following dialplan
;Case A
;exten = 29,1,Dial(PJSIP/${EXTEN})
;Case B
;exten = 29,1,Gosub(foo,${EXTEN},1)
;Case C
exten = 29,1,Gosub(bar,s,1(${EXTEN}))
[foo]
exten = _X.,1,Dial(PJSIP/${EXTEN})
same = n,Return()
[bar]
exten = s,1,Set(TGT=${ARG1})
same = n,Dial(PJSIP/${TGT})
same = n,Return()
When running case A et B, dialing phone gets a Ringing signal from Asterisk
with a P-Asserted-Id
header set to 29: this is expected.
When running case C, dialing phone gets a Ringing signal from Asterisk with
a P-Asserted-Id
header set to s: this is not what I expected.
Thoughts ?
Is there a way to work around this without using pattern B above ?
Best regards
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Re: [asterisk-users] Which is the most FHS-esque way to run several Asterisk instances on a single host ?
Link [1] interestingly details how you can run several daemon instances with systemctl. Note that the author uses things like /run/asterisk/instance-foo /var/lib/asterisk/instance-foo [1] https://opensource.com/article/20/12/multiple-service-instances-systemctl Le ven. 20 nov. 2020 à 18:29, Olivier a écrit : > Hello, > > What is the most FHS-esque (see [1]) way to run several Asterisk > instances on a single (Debian) host ? > > What would you recommend ? > Would gather each instance directories (etc/, run/, lib/, ...) in > something like /srv/instance1/ > (it doesn't please me as I like to put variable data in /var and on so) ? > > Alternatively, would you with /etc/asterisk1/, /var/lib/asterisk1, ... ? > > Would you even create a dedicated system user, one per instance, to > further isolate asterisk instances data ? > > [1] https://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard > > Cheers > > PS: On a Debian-packaged Asterisk, I've got /usr/sbin/rasterisk linked to > /usr/sbin/asterisk. How can you explain directly running /usr/sbin/asterisk > "requires" a -r option while /usr/sbin/rasterisk does not ? > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which is the most FHS-esque way to run several Asterisk instances on a single host ?
Hello, What is the most FHS-esque (see [1]) way to run several Asterisk instances on a single (Debian) host ? What would you recommend ? Would gather each instance directories (etc/, run/, lib/, ...) in something like /srv/instance1/ (it doesn't please me as I like to put variable data in /var and on so) ? Alternatively, would you with /etc/asterisk1/, /var/lib/asterisk1, ... ? Would you even create a dedicated system user, one per instance, to further isolate asterisk instances data ? [1] https://en.wikipedia.org/wiki/Filesystem_Hierarchy_Standard Cheers PS: On a Debian-packaged Asterisk, I've got /usr/sbin/rasterisk linked to /usr/sbin/asterisk. How can you explain directly running /usr/sbin/asterisk "requires" a -r option while /usr/sbin/rasterisk does not ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load testing SIP registration attempts
Hello, How would you test how a PJSIP-powered Asterisk 13 instance resist to hostile REGISTRATION attempts ? Would you use SIPp ? Any example scenario ? Would you go with an alternative tool ? Which one would you pick ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suden "ast_db_put: Couldn't execute statment" in 13.14.1 after high rate of incoming REGISTERs
Hello, The other day, a 13.14.1 server suddenly stopped working correctly. First, it printed: Oct 23 21:53:40 FOOBAR asterisk[2377]: WARNING[27942]: db.c:332 in ast_db_put: Couldn't execute statment: SQL logic error or missing database This occurred while this server received a lot incoming REGISTER such as: Oct 23 21:53:40 FOOBAR asterisk[2377]: [Oct 23 21:53:40] #033[1;33mNOTICE#033[0m[20418]: #033[1;37mres_pjsip/pjsip_distributor.c#033[0m:#033[1;37m525#033[0m #033[1;37mlog_failed_request#033[0m: Request 'REGISTER' from '< sip:johndoe@100.64.0.1>' failed for '10.50.59.5:5060' (callid: OA48732DD2johndoe42259F1C) - Failed to authenticate I estimate incoming (invalid) REGISTER rate to be 120 per second, (which is by the way, well above the upper limit we load tested our system before putting it online). What would you suggest me to do ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP tight loop on auth failure
Hi, What if some fail2ban magic could keep OpenSIPs response from hitting Asterisk after N attempts ? Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd < kingsley.t...@barritel.com> a écrit : > Hi, > > We're using Asterisk 13.17.0 with PJSIP 2.8 bundled. > > I've found an issue when Asterisk tries to make a SIP call out using > auth, but has the wrong credentials and keeps getting returned a SIP > 407, in this example to an OpenSIPs server requiring user auth. > > Basically this happens: > >1. Asterisk sends plain INVITE to OpenSIPs >2. OpenSIPs responds with SIP 407 auth required with a Proxy- > Authenticate header >3. Asterisk re-sends INVITE to OpenSIPs with Proxy-Authorization > header, but has the wrong password >4. goto step 2 and repeat forever > > So what we're seeing is Asterisk re-sending an INVITE with incorrect > auth (which is clearly never going to work), about every 2ms. > > The Call-ID remains the same all of the time. > > Shouldn't PJSIP realise that this isn't going to work after a few tries > and give up? > > The only way I've found of stopping the seemingly infinite loop is to > either restart Asterisk or temporarily block network traffic between > the two machines in order to break the cycle. > > Any idea whether this has been fixed in a later version? > > This is basically the response coming back from OpenSIPs (anonymised), > whether Asterisk didn't provide > > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 100.101.102.103:5060 > ;received=100.101.102.103;rport=5060;branch=z9hG4bKPja942e87d-c501-4834-9184-f002c3fd53d2 > From: >;tag=075f669f-9115-42a8-8c98-6170a2910e4b > To: >;tag=c97b4d1cb1f3d0da549e06a8d482ef63.fefa > Call-ID: f79caf90-5b95-4db7-966b-a42e2d372c90 > CSeq: 34157 INVITE > Proxy-Authenticate: Digest realm="sip.example.com", > nonce="5f96c21800011caac9f7e901848de60a1e186b402bd9b710", qop="auth" > Server: OpenSIPS (1.11.6-tls (x86_64/linux)) > Content-Length: 0 > > The caveat is that whether what OpenSIPs is doing is correct or broken, > our customers can edit the auth on their own SIP gateways, so our > system needs to be able to handle it properly. > > Cheers, > Kingsley. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Doc for PJSIP ICE support ?
Thanks Joshua for replying ! What would you advise : - leaving STUN address empty, in rtp.conf, as "STUN is not required for ICE" - configure it with one public STUN (I'm using stun.voip.ovh.net for this but I don't know how this server really works) Cheers Le mar. 27 oct. 2020 à 09:53, Joshua C. Colp a écrit : > On Tue, Oct 27, 2020 at 5:35 AM Olivier wrote: > >> Hello, >> >> Where can I find doc about PJSIP's ice_support parameter ? >> >> Do you need to configure things elsewhere in Asterisk config files >> (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? >> > > It needs to also be enabled in rtp.conf. > > >> I'm asking because, if I'm not mistaken, STUN requires setting a STUN >> server so I think ICE most probably, should also require settings some >> public resources. >> > > STUN is not required for ICE, but can be used to identify another > possible ICE candidate. The implementation itself is the same as chan_sip, > so you would need to also configure a STUN server in rtp.conf if you want > to use STUN with ICE. > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Doc for PJSIP ICE support ?
Hello, Where can I find doc about PJSIP's ice_support parameter ? Do you need to configure things elsewhere in Asterisk config files (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? I'm asking because, if I'm not mistaken, STUN requires setting a STUN server so I think ICE most probably, should also require settings some public resources. Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different atxfer, pickup sequences for different phone users
Hello, Is it possible to set different features.conf dialing sequences (atxfer, pickup, ...) for different users ? For instance, what if I want Alice to dial *8 to pickup a call and Bob to dial ** to pickup calls ? I can see that features.conf includes application maps but can these be used for the above goal ? Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommandation for PJSIP trunking when two WAN access are available
Hello, In project, a customer has two WAN access. More precisely: Internet - --- Router1 --- FortiGate Firewall Router -- Asterisk | | - --- Router2 -- Both WAN access offer Gigabit-like performance and each is able to transport appropriately both Data and VoIP traffic. I'm not directly managing the FortiGate but I suppose it has some tools to monitor each WAN link and route accordingly. My ITSP requires REGISTER so that if I ever has to switch from one route to another, I should re-send a new REGISTER to get incoming call back in (but I think can still send outbound call using a second route without needing e-Registration first) 1. How would you make use of both access to increase uptime ? 2. What is the most efficient way to monitor route states ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP AoR vs Endpoint
Le sam. 18 juil. 2020 à 08:02, Andrew Yager a écrit : > Hi, > > I realise this is an old question, but I’m struggling to get my head > around it. > > The ERD suggests that endpoints can link to multiple AoRs > > In what situation would you actually use this? Given that mapping of > inbound calls is primary done to the endpoint, > In my very limited understanding, it seems inbound calls, if properly configured, are targeted to AoRs, not endpoints. This allows for several physical devices to be notified of the same incoming call. Following this route, one hurdle to take on is when transferring calls from one device to the other one(s), with the same endpoint. Maybe, some with a better understanding or knowledge may correct or confirm the above statements. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Example of Jitsi Desktop provisioning file
>From [3], it seems possible to change provisioning URLs with (untested, yet): net.java.sip.communicator.plugin.provisioning.URL net.java.sip.communicator.plugin.provisioning.auth.USERNAME net.java.sip.communicator.plugin.provisioning.auth net.java.sip.communicator.plugin.provisioning.METHOD [3] https://searchcode.com/codesearch/view/50276540/ Le mar. 21 juil. 2020 à 12:00, Sylvain Boily a écrit : > Hello, > > On 2020-07-21 3:57 a.m., Olivier wrote: > > Hi, > Le ven. 10 juil. 2020 à 16:56, Sylvain Boily a écrit : > >> >> It probably can help you: >> >> https://github.com/wazo-platform/wazo-provd-plugins/blob/master/plugins/xivo-jitsi/1/templates/base.tpl >> >> Sylvain >> >> Yes, provided example was exactly what I was after ! > Thank you very much ! > > > You're welcome. > Sylvain > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Example of Jitsi Desktop provisioning file
Hi, Le ven. 10 juil. 2020 à 16:56, Sylvain Boily a écrit : > > It probably can help you: > > https://github.com/wazo-platform/wazo-provd-plugins/blob/master/plugins/xivo-jitsi/1/templates/base.tpl > > Sylvain > > Yes, provided example was exactly what I was after ! Thank you very much ! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redis in place of astdb
Though ASTDB use is very flexible with DB() function, abandoning this
flexibility to gain REDIS features (clustering, speed, ..) has benefits.
Still, being able to "implement ASTDB with REDIS" would allow a lot
currently impossible setups (synchronizing SIP registrations on multiples
Asterisk instances, ...)
Le mer. 8 juil. 2020 à 21:24, John Kiniston a
écrit :
> Dovid, You could use func_odb + a ODBC Redis driver to keep from having to
> shell out.
>
> On Wed, Jul 8, 2020 at 4:37 AM Dovid Bender wrote:
>
>> Hi,
>>
>> Does anyone know of any projects that would allow you to use Redis in
>> place of AstDB? By in place of I don't mean for what Asterisk needs but to
>> store values. For instance for CNAM currently we need to use an AGI to
>> connect to redis to pull CNAM. So in place of:
>> Set(CALLERID(name)=${DB(CNAM/${CALLERID(num)})}
>> it would be done with redis for example:
>> Set(CALLERID(name)=${REDIS1(CNAM_${CALLERID(num)})}
>>
>> If not can the devs here give me a pointer to where to look?
>>
>> TIA.
>>
>> Regards,
>>
>> Dovid
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> 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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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Example of Jitsi Desktop provisioning file
Hello, 1. I'm looking for an (anomized) example of a Jitsi Desktop provisioning file compliant with Asterisk ? Jitsi Doc mentions it should adhere to Java properties file syntax (see [1]) but a working example would help. If this example file included the following settings, it would be perfect: - sip username, secret, authname, ... - codec G711, RFC2833 - BLF - local or remote phonebook - provisioning through HTTP(S) [1] https://desktop.jitsi.org/provisioning.html 2. Have ever tried passing some setting to installer (see [2]). Comments ? [2] https://desktop.jitsi.org/Documentation/MSIParameters.html Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting to get BLF working with linphone
It seems a new Linphone 4.2 is to be published next week ! Hopefully, ... Le ven. 5 juin 2020 à 13:34, John Hughes a écrit : > On 26/05/2020 15:33, Olivier wrote: > > Hi John, > > 1. Could you get any further, in your quest for working BLF with linphone ? > > The patches to get linphone-3.12 BLF working with Asterisk are here: > > http://perso.calvaedi.com/~john/linphone-3/ > > They're pretty damnned trivial: > > 1. add the "Accept" header to the SUBSCRIBE message so asterisk doesn't > reject it. > > 2. don't trash the SIP dialog if the SUBSCRIBE refresh is rejected because > of a stale nonce. > > 3. If asterisk says the user is on the phone set the status to on the > phone. > > All except the 3rd one are compatible with linphone-4. Implementing the > same feature with linphone-4 is left as an exercise for the reader. > > > 2. Have you tried with a different Linphone version (4.12 is pending on > Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? > > > Version 4 of linphone is, frankly, rubbish. I have managed to hack it to > the point where presence shows green for connected contact and grey for > disconnected. However this requires setting the "send subscribe" flag in > the linphone contacts db and linphone 4 has no UI for setting this flag, > you have to do it using sqlite3 directly (or setting up your contacts in > linphone 3). > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting to get BLF working with linphone
Lately, I read [1]. So it seems both Jitsi desktop and Linphone are on par, on this ;-))) [1] https://community.jitsi.org/t/busy-lamp-field-bug/15931 Le ven. 5 juin 2020 à 13:34, John Hughes a écrit : > On 26/05/2020 15:33, Olivier wrote: > > Hi John, > > 1. Could you get any further, in your quest for working BLF with linphone ? > > The patches to get linphone-3.12 BLF working with Asterisk are here: > > http://perso.calvaedi.com/~john/linphone-3/ > > They're pretty damnned trivial: > > 1. add the "Accept" header to the SUBSCRIBE message so asterisk doesn't > reject it. > > 2. don't trash the SIP dialog if the SUBSCRIBE refresh is rejected because > of a stale nonce. > > 3. If asterisk says the user is on the phone set the status to on the > phone. > > All except the 3rd one are compatible with linphone-4. Implementing the > same feature with linphone-4 is left as an exercise for the reader. > > > 2. Have you tried with a different Linphone version (4.12 is pending on > Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? > > > Version 4 of linphone is, frankly, rubbish. I have managed to hack it to > the point where presence shows green for connected contact and grey for > disconnected. However this requires setting the "send subscribe" flag in > the linphone contacts db and linphone 4 has no UI for setting this flag, > you have to do it using sqlite3 directly (or setting up your contacts in > linphone 3). > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting to get BLF working with linphone
Hi John, 1. Could you get any further, in your quest for working BLF with linphone ? 2. Have you tried with a different Linphone version (4.12 is pending on Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? Best regards Le mer. 25 mars 2020 à 15:06, John Hughes a écrit : > > On 23/03/2020 18:51, Joshua C. Colp wrote: > > On Mon, Mar 23, 2020 at 2:45 PM John Hughes wrote: > >> >> Why is asterisk giving an error 500? I can find no reason, there is >> nothing in any log. >> > > The sequence number is from the past. The first SUBSCRIBE is sequence > number 22 (check the CSeq header). The second is 20. The third is 21. It > appears as though this is from the past, so it receives a 500. > > Ok, I've had some back and forth with the linphone developers and they > contend that although the sequence number on the 2nd and 3rd SUBSCRIBE > messages start a new sequence this is legal as it is a new conversation -- > the "tag=" on the From has changed. > > Are they right? (Notice that the tag= from asterisk also changes). > > <--- SIP read from UDP:10.27.128.3:5060 ---> > SUBSCRIBE sip:jacques@10.27.128.1:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport > From: ;*tag=iGH81k5xf* > To: ;tag=as3c7de68c > CSeq: 22 SUBSCRIBE > Call-ID: SQOclJgm4O > Max-Forwards: 70 > Supported: replaces, outbound > Event: presence > Expires: 600 > Accept: application/pidf+xml > Contact: > ;+sip.instance="" > User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) > Authorization: Digest realm="asterisk", nonce="188b095b", algorithm=MD5, > username="john", uri="sip:jacques@10.27.128.1:5060", > response="bdbc7cbac4453fd643050bf28996a68e" > > <-> > --- (14 headers 0 lines) --- > Found peer 'john' for 'john' from 10.27.128.3:5060 > > <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 10.27.128.3:5060 > ;branch=z9hG4bK.NYP-ux0Zx;received=10.27.128.3;rport=5060 > From: ;*tag=iGH81k5xf* > To: ;tag=as3c7de68c > Call-ID: SQOclJgm4O > CSeq: 22 SUBSCRIBE > Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > nonce="3144c0a9", stale=true > Content-Length: 0 > > > <> > Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: > SUBSCRIBE) > > <--- SIP read from UDP:10.27.128.3:5060 ---> > SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 > Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport > From: ;*tag=c3Wvuu2XH <= new > conversation* > To: sip:jacq...@masked.masked.com > CSeq: *20 SUBSCRIBE <=== sequence restarts* > Call-ID: SQOclJgm4O > Max-Forwards: 70 > Supported: replaces, outbound > Event: presence > Expires: 600 > Accept: application/pidf+xml > Contact: > ;+sip.instance="" > User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) > > <-> > --- (13 headers 0 lines) --- > Sending to 10.27.128.3:5060 (no NAT) > Creating new subscription > Sending to 10.27.128.3:5060 (no NAT) > sip_route_dump: route/path hop: > Found peer 'john' for 'john' from 10.27.128.3:5060 > > <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 10.27.128.3:5060 > ;branch=z9hG4bK.oxfLJBaRw;received=10.27.128.3;rport=5060 > From: ;tag=c3Wvuu2XH > To: sip:jacq...@masked.masked.com;tag=as007ffc64 > Call-ID: SQOclJgm4O > CSeq: 20 SUBSCRIBE > Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4224acfb" > Content-Length: 0 > > > <> > Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: > SUBSCRIBE) > > <--- SIP read from UDP:10.27.128.3:5060 ---> > SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 > Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport > From: ;tag=c3Wvuu2XH > To: sip:jacq...@masked.masked.com > CSeq: 21 SUBSCRIBE > Call-ID: SQOclJgm4O > Max-Forwards: 70 > Supported: replaces, outbound > Event: presence > Expires: 600 > Accept: application/pidf+xml > Contact: > ;+sip.instance="" > User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) > Authorization: Digest realm="asterisk", nonce="4224acfb", algorithm=MD5, > username="john", uri="sip:jacq...@masked.masked.com", > response="eb30a9801e78d2cb2c58c61200c50cb1" > > <-> > --- (14 headers 0 lines) --- > > <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> > *SIP/2.0 500 Server error* > Via: SIP/2.0/UDP 10.27.128.3:5060 > ;branch=z9hG4bK.RNv418~xv;received=10.27.128.3;rport=5060 > From: ;tag=c3Wvuu2XH > To: sip:jacq...@masked.masked.com;tag=as3c7de68c > Call-ID: SQOclJgm4O > CSeq: 21 SUBSCRIBE > Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE
[asterisk-users] Asterisk and Prometheus
Hello, I've seen that Asterisk 17 supports Prometheus but beside [1], I've not much about how to use this. Can someone shed some light on this ? 1. If I'm not mistaken, Prometheus favors "a pull model over HTTP". So basically, a Prometheus instance should be able to query Asterisk "core metrics", right ? 2. Is there an exhaustive list of those "core metrics" ? Does channel stats (core show channels, group show channels, ...) belongs to these "core metrics" ? 3. Did I miss any Dialplan function or app that uses or works with Prometheus (I couldn't one) ? [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Configuration_res_prometheus Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to build language selection menu ?
Hello, What is the recommended way to build a language selection menu like "For english, press 1, Pour le français, tapez 2, ..." ? Is possible to rely on CORE-SOUNDS-EN, CORE-SOUNDS-IT, ... to find needed audio files ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls
Hello, Hard to tell but from [1], " Exceptionally long voice queue length queuing" should not happen. 1. Are you seeing WARNING or ERROR instances in log before this "Exceptionally long voice" WARNING ? 2. If possible, I would try to load test a similar setup with SIPp and see if I can reach or exceed this 320 channels/160 calls limit. [1] https://community.asterisk.org/t/exceptionally-long-voice-queue-length-queuing/70456/15 Le mar. 21 avr. 2020 à 16:23, Stefan Viljoen a écrit : > Hi all > > I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 - > 32GB RAM. > When I approach about 320 channels, I -sometimes- get thousands of these > messages suddenly streamed in the CLI / Asterisk log: > > WARNING[60753][C-00022cb9] channel.c: Exceptionally long voice queue length > queuing to Local/xx@local-0002dbea;2 > WARNING[71993][C-00022dcc] channel.c: Exceptionally long voice queue length > queuing to Local/xx@local-0002dd10;2 > WARNING[72848][C-00022dd1] channel.c: Exceptionally long voice queue length > queuing to Local/@local-0002dd30;1 > > Top reports system load as 30.0, but the box is still very responsive over > SSH. Asterisk is shown as consuming 600% CPU when NOT in this state (but > close to it) and doesn't consume more as this state comes into effect. > > Top also shows that HT CPU instances are idle 40% of their time. > > What then happens when these messages appear is Asterisk stops accepting > originates on the AMI, and that no CDRs write through to the CDR DB anymore > - MySQL itself is idle and has no locks or running inserts indicated in > "show processlist" in MySQL CLI. > > The Asterisk instance will then effectively freeze, never hanging up > terrminated calls, and never allowing new cals to start, keeping around 320 > channels open, forever. > > Thing is I can sometimes peak at over 350 channels, then come back down as > answers, hangups, etc. take place - no problem. But -sustaining- more than > 320 channels over a two or three minutes seems to reliably cause the above > to happen. > > Asterisk has to be kill -9'ed to get it restarted, it ignores kill -1 / > sighup / cli "core stop now" > > CLI does work and is still repsonsive in this state, until I do a "core > stop > now", which doesn't stop the instance even after 20 to 30 minutes. > > Asterisk max load is set to 40 and max calls to 350. > > Any idea what I can look at? Any tips for running specifically 13 under > high > load? > > Thanks > > Stefan > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Advice on building a REST API over ASTDB
Hello,
I'm using an Asterisk 17 dialplan that currently includes:
1. many "DB gets" calls (ie statements like Set(FOO=${DB(Foo/Bar)})
2. and a couple of "DB puts" (ie statements like Set(DB(Foo/Bar)=Foo) or
DB_DELETE(Foo/bar))
I would like to add an HTTP Provisionning API that would allow an external
program to mostly change or sometimes read some ASTDB values.
Which architecture would you suggest for this ?
I can foresee the following options but I would very curious to discover
alternatives.
1. Use AMI over HTTP
2. Use ARI but if I'm not mistaken, AR does not cover ASTDB
3. Develop a custom Web App that either:
3.1 Directly read or write on /var/lib/asterisk/astdb.sqlite3 but what
about concurrency
3.2 Use shell CLI (asterisk -rx 'database put Foo Bar Baz')
4. Replace dialplan's ASTDB calls with CURL statements (if positive, how to
avoid repeating CURLOPT calls all over my dialplan).
Best regards
--
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[asterisk-users] [SOLVED]Re: TLS/SSL error loading cert file. [Almost SOLVED]
Hello, After countless hours on, this I found the root cause of HTTPS settings on Debian Buster. All this came from ast_tls_cert script using 1024 bits-long keys where Debian's defaut was to require at least 2048-long keys ! Simply passing -b 2048 to ast_tls_cert solved it. 1. May I suggest mentioning explicitly this possibility in wiki page [1] ? 2. What would you say of adding an extra input argument to have certificates built for a specific duration (default is 365 days and some may expect a different duration) ? Cheers [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients Le mer. 8 janv. 2020 à 10:04, Olivier a écrit : > Hello, > > Le lun. 6 janv. 2020 à 19:01, Olivier a écrit : > >> May I add I could successfully (if pjsip show transports has any meaning) >> add a PJSIP TLS-transport with: >> >> [transport-tls] >> type=transport >> protocol=tls >> bind=0.0.0.0:5061 >> cert_file=/etc/asterisk/keys/asterisk.crt >> priv_key_file=/etc/asterisk/keys/asterisk.key >> method=tlsv1 >> >> Le lun. 6 janv. 2020 à 18:33, Olivier a écrit : >> >>> Hello, >>> >>> On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a >>> way to enable HTTPS. >>> Asterisk is running as asterisk:asterisk: >>> >>> asterisk 11097 0.3 6.7 741352 67984 ?Ssl 17:53 0:06 >>> /usr/sbin/asterisk -g -f -p -U asterisk >>> >>> # cat /etc/asterisk/http.conf >>> [general] >>> servername=Asterisk >>> enabled=yes >>> bindaddr=0.0.0.0 >>> bindport=8088 >>> tlsenable=yes >>> tlsbindaddr=0.0.0.0:8089 >>> tlscertfile=/etc/asterisk/keys/asterisk.pem >>> ;tlsprivatekey=keys/asterisk.key >>> >>> # ls -lR /etc/asterisk/keys >>> /etc/asterisk/keys: >>> total 32 >>> -rw-rw-r-- 1 asterisk asterisk 1229 janv. 6 16:00 asterisk.crt >>> -rw-rw-r-- 1 asterisk asterisk 586 janv. 6 15:59 asterisk.csr >>> -rw-rw-r-- 1 asterisk asterisk 887 janv. 6 15:59 asterisk.key >>> -rw-rw-r-- 1 asterisk asterisk 2116 janv. 6 16:00 asterisk.pem >>> -rw-rw-r-- 1 asterisk asterisk 158 janv. 6 15:59 ca.cfg >>> -rw-rw-r-- 1 asterisk asterisk 1773 janv. 6 15:59 ca.crt >>> -rw-rw-r-- 1 asterisk asterisk 3311 janv. 6 15:59 ca.key >>> -rw-rw-r-- 1 asterisk asterisk 132 janv. 6 15:59 tmp.cfg >>> >>> # grep TLS /var/log/asterisk/full | tail -1 >>> [Jan 6 18:24:45] ERROR[11221] tcptls.c: TLS/SSL error loading cert >>> file. >>> >>> # su - asterisk --shell /bin/sh --command 'cat >>> /etc/asterisk/keys/asterisk.pem' >>> -BEGIN RSA PRIVATE KEY- >>> MIICXAIBAAKBgQCxllxfOR9sFwyKiKPZErUcBF1zlwTVZ9XvemA/8yQY7aIVw2ce >>> ... >>> RE3X5iJqFIRupoIQZQJBAJnDX8dCQbqLvmAV6/Ubiz0XHjHzLEkhMKtF/ksbgou1 >>> zykmu2rlUbnZ+DPFj/lw9WH7DaIxtogZ7qKSp0dd95g= >>> -END RSA PRIVATE KEY- >>> -BEGIN CERTIFICATE- >>> MIIDXzCCAUcCAQEwDQYJKoZIhvcNAQELBQAwNTEcMBoGA1UEAwwTQXN0ZXJpc2sg >>> ... >>> XkVjfneCBgllQhLrnb9oUBuHQCy3qtlPkXpXfAtIsodnoV1mrpI3+iKH7xWc4AtQ >>> Rbrt >>> -END CERTIFICATE- >>> >>> >>> Any clue ? >>> >>> Best regards >>> >> > > After tens of trying different settings, I tried this morning to simply > copy certs files from a running FreePBX 15 instance to my Debian Buster > target. To my surprise, it worked as for the very first time, I now have : > > # asterisk -rx 'http show status' > HTTP Server Status: > Prefix: > Server: Asterisk/16.7.0 > Server Enabled and Bound to [::]:8088 > > HTTPS Server Enabled and Bound to [::]:8089 > > > Now, to fully solve the issue, I need to understand why things didn't work > previously and now do work correctly. > > Current /etc/asterisk/keys is: > # ls -alR keys > keys: > total 56 > drwxr-xr-x 3 asterisk asterisk 4096 janv. 8 09:31 . > drwxrwxr-x 3 asterisk asterisk 4096 janv. 8 09:35 .. > -rw--- 1 asterisk asterisk 1675 janv. 8 09:31 api_oauth.key > -rw--- 1 asterisk asterisk 451 janv. 8 09:31 api_oauth_public.key > -rw-r--r-- 1 asterisk asterisk 191 janv. 8 09:31 ca.cfg > -rw-r--r-- 1 asterisk asterisk 1724 janv. 8 09:31 ca.crt > -rw-r--r-- 1 asterisk asterisk 3243 janv. 8 09:31 ca.key > -rw--- 1 asterisk asterisk 1712 janv. 8 09:31 default.crt > -rw--- 1 asterisk asterisk 1610 janv. 8 09:31 default.csr > -rw--- 1 asterisk asterisk 3247 janv. 8 09:31 default.key > -rw--- 1 asterisk asterisk 4959 janv. 8 09:31 default.pem &
[asterisk-users] [SOLVED] Re: Asterisk 17.3: No VoiceMailMain when enabling IMAP and ODBC
Hello, Adding tthe following lines in modules.conf made VoiceMailMain re-appear. load = app_voicemail.so noload = app_voicemail_odbc.so noload = app_voicemail_imap.so My previous modules.conf only included: load = app_voicemail.so I hope this would help others. Best regards Le mer. 25 mars 2020 à 15:11, Olivier a écrit : > Hello, > > On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. > > I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using > classical File module (in modules;conf and voicemail.conf): > cd asterisk-17.3.0 > ... > make menuselect.makeopts > menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; > done > menuselect/menuselect --enable app_voicemail_odbc menuselect.makeopts; > done > ... > > I've got this: > > CLI> core show application VoiceMailMain > Your application(s) is (are) not registered > Command 'core show application VoiceMailMain' failed. > > CLI> module reload app_voicemail.so > Module 'app_voicemail.so' reloaded successfully. > -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail > System)) > > CLI> module show like app_voicemail.so > Module Description > Use Count Status Support Level > app_voicemail.so Comedian Mail (Voicemail System) 0 > Running core > 1 modules loaded > > > Then I re-compiled from source removing both app_voicemail_imap and > app_voicemail_odbc menuselect options and I could successfully get : > > CLI> core show application VoiceMailMain > > -= Info about application 'VoiceMailMain' =- > ... > > > What are the necessary steps to have app_voicemail, app_voicemail_imap and > app_voicemail_odbc available along with VoiceMailMain when still using > app_voicemail as preferred Voicemail module ? > > Best regards > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 17.3: No VoiceMailMain when enabling IMAP and ODBC
Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc menuselect.makeopts; done ... I've got this: CLI> core show application VoiceMailMain Your application(s) is (are) not registered Command 'core show application VoiceMailMain' failed. CLI> module reload app_voicemail.so Module 'app_voicemail.so' reloaded successfully. -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System)) CLI> module show like app_voicemail.so Module Description Use Count Status Support Level app_voicemail.so Comedian Mail (Voicemail System) 0 Running core 1 modules loaded Then I re-compiled from source removing both app_voicemail_imap and app_voicemail_odbc menuselect options and I could successfully get : CLI> core show application VoiceMailMain -= Info about application 'VoiceMailMain' =- ... What are the necessary steps to have app_voicemail, app_voicemail_imap and app_voicemail_odbc available along with VoiceMailMain when still using app_voicemail as preferred Voicemail module ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] install_prereq install-unpackaged fails on Debian Buster
Hello, For the very first time, I tried the command bellow on a newly build Debian Buster box on which I successfully built Asterisk 17.2.0 before. I got : # contrib/scripts/install_prereq install-unpackaged *** Installing NBS (Network Broadcast Sound) *** Anbs-trunk/LICENSE Anbs-trunk/nbsclient.c Anbs-trunk/nbsd.c Anbs-trunk/nbs.h Anbs-trunk/nbscat.c Anbs-trunk/Makefile Anbs-trunk/nbscat8k Anbs-trunk/xmms-nbs-1.2.10.patch Anbs-trunk/xmms-nbs-orig.patch Révision 15 extraite. cc -g -Wall -O2 -D_NBS_PRIVATE -Wmissing-prototypes -Werror -Wno-pointer-sign -D_REENTRANT -c -o nbsd.o nbsd.c nbsd.c: In function ‘handle_network’: nbsd.c:453:5: error: ‘strncpy’ output may be truncated copying 79 bytes from a string of length 79 [-Werror=stringop-truncation] strncpy(ns->name, si->streamname, sizeof(ns->name) - 1); ^~~ nbsd.c:454:5: error: ‘strncpy’ output may be truncated copying 15 bytes from a string of length 15 [-Werror=stringop-truncation] strncpy(ns->app, si->appname, sizeof(ns->app) - 1); ^~ In function ‘audio_acquire’, inlined from ‘merge_stream’ at nbsd.c:276:9, inlined from ‘timing_ready’ at nbsd.c:566:10: nbsd.c:221:4: error: ‘strncpy’ output may be truncated copying 79 bytes from a string of length 79 [-Werror=stringop-truncation] strncpy(ih.i.streamname, ns->name, sizeof(ih.i.streamname) - 1); ^~~ nbsd.c:222:4: error: ‘strncpy’ output may be truncated copying 15 bytes from a string of length 15 [-Werror=stringop-truncation] strncpy(ih.i.appname, ns->app, sizeof(ih.i.appname) - 1); ^~~~ cc1: all warnings being treated as errors make: *** [: nbsd.o] Error 1 Thoughts ? Suggestions ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Resource List Subscriptions: how to remove an item from another list ?
Hi, As mentioned in [1], a common pattern is to let everyone monitor everyone except oneself. How do implement this ? Is there something like this: [alice_list] list_item = full_list list_exclude_item = alice [bob_list] list_item = full_list list_exclude_item = bob If negative, would it complex to add to current Asterisk ? [1] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158 Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Example of ${CHANNEL(contact)} output ?
Hello,
My Asterisk 16.2 instance (Debian Buster package) has:
same = n,Verbose(0,CHANNEL is ${CHANNEL})
same = n,Verbose(0,CHANNEL(accountcode) is ${CHANNEL(accountcode)})
same = n,Verbose(0,CHANNEL(contact) is ${CHANNEL(contact)})
same = n,Verbose(0,CHANNEL(endpoint) is ${CHANNEL(endpoint)})
and prints:
CHANNEL is PJSIP/9150-0016
CHANNEL(accountcode) is GENERAL
CHANNEL(contact) is
CHANNEL(endpoint) is 9150
In my testing, ${CHANNEL(contact)} is always empty.
1. Can someone show me the output of a successful CHANNEL(contact) ?
2. Suppose Alice and Bob phones are both registered as extension 1000, what
is the most efficient way to remove Alice's contact from
${PJSIP_DIAL_CONTACTS(1000)} value if Alice ever dials 1000 (and hopes to
ring Bob's phone only) ?
Best regards
--
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[asterisk-users] [SOLVED] Re: What does "all 3 app_voicemail variants can now be built" implies exactly ?
How could I miss this blog post ? Thank you very much, Ben, for replying ! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What does "all 3 app_voicemail variants can now be built" implies exactly ?
Hello, >From Astricon 2019 notes [1], you can read "[a]ll 3 app_voicemail variants can now be built". What does it mean ? Is this change tied with a specific Asterisk version ? Is possible to change from ODBC to IMAP without re-compilation ? Is it also possible to mix mailbox types on a single system ? [1] https://wiki.asterisk.org/wiki/display/AST/Agenda+2019 Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Solved: Re: Asterisk 13.18.3 PJSIP. Wrong Port in Contact Header in Reply to REGISTER?
Hello, Days ago, I banged on a similar issue with Debian Buster's asterisk (16.2): my box had two interfaces (one North and one South) both with private addresses when relaying calls from South to North, my box used South Address for media handling. Upgrading to 16.7.0 without changing configuration, immediately solved the issue. I didn't have time to dig into this (ie reading Changelogs or debugging step by step). I don't think my config was perfect either but if I had to implement anything on Debian Buster again, I'll prepare myself to spend some times on this. My 0.01 cent Le lun. 13 janv. 2020 à 15:51, Benoit Panizzon a écrit : > Hi Joshua > > Thank you for your reply. > > Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via > PPA. Problem persisted. > > Well, I already mentioned that this is a machine with two physical > interfaces with different routes which on the 'external' side handles > SIP customer registrations and has an 'internal' IC Trunk to a > commercial Voice Switch via private IP Range. > > I had the problem, that some of the packets sent out on the 'external' > side contained 'private' IP addresses in either signaling or SDP. > > So I threw all options I could find into the config to bind > transports, endpoints, media and so on to the corresponding interface ip > address. First this looked good. I had the correct IP in every header > and SDP I expected. > > well setting under transport: > > external_signaling_address=[local IP] > > Assuming this is the own interface IP that would be told to external > endpoints was obviously wrong. > > In the end this caused the Proxy-SBC to believe it was not getting an > OK to it's forwarded registration and discard this session. > > Mit freundlichen Grüssen > > -Benoît Panizzon- > -- > I m p r o W a r e A G-Leiter Commerce Kunden > __ > > Zurlindenstrasse 29 Tel +41 61 826 93 00 > CH-4133 PrattelnFax +41 61 826 93 01 > Schweiz Web http://www.imp.ch > __ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS/SSL error loading cert file. [Almost SOLVED]
Hello, Le lun. 6 janv. 2020 à 19:01, Olivier a écrit : > May I add I could successfully (if pjsip show transports has any meaning) > add a PJSIP TLS-transport with: > > [transport-tls] > type=transport > protocol=tls > bind=0.0.0.0:5061 > cert_file=/etc/asterisk/keys/asterisk.crt > priv_key_file=/etc/asterisk/keys/asterisk.key > method=tlsv1 > > Le lun. 6 janv. 2020 à 18:33, Olivier a écrit : > >> Hello, >> >> On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a >> way to enable HTTPS. >> Asterisk is running as asterisk:asterisk: >> >> asterisk 11097 0.3 6.7 741352 67984 ?Ssl 17:53 0:06 >> /usr/sbin/asterisk -g -f -p -U asterisk >> >> # cat /etc/asterisk/http.conf >> [general] >> servername=Asterisk >> enabled=yes >> bindaddr=0.0.0.0 >> bindport=8088 >> tlsenable=yes >> tlsbindaddr=0.0.0.0:8089 >> tlscertfile=/etc/asterisk/keys/asterisk.pem >> ;tlsprivatekey=keys/asterisk.key >> >> # ls -lR /etc/asterisk/keys >> /etc/asterisk/keys: >> total 32 >> -rw-rw-r-- 1 asterisk asterisk 1229 janv. 6 16:00 asterisk.crt >> -rw-rw-r-- 1 asterisk asterisk 586 janv. 6 15:59 asterisk.csr >> -rw-rw-r-- 1 asterisk asterisk 887 janv. 6 15:59 asterisk.key >> -rw-rw-r-- 1 asterisk asterisk 2116 janv. 6 16:00 asterisk.pem >> -rw-rw-r-- 1 asterisk asterisk 158 janv. 6 15:59 ca.cfg >> -rw-rw-r-- 1 asterisk asterisk 1773 janv. 6 15:59 ca.crt >> -rw-rw-r-- 1 asterisk asterisk 3311 janv. 6 15:59 ca.key >> -rw-rw-r-- 1 asterisk asterisk 132 janv. 6 15:59 tmp.cfg >> >> # grep TLS /var/log/asterisk/full | tail -1 >> [Jan 6 18:24:45] ERROR[11221] tcptls.c: TLS/SSL error loading cert file. >> >> >> # su - asterisk --shell /bin/sh --command 'cat >> /etc/asterisk/keys/asterisk.pem' >> -BEGIN RSA PRIVATE KEY- >> MIICXAIBAAKBgQCxllxfOR9sFwyKiKPZErUcBF1zlwTVZ9XvemA/8yQY7aIVw2ce >> ... >> RE3X5iJqFIRupoIQZQJBAJnDX8dCQbqLvmAV6/Ubiz0XHjHzLEkhMKtF/ksbgou1 >> zykmu2rlUbnZ+DPFj/lw9WH7DaIxtogZ7qKSp0dd95g= >> -END RSA PRIVATE KEY- >> -BEGIN CERTIFICATE- >> MIIDXzCCAUcCAQEwDQYJKoZIhvcNAQELBQAwNTEcMBoGA1UEAwwTQXN0ZXJpc2sg >> ... >> XkVjfneCBgllQhLrnb9oUBuHQCy3qtlPkXpXfAtIsodnoV1mrpI3+iKH7xWc4AtQ >> Rbrt >> -END CERTIFICATE- >> >> >> Any clue ? >> >> Best regards >> > After tens of trying different settings, I tried this morning to simply copy certs files from a running FreePBX 15 instance to my Debian Buster target. To my surprise, it worked as for the very first time, I now have : # asterisk -rx 'http show status' HTTP Server Status: Prefix: Server: Asterisk/16.7.0 Server Enabled and Bound to [::]:8088 HTTPS Server Enabled and Bound to [::]:8089 Now, to fully solve the issue, I need to understand why things didn't work previously and now do work correctly. Current /etc/asterisk/keys is: # ls -alR keys keys: total 56 drwxr-xr-x 3 asterisk asterisk 4096 janv. 8 09:31 . drwxrwxr-x 3 asterisk asterisk 4096 janv. 8 09:35 .. -rw--- 1 asterisk asterisk 1675 janv. 8 09:31 api_oauth.key -rw--- 1 asterisk asterisk 451 janv. 8 09:31 api_oauth_public.key -rw-r--r-- 1 asterisk asterisk 191 janv. 8 09:31 ca.cfg -rw-r--r-- 1 asterisk asterisk 1724 janv. 8 09:31 ca.crt -rw-r--r-- 1 asterisk asterisk 3243 janv. 8 09:31 ca.key -rw--- 1 asterisk asterisk 1712 janv. 8 09:31 default.crt -rw--- 1 asterisk asterisk 1610 janv. 8 09:31 default.csr -rw--- 1 asterisk asterisk 3247 janv. 8 09:31 default.key -rw--- 1 asterisk asterisk 4959 janv. 8 09:31 default.pem drwxr-xr-x 2 asterisk asterisk 4096 janv. 8 09:31 integration -rw-r--r-- 1 asterisk asterisk 1024 janv. 8 09:31 .rnd keys/integration: total 24 drwxr-xr-x 2 asterisk asterisk 4096 janv. 8 09:31 . drwxr-xr-x 3 asterisk asterisk 4096 janv. 8 09:31 .. -rw--- 1 asterisk asterisk 4959 janv. 8 09:31 certificate.pem -rw--- 1 asterisk asterisk 1712 janv. 8 09:31 webserver.crt -rw--- 1 asterisk asterisk 3247 janv. 8 09:31 webserver.key Asterisk is running as asterisk:asterisk. /etc/asterisk/http.conf is: # cat http.conf [general] enabled=yes enablestatic=no bindaddr=:: bindport=8088 prefix= sessionlimit=100 session_inactivity=3 session_keep_alive=15000 tlsenable=yes tlsbindaddr=[::]:8089 tlscertfile=/etc/asterisk/keys/integration/certificate.pem tlsprivatekey=/etc/asterisk/keys/integration/webserver.key # cat /etc/asterisk/keys/ca.cfg [req] distinguished_name = req_distinguished_name prompt = no default_md = sha256 [ca] default_md = sha256 [req_distinguished_name] CN=localhost O=localhost [ext] basicConstraints=CA:TRUE Is there a way to find how FreePBX generated the /etc/asterisk/keys tree ? Best re
[asterisk-users] Improve Wiki's "WebRTC config" page
Hello, Reading [1], I would be happy to discuss here, the changes bellow. 1. In "Create certificate" section, instead of 'ls -w 1 /etc/asterisk/keys', could a 'ls -l /etc/asterisk/keys' be used ? This would help to check file permissions. If possible, having those file permissions shown when Asterisk is run as asterisk:asterisk would be very helpful. 2. Instead of a generic tlscertfile= tlsprivatekey= could a specific writing be preferred tlscertfile=/etc/asterisk/keys/asterisk.crt tlsprivatekey=/etc/asterisk/keys/asterisk.key This would be consistent with the "We'll use the asterisk.crt, asterisk.key and ca.crt" text, a couple of lines above. 3. If I'm not mistaken, http.conf file refers to two different files. How exactly can asterisk.crt, asterisk.key and ca.crt be "later used" ? Should the "We'll use the asterisk.crt, asterisk.key and ca.crt ..." be rephrased ? [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients Thoughts ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS/SSL error loading cert file.
May I add I could successfully (if pjsip show transports has any meaning) add a PJSIP TLS-transport with: [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 Le lun. 6 janv. 2020 à 18:33, Olivier a écrit : > Hello, > > On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a > way to enable HTTPS. > Asterisk is running as asterisk:asterisk: > > asterisk 11097 0.3 6.7 741352 67984 ?Ssl 17:53 0:06 > /usr/sbin/asterisk -g -f -p -U asterisk > > # cat /etc/asterisk/http.conf > [general] > servername=Asterisk > enabled=yes > bindaddr=0.0.0.0 > bindport=8088 > tlsenable=yes > tlsbindaddr=0.0.0.0:8089 > tlscertfile=/etc/asterisk/keys/asterisk.pem > ;tlsprivatekey=keys/asterisk.key > > # ls -lR /etc/asterisk/keys > /etc/asterisk/keys: > total 32 > -rw-rw-r-- 1 asterisk asterisk 1229 janv. 6 16:00 asterisk.crt > -rw-rw-r-- 1 asterisk asterisk 586 janv. 6 15:59 asterisk.csr > -rw-rw-r-- 1 asterisk asterisk 887 janv. 6 15:59 asterisk.key > -rw-rw-r-- 1 asterisk asterisk 2116 janv. 6 16:00 asterisk.pem > -rw-rw-r-- 1 asterisk asterisk 158 janv. 6 15:59 ca.cfg > -rw-rw-r-- 1 asterisk asterisk 1773 janv. 6 15:59 ca.crt > -rw-rw-r-- 1 asterisk asterisk 3311 janv. 6 15:59 ca.key > -rw-rw-r-- 1 asterisk asterisk 132 janv. 6 15:59 tmp.cfg > > # grep TLS /var/log/asterisk/full | tail -1 > [Jan 6 18:24:45] ERROR[11221] tcptls.c: TLS/SSL error loading cert file. > > > # su - asterisk --shell /bin/sh --command 'cat > /etc/asterisk/keys/asterisk.pem' > -BEGIN RSA PRIVATE KEY- > MIICXAIBAAKBgQCxllxfOR9sFwyKiKPZErUcBF1zlwTVZ9XvemA/8yQY7aIVw2ce > ... > RE3X5iJqFIRupoIQZQJBAJnDX8dCQbqLvmAV6/Ubiz0XHjHzLEkhMKtF/ksbgou1 > zykmu2rlUbnZ+DPFj/lw9WH7DaIxtogZ7qKSp0dd95g= > -END RSA PRIVATE KEY- > -BEGIN CERTIFICATE- > MIIDXzCCAUcCAQEwDQYJKoZIhvcNAQELBQAwNTEcMBoGA1UEAwwTQXN0ZXJpc2sg > ... > XkVjfneCBgllQhLrnb9oUBuHQCy3qtlPkXpXfAtIsodnoV1mrpI3+iKH7xWc4AtQ > Rbrt > -END CERTIFICATE- > > > Any clue ? > > Best regards > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TLS/SSL error loading cert file.
Hello, On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a way to enable HTTPS. Asterisk is running as asterisk:asterisk: asterisk 11097 0.3 6.7 741352 67984 ?Ssl 17:53 0:06 /usr/sbin/asterisk -g -f -p -U asterisk # cat /etc/asterisk/http.conf [general] servername=Asterisk enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/keys/asterisk.pem ;tlsprivatekey=keys/asterisk.key # ls -lR /etc/asterisk/keys /etc/asterisk/keys: total 32 -rw-rw-r-- 1 asterisk asterisk 1229 janv. 6 16:00 asterisk.crt -rw-rw-r-- 1 asterisk asterisk 586 janv. 6 15:59 asterisk.csr -rw-rw-r-- 1 asterisk asterisk 887 janv. 6 15:59 asterisk.key -rw-rw-r-- 1 asterisk asterisk 2116 janv. 6 16:00 asterisk.pem -rw-rw-r-- 1 asterisk asterisk 158 janv. 6 15:59 ca.cfg -rw-rw-r-- 1 asterisk asterisk 1773 janv. 6 15:59 ca.crt -rw-rw-r-- 1 asterisk asterisk 3311 janv. 6 15:59 ca.key -rw-rw-r-- 1 asterisk asterisk 132 janv. 6 15:59 tmp.cfg # grep TLS /var/log/asterisk/full | tail -1 [Jan 6 18:24:45] ERROR[11221] tcptls.c: TLS/SSL error loading cert file. # su - asterisk --shell /bin/sh --command 'cat /etc/asterisk/keys/asterisk.pem' -BEGIN RSA PRIVATE KEY- MIICXAIBAAKBgQCxllxfOR9sFwyKiKPZErUcBF1zlwTVZ9XvemA/8yQY7aIVw2ce ... RE3X5iJqFIRupoIQZQJBAJnDX8dCQbqLvmAV6/Ubiz0XHjHzLEkhMKtF/ksbgou1 zykmu2rlUbnZ+DPFj/lw9WH7DaIxtogZ7qKSp0dd95g= -END RSA PRIVATE KEY- -BEGIN CERTIFICATE- MIIDXzCCAUcCAQEwDQYJKoZIhvcNAQELBQAwNTEcMBoGA1UEAwwTQXN0ZXJpc2sg ... XkVjfneCBgllQhLrnb9oUBuHQCy3qtlPkXpXfAtIsodnoV1mrpI3+iKH7xWc4AtQ Rbrt -END CERTIFICATE- Any clue ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is PJSIP equivalent of users.conf hassip setting ?
Hello, In /etc/asterisk/users.conf, you can set hassip=yes to declare a chansip entity. Is there any equivalent for PJSIP ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bitwise operations in dialplan ?
Hello, I'm thinking about using a single (long) integer value as a compact way to store user privileges. Do bitwise operations as in [1], exist in Asterisk's dialplan ? Any workaround ? [1] https://wiki.python.org/moin/BitwiseOperators Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Experiences with Jitsi's jigasi
Hello, I've just discovered jigasi : "a server-side application acting as a gateway to Jitsi Meet conferences. Currently allows regular SIP clients to join meetings and provides transcription capabilities" Have someone used it with Asterisk ? How does it work ? [1] https://github.com/jitsi/jigasi Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Low cost routing
Hello, I need some advice: I use 2 different suppliers of trunk SIP in my infrastructure, both send me regularly prices in a .csv format. So I have two SQL tables that contain the prefix and the tariff. For now, I generate a dialplan with a Perl script that allows me to select the prefix trunk to use but the problem is that I change it manually in some cases. For example Trunk A: +3550.1698€ +35521150 0.12815€ +35521151 0.12815€ Trunk B: +3550.1144€ Currently my script sees that +35521150 exists in Trunk A and will therefore use it while on the Trunk B it is less expensive but the prefix +35521150 is not directly indicated since it is covered by the +355 I would like to change that, generate a unique table of prefixes. Someone would have done that already? Thank you, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On Register, run a script, validate source IP
Hello,
Have you tried with ACL (acl.conf) ?
Cheers
Le lun. 18 nov. 2019 à 13:22, Benoit Panizzon a
écrit :
> Hi Gang
>
> To increase security against phished passwords and similar attacks, we
> consider offering customers to define IP ranges (or GeoIP locations)
> from which their dynamic registrations are being accepted.
>
> I can already look at the source IP in the dial plan, so no issue with
> validate an INVITE against a source IP.
>
> But I would also like to prevent registrations from outside of this
> client's specific allowed ip addresses as well, so the line cannot be
> hijacked.
>
> So I'm looking for something like
>
> On Register:
> If check_allowed_ip(auth_username) {
> return;
> } else {
> Reply(403 Wrong IP for this user);
> }
>
> Any ideas how to do that? (Yes, I asked Google and found nothing
> useful yet)
>
> Mit freundlichen Grüssen
>
> -Benoît Panizzon-
> --
> I m p r o W a r e A G-Leiter Commerce Kunden
> __
>
> Zurlindenstrasse 29 Tel +41 61 826 93 00
> CH-4133 PrattelnFax +41 61 826 93 01
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[asterisk-users] Experience with WebRTC on MacOS ?
Hello, Reading [1], I would be very curious to read about WebRTC on MacOS, either for Voice or Voice and Video calls. How does MacOS compare today to Windows or Linux regarding WebRTC support ? Do you need to use Chrome or Firefox to get WebRTC ? [1] https://webkit.org/blog/8672/on-the-road-to-webrtc-1-0-including-vp8/ Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which architecture for WebRTC on a LAN with PSTN access ?
Hello, I would like to offer end users in a LAN, asking for this (why ? I don't know) the capability to use a laptop (along or in replacement of hardphones) to emit and receive PSTN calls. PSTN pass through a plain SIP trunk which does not support video (nor Opus) How can I best integrate off-the-shelf components to implement this ? My initial thoughts are to implement Cyber Mega Phone and asterisk on a single Linux host but I'm having a hard time to evaluate if Cyber Mega Phone is suitable for this and if enduser experience would be positive. What do you think? [1] https://github.com/asterisk/cyber_mega_phone_2k Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WebRTC: which ACL and modes for /etc/asterisk/keys/asterisk.pem ?
Hello, Following [1], you get precious help for webRTC installation. Something that is missing there, though, is a note expliciting /etc/asterisk/keys files ownerships and modes. As people are either running asterisk as root:root, asterisk:root and others or as asterisk:asterisk, the number of files ownerships and modes combinations quickly grows. Can someone who has successfully run Asterisk HTTPS server, be kind enough to simply shows here, what is: - the user:group running Asterisk on its machine - the output of its "ls -al /etc/asterisk/keys" - the content of its /etc/asterisk/http.conf [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients Thanks in advance. Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?
May I add my modules.conf conf (this modules.conf was generated by make basic-pbx) ? Maybe a module is missing there ? Shall I check something in menuselect ? (Replacing its content with a simple "autoload = yes" does not change either) # cat modules.conf [modules] autoload = no ; This is a minimal module load. We are loading only the modules required for ; the Asterisk features used in the Super Awesome Company configuration. ; Applications load = app_bridgewait.so load = app_dial.so load = app_playback.so load = app_stack.sothis load = app_verbose.so load = app_voicemail.so load = app_directory.so load = app_confbridge.so ; Bridging load = bridge_builtin_features.so load = bridge_builtin_interval_features.so load = bridge_holding.so load = bridge_native_rtp.so load = bridge_simple.so load = bridge_softmix.so ; Call Detail Records load = cdr_custom.so ; Channel Drivers load = chan_bridge_media.so load = chan_pjsip.so ; Codecs load = codec_gsm.so load = codec_resample.so load = codec_ulaw.so load = codec_g722.so ; Formats load = format_gsm.so load = format_pcm.so load = format_wav_gsm.so load = format_wav.so ; Functions load = func_callerid.so load = func_cdr.so load = func_pjsip_endpoint.so load = func_sorcery.so load = func_devstate.so load = func_strings.so ; Core/PBX load = pbx_config.so ; Resources load = res_http_websocket.so load = res_musiconhold.so load = res_pjproject.so load = res_pjsip_acl.so load = res_pjsip_authenticator_digest.so load = res_pjsip_caller_id.so load = res_pjsip_dialog_info_body_generator.so load = res_pjsip_diversion.so load = res_pjsip_dtmf_info.so load = res_pjsip_endpoint_identifier_anonymous.so load = res_pjsip_endpoint_identifier_ip.so load = res_pjsip_endpoint_identifier_user.so load = res_pjsip_exten_state.so load = res_pjsip_header_funcs.so load = res_pjsip_logger.so load = res_pjsip_messaging.so load = res_pjsip_mwi_body_generator.so load = res_pjsip_mwi.so load = res_pjsip_nat.so load = res_pjsip_notify.so load = res_pjsip_one_touch_record_info.so load = res_pjsip_outbound_authenticator_digest.so load = res_pjsip_outbound_publish.so load = res_pjsip_outbound_registration.so load = res_pjsip_path.so load = res_pjsip_pidf_body_generator.so load = res_pjsip_pidf_digium_body_supplement.so load = res_pjsip_pidf_eyebeam_body_supplement.so load = res_pjsip_publish_asterisk.so load = res_pjsip_pubsub.so load = res_pjsip_refer.so load = res_pjsip_registrar.so load = res_pjsip_rfc3326.so load = res_pjsip_sdp_rtp.so load = res_pjsip_send_to_voicemail.so load = res_pjsip_session.so load = res_pjsip.so load = res_pjsip_t38.so load = res_pjsip_transport_websocket.so load = res_pjsip_xpidf_body_generator.so load = res_rtp_asterisk.so load = res_sorcery_astdb.so load = res_sorcery_config.so load = res_sorcery_memory.so load = res_sorcery_realtime.so load = res_timing_timerfd.so ; Don't load res_hep.so and kin unless you are using hep monitoring in your network noload = res_hep.so noload = res_hep_pjsip.so noload = res_hep_rtcp.so Le lun. 18 nov. 2019 à 22:18, Olivier a écrit : > Unfortunately, changing ownership did not solve the issue: > > # ls -al keys/ > total 40 > drwxr-xr-x 2 asterisk asterisk 4096 nov. 18 20:47 . > drwxr-x--- 3 asterisk asterisk 4096 nov. 18 20:53 .. > -rw--- 1 asterisk asterisk 1224 nov. 18 20:47 asterisk.crt > -rw--- 1 asterisk asterisk 578 nov. 18 20:46 asterisk.csr > -rw--- 1 asterisk asterisk 887 nov. 18 20:46 asterisk.key > -rw--- 1 asterisk asterisk 2111 nov. 18 20:47 asterisk.pem > -rw--- 1 asterisk asterisk 161 nov. 18 20:46 ca.cfg > -rw--- 1 asterisk asterisk 1781 nov. 18 20:46 ca.crt > -rw--- 1 asterisk asterisk 3311 nov. 18 20:46 ca.key > -rw--- 1 asterisk asterisk 124 nov. 18 20:46 tmp.cfg > > # service asterisk stop > # service asterisk start > # asterisk -rx "http show status" > HTTP Server Status: > Prefix: > Server: Asterisk/17.0.0 > Server Enabled and Bound to 0.0.0.0:8088 > > Enabled URI's: > /httpstatus => Asterisk HTTP General Status > /static/... => Asterisk HTTP Static Delivery > > Enabled Redirects: > > Le lun. 18 nov. 2019 à 22:08, Richard Mudgett a > écrit : > >> >> >> On Mon, Nov 18, 2019 at 2:53 PM Olivier wrote: >> >>> Hello, >>> >>> I've installed a new Asterisk 17.0.0 on a Debian Buster system. >>> >>> This Asterisk instance is run by asterisk user (and group). >>> I've got: >>> >>> # ls -l /etc/asterisk >>> total 68 >>> -rw-r--r-- 1 asterisk asterisk 501 nov. 18 19:12 asterisk.conf >>> -rw-r--r-- 1 asterisk asterisk 135 nov. 18 18:57 cdr.conf >>> -rw-r--r-- 1 asterisk asterisk 684 nov. 18 18:57 cdr_custom.conf >>> -rw-r--r-- 1 asterisk asterisk 103 nov. 18 18:57 confbridge.conf >>>
Re: [asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?
Unfortunately, changing ownership did not solve the issue: # ls -al keys/ total 40 drwxr-xr-x 2 asterisk asterisk 4096 nov. 18 20:47 . drwxr-x--- 3 asterisk asterisk 4096 nov. 18 20:53 .. -rw--- 1 asterisk asterisk 1224 nov. 18 20:47 asterisk.crt -rw--- 1 asterisk asterisk 578 nov. 18 20:46 asterisk.csr -rw--- 1 asterisk asterisk 887 nov. 18 20:46 asterisk.key -rw--- 1 asterisk asterisk 2111 nov. 18 20:47 asterisk.pem -rw--- 1 asterisk asterisk 161 nov. 18 20:46 ca.cfg -rw--- 1 asterisk asterisk 1781 nov. 18 20:46 ca.crt -rw--- 1 asterisk asterisk 3311 nov. 18 20:46 ca.key -rw--- 1 asterisk asterisk 124 nov. 18 20:46 tmp.cfg # service asterisk stop # service asterisk start # asterisk -rx "http show status" HTTP Server Status: Prefix: Server: Asterisk/17.0.0 Server Enabled and Bound to 0.0.0.0:8088 Enabled URI's: /httpstatus => Asterisk HTTP General Status /static/... => Asterisk HTTP Static Delivery Enabled Redirects: Le lun. 18 nov. 2019 à 22:08, Richard Mudgett a écrit : > > > On Mon, Nov 18, 2019 at 2:53 PM Olivier wrote: > >> Hello, >> >> I've installed a new Asterisk 17.0.0 on a Debian Buster system. >> >> This Asterisk instance is run by asterisk user (and group). >> I've got: >> >> # ls -l /etc/asterisk >> total 68 >> -rw-r--r-- 1 asterisk asterisk 501 nov. 18 19:12 asterisk.conf >> -rw-r--r-- 1 asterisk asterisk 135 nov. 18 18:57 cdr.conf >> -rw-r--r-- 1 asterisk asterisk 684 nov. 18 18:57 cdr_custom.conf >> -rw-r--r-- 1 asterisk asterisk 103 nov. 18 18:57 confbridge.conf >> -rw-r--r-- 1 asterisk asterisk 6834 nov. 18 18:57 extensions.conf >> -rw-r--r-- 1 asterisk asterisk 138 nov. 18 21:42 http.conf >> -rw-r--r-- 1 asterisk asterisk 681 nov. 18 18:57 indications.conf >> drwxr-xr-x 2 root root 4096 nov. 18 20:47 keys >> -rw-r--r-- 1 asterisk asterisk 160 nov. 18 18:57 logger.conf >> -rw-r--r-- 1 asterisk asterisk 2769 nov. 18 18:57 modules.conf >> -rw-r--r-- 1 asterisk asterisk 50 nov. 18 18:57 musiconhold.conf >> -rw-r--r-- 1 asterisk asterisk 6360 nov. 18 18:57 pjsip.conf >> -rw-r--r-- 1 asterisk asterisk 790 nov. 18 18:57 pjsip_notify.conf >> -rw-r--r-- 1 asterisk asterisk 768 nov. 18 18:57 README >> -rw-r--r-- 1 asterisk asterisk 513 nov. 18 18:57 voicemail.conf >> >> # ls -l /etc/asterisk/keys/ >> total 32 >> -rw--- 1 root root 1224 nov. 18 20:47 asterisk.crt >> -rw--- 1 root root 578 nov. 18 20:46 asterisk.csr >> -rw--- 1 root root 887 nov. 18 20:46 asterisk.key >> -rw--- 1 root root 2111 nov. 18 20:47 asterisk.pem >> > > I'd say that asterisk running as the asterisk user has no permission to > see the .pem file as only root can see it. > > Richard > > >> -rw--- 1 root root 161 nov. 18 20:46 ca.cfg >> -rw--- 1 root root 1781 nov. 18 20:46 ca.crt >> -rw--- 1 root root 3311 nov. 18 20:46 ca.key >> -rw--- 1 root root 124 nov. 18 20:46 tmp.cfg >> >> # cat /etc/asterisk/http.conf >> [general] >> enabled=yes >> bindaddr=0.0.0.0 >> bindport=8088 >> tlsenable=yes >> tlsbindaddr=0.0.0.0:8089 >> tlscertfile=/etc/asterisk/keys/asterisk.pem >> >> But, still I don't have any HTTPS server running: >> >> # asterisk -rx "http show status" >> HTTP Server Status: >> Prefix: >> Server: Asterisk/17.0.0 >> Server Enabled and Bound to 0.0.0.0:8088 >> >> Enabled URI's: >> /httpstatus => Asterisk HTTP General Status >> /static/... => Asterisk HTTP Static Delivery >> >> Enabled Redirects: >> None. >> >> >> >> Can someone help me on this ? >> Is http;conf correct ? >> Am I mis-configuring files access rights or ownership ? >> Something else ? >> >> Best regards >> >> >> >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> 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 -- > > Check out the new Asterisk community forum at: > https://community.asteris
Re: [asterisk-users] Question on WebRTC configuration
Hello, Reading this old thread, isn't there also an error in [1] as It also mentions a tlscafile setting. Cheers [1] https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone Le ven. 7 déc. 2018 à 16:41, Kevin Harwell a écrit : > On Fri, Dec 7, 2018 at 9:11 AM Dan Cropp wrote: > >> In the asterisk wiki instructions for Configuring Asterisk for WebRTC >> clients… >> >> >> >> >> https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients >> >> >> >> “To communicate with websocket clients, Asterisk uses its built-in HTTP >> daemon. Configure */etc/asterisk/http.conf* as follows: >> >> >> >> [general] >> >> enabled=yes >> >> bindaddr=0.0.0.0 >> >> bindport=8088 >> >> tlsenable=yes >> >> tlsbindaddr=0.0.0.0:8089 >> >> tlscertfile= >> >> tlsprivatekey= >> >> tlscafile=” >> >> >> >> What is the tlscafile setting? >> >> >> >> When I look at the http.conf samples it doesn’t mention the tlscafile >> setting. >> >> I see there is a tlscafile setting in sip.conf, but I don’t find this >> anywhere else. >> >> >> >> Is the wiki web page mistaken or is this an actual http.conf setting that >> is undocumented? >> > > The page is mistaken. It should not be there. the 'tlscafile' option is > not supported by the Asterisk http server. I've removed it from the wiki. > Thanks for catching that! > > >> >> >> Have a great day! >> > > You too! > > >> Dan >> -- >> > > -- > Kevin Harwell > Digium - A Sangoma Company | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: https://digium.com & https://asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?
Hello, I've installed a new Asterisk 17.0.0 on a Debian Buster system. This Asterisk instance is run by asterisk user (and group). I've got: # ls -l /etc/asterisk total 68 -rw-r--r-- 1 asterisk asterisk 501 nov. 18 19:12 asterisk.conf -rw-r--r-- 1 asterisk asterisk 135 nov. 18 18:57 cdr.conf -rw-r--r-- 1 asterisk asterisk 684 nov. 18 18:57 cdr_custom.conf -rw-r--r-- 1 asterisk asterisk 103 nov. 18 18:57 confbridge.conf -rw-r--r-- 1 asterisk asterisk 6834 nov. 18 18:57 extensions.conf -rw-r--r-- 1 asterisk asterisk 138 nov. 18 21:42 http.conf -rw-r--r-- 1 asterisk asterisk 681 nov. 18 18:57 indications.conf drwxr-xr-x 2 root root 4096 nov. 18 20:47 keys -rw-r--r-- 1 asterisk asterisk 160 nov. 18 18:57 logger.conf -rw-r--r-- 1 asterisk asterisk 2769 nov. 18 18:57 modules.conf -rw-r--r-- 1 asterisk asterisk 50 nov. 18 18:57 musiconhold.conf -rw-r--r-- 1 asterisk asterisk 6360 nov. 18 18:57 pjsip.conf -rw-r--r-- 1 asterisk asterisk 790 nov. 18 18:57 pjsip_notify.conf -rw-r--r-- 1 asterisk asterisk 768 nov. 18 18:57 README -rw-r--r-- 1 asterisk asterisk 513 nov. 18 18:57 voicemail.conf # ls -l /etc/asterisk/keys/ total 32 -rw--- 1 root root 1224 nov. 18 20:47 asterisk.crt -rw--- 1 root root 578 nov. 18 20:46 asterisk.csr -rw--- 1 root root 887 nov. 18 20:46 asterisk.key -rw--- 1 root root 2111 nov. 18 20:47 asterisk.pem -rw--- 1 root root 161 nov. 18 20:46 ca.cfg -rw--- 1 root root 1781 nov. 18 20:46 ca.crt -rw--- 1 root root 3311 nov. 18 20:46 ca.key -rw--- 1 root root 124 nov. 18 20:46 tmp.cfg # cat /etc/asterisk/http.conf [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/keys/asterisk.pem But, still I don't have any HTTPS server running: # asterisk -rx "http show status" HTTP Server Status: Prefix: Server: Asterisk/17.0.0 Server Enabled and Bound to 0.0.0.0:8088 Enabled URI's: /httpstatus => Asterisk HTTP General Status /static/... => Asterisk HTTP Static Delivery Enabled Redirects: None. Can someone help me on this ? Is http;conf correct ? Am I mis-configuring files access rights or ownership ? Something else ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is Debian Buster's asterisk equivalent of Digium's contrib/scripts/ast_tls_cert ?
Hello, With Debian Buster's asterisk package, what can you use instead of Digium's contrib/scripts/ast_tls_cert ? If that matters, this is for using WebRTC and Cyber Mega Phone 2K (both on the same box) in a private LAN environment. My intent was to use easy-rsa package but I wouldn't mind dropping it from something more efficient. Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segfault in libpjnath.so.2 though PJSIP not present in dialplan
Le lun. 26 août 2019 à 14:21, Joshua C. Colp a écrit : > On Mon, Aug 26, 2019, at 9:00 AM, Olivier wrote: > > > > > > Le lun. 26 août 2019 à 12:07, Joshua C. Colp a écrit > : > > > ... > > > > > > libpjnath is the ICE/STUN/TURN library which is used by > res_rtp_asterisk for that functionality. If you're using WebRTC or > ICE/STUN/TURN, then you would be using that library. > > > > Yes, I'm using ICE/STUN/TURN. > > That explains libpjnath usage. > > > > Thank you sharing this here. > > > > Now finding why using ICE/STUN/TURN triggered a segfault is another > story ... > > There have been fixes in race conditions with STUN in pjnath since then, > it could be one of those. > Yes probably. I'll try to upgrade to Asterisk 16, before digging deeper, if necessary. > > -- > Joshua C. Colp > Digium - A Sangoma Company | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segfault in libpjnath.so.2 though PJSIP not present in dialplan
Le lun. 26 août 2019 à 12:07, Joshua C. Colp a écrit : > ... > > libpjnath is the ICE/STUN/TURN library which is used by res_rtp_asterisk > for that functionality. If you're using WebRTC or ICE/STUN/TURN, then you > would be using that library. > Yes, I'm using ICE/STUN/TURN. That explains libpjnath usage. Thank you sharing this here. Now finding why using ICE/STUN/TURN triggered a segfault is another story ... Thanks again -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Segfault in libpjnath.so.2 though PJSIP not present in dialplan
Hello, I've got an Asterisk 11.13.1 system running on a Debian Jessie platform. This system's extensions.conf doesn't include any reference to PJSIP, yet (only using chan_sip at the moment). This morning, it failed with: Aug 26 09:07:33 foobar kernel: [6534231.776418] asterisk[9701]: segfault at 3c ip 7f02f5d9a7dd sp 7f02f0b0b260 error 4 in libpjnath.so.2[7f02f5d84000+26000] What possible root cause can trigger such system to run libpjnath.so.2 in the first place ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enhanced Messaging and softphones
Le ven. 18 janv. 2019 à 17:30, Joshua C. Colp a écrit : > > > > > > You mean with a softphone you can't select a single (or several) video > > among those available, can you ? > > Even with DTMF sequence and some features.conf magic, a user cannot ask > > to receive a specific video stream ? > > > > (forgive me if my question seems naive or irrelevant but I didn't dare > > to try or install anything yet) > > This is not currently supported in any video mode in ConfBridge. > So currently, any incoming video flux is forwarded to all ConfBridge participants (except if forwarding is cobntrolled/tuned by an external app) ? > > > > > > From a messaging perspective it has to support the JSON payload that > > Asterisk is using in ConfBridge. This conveys additional information > > that a straight up chat message wouldn't. > > Do you mean Asterisk receives and send SIP messages with embeded JSON > > Payload ? > > Is there a specific name for such behaviour (something that could help > > to check if a SIP library or client supports this) ? > > There's not really a name or anything, except for our content type that we > came up with. It's not defined as a standard or an RFC. Within ConfBridge > the messages are exchanged using JSON between Asterisk and client. > This is now very clear, for me. Thanks again > > -- > Joshua C. Colp > Digium - A Sangoma Company | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enhanced Messaging and softphones
Thanks for your (fast) reply ! Le ven. 18 janv. 2019 à 16:32, Joshua C. Colp a écrit : > On Fri, Jan 18, 2019, at 11:22 AM, Olivier wrote: > > Hello, > > > > I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and > > ConfBridge. > > It seems very interesting addition as it brings the capability to mix > > voice, video and text in conferencing. > > > > On an other hand, there are some softphones (Zoiper, Bria, ...) that > > tout voice, video and chat capability. > > Though Enhanced Messaging solution described in [1] seems more > > attractive to me in the long run, I wonder if it could easier to > > quickly set a demo using softphone instead of a web application. > > > > In this case, was I'm failing to understand is is there anything > > missing in Confbridge for chat conferencing with softphones. > > May I add I read [2] which mentions sending of "custom JSON message". > > > > My questions are: > > > > 1. Do such softphones use compatible SIP MESSAGE to implement chat > feature ? > > If negative, what is the missing block that should be present along > Asterisk ? > > > > 2. Is there a different class of clients (I'm think of Riot clients) as > > opposed to softphones that support Confbridge voice, video and test > > features ? > > I'm not aware of any softphone that supports the functionality. From a > video perspective it has to support multiple video streams and displaying > them (Asterisk doesn't do video transcoding or mixing). You mean with a softphone you can't select a single (or several) video among those available, can you ? Even with DTMF sequence and some features.conf magic, a user cannot ask to receive a specific video stream ? (forgive me if my question seems naive or irrelevant but I didn't dare to try or install anything yet) > From a messaging perspective it has to support the JSON payload that > Asterisk is using in ConfBridge. This conveys additional information that a > straight up chat message wouldn't. > Do you mean Asterisk receives and send SIP messages with embeded JSON Payload ? Is there a specific name for such behaviour (something that could help to check if a SIP library or client supports this) ? > > -- > Joshua C. Colp > Digium - A Sangoma Company | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Enhanced Messaging and softphones
Hello, I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and ConfBridge. It seems very interesting addition as it brings the capability to mix voice, video and text in conferencing. On an other hand, there are some softphones (Zoiper, Bria, ...) that tout voice, video and chat capability. Though Enhanced Messaging solution described in [1] seems more attractive to me in the long run, I wonder if it could easier to quickly set a demo using softphone instead of a web application. In this case, was I'm failing to understand is is there anything missing in Confbridge for chat conferencing with softphones. May I add I read [2] which mentions sending of "custom JSON message". My questions are: 1. Do such softphones use compatible SIP MESSAGE to implement chat feature ? If negative, what is the missing block that should be present along Asterisk ? 2. Is there a different class of clients (I'm think of Riot clients) as opposed to softphones that support Confbridge voice, video and test features ? Best regards [1] https://blogs.asterisk.org/2018/08/29/enrich-your-conference-app-with-asterisk-enhanced-messaging-part-1/ [2] https://community.asterisk.org/t/conference-participant-messaging/77755 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it possible to find real domain names instead of IP in SIP URI ?
Hello, Is it possible to find real domain names instead of IP addresses in SIP URI ? For instance, in a book dedicated to SIP (Understanding the Session Initiation Protocol), I'm reading an example of a SIP INVITE that looks like: INVITE sip:4...@salzburg.at;user=phone SIP/2.0 ... In my testing (with a Yealink phone), I can configure it either with: account.1.sip_server.1.address = ipbx1.foobar.lan account.1.sip_server.1.address = ipbx1 account.1.sip_server.1.address =192.168.1.1 Any of the above works but Asterisk sip set debug command shows something like: <--- SIP read from UDP:192.168.1.42:5060 ---> REGISTER sip:192.168.1.1:5060 SIP/2.0 I would the above last line to be respectively: REGISTER sip:ipbx1.foobar.lan:5060 SIP/2.0 REGISTER sip:ipbx1:5060 SIP/2.0 REGISTER sip:192.168.1.1:5060 SIP/2.0 Any hint ? Best -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Are anonymous international calls allowed ?
Hello, These questions crossed my mind this morning : In general, are anonymous international calls allowed (ie calling from one country to a number in an other country while hiding your own caller id) ? Are there special rules in Europe for this ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR/CEL Radius features
Hello, Page [1] gathers information on how to configure Asterisk CDR Radius backend. I'm not familiar at all with Radius in IP Telephony. 1. Would a Radius database and its associated tools allow live call accounting data displaying of an Asterisk instance powered by such CDR Radius backend ? Would it scale to several Asteruisk instances ? 2. I know Freeradius SimultaneousUse setting. Could this be used to control the total call count of a given accountcode when these calls are processed by several Asterisk boxes ? Best regards [1] https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to build and use your custom asterisk .deb package ?
Hello, There is question that bounces in my mind for quite a long time. Today, I dare to ask it here: how do you package and use your custom asterisk .deb package ? The background is: - I'm now a long time Debian user and I learned to appreciate Debian's deb package benefits specially when dealing with complex softwares such as Asterisk - On another hand, new Asterisk versions are regularly published. Looking at Debian's asterisk source packages, beside patching Asterisk source code, it seems possible to build a new Asterisk 13.X+1 package copying 13.X package - If possible (ie not too hard), I would be happy to build and maintain an Asterisk 16.X binary package for Buster (amd targets). So my questions are: 1. Would you evaluate porting Debian's patches from one Asterisk version to the next one (Asterisk 16.4 to 16.5, for example), to be a complex or time consuming task ? 2. What is the simplest and safest way to deal with the existence of both custom and original packages ? Using deb packages terminology, would you simply create a mycustom-asterisk package which both conflicts and provides asterisk, if that is possible ? Alternatively, is apt-pinning recommended ? 3. Can I realistically hope, that I won't have to change Asterisk 16 dependencies during Asterisk evolution, at least for Asterisk core features or I shall prepare to also upgrade some libraries ? 4. Suggestions ? Pointers ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding Subscribe Handlers in PJSIP
A site question: which of the following RFC would describe as-feature-event ? [1] https://www.iana.org/assignments/sip-events/sip-events.xhtml Le mer. 1 mars 2017 à 21:03, Trey Hilyard a écrit : > Is there any "easy" way to add a custom subscribe handler? I have a set of > users with Polycom phones that attempt to Events that Asterisk/PJSIP > doesn't recognize, "call-info" and "as-feature-event". It just generates a > warning, but it got me wondering if I could add my own handlers for those > that didn't actually do anything but simply responded with a 200 OK. > > Yes, I can probably stop the phones from subscribing, but this is more > academic at this point. I assume there are things that I could do if I > wanted to make changes and recompile, but is there an easier way to add a > handler? I am a little confused about whether there is a subscribe handler > in front of res_pjsip_pubsub, or if that is the first place that a > SUBSCRIBE could get caught. > > -Trey > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding Subscribe Handlers in PJSIP
Hi all, Is there a way with Polycom phones or alternatives, to configure a specific SIP server for such as-feature-event or call-info events ? If positive, maybe a third party SIP server (Kamailio, ...) supporting those events would allow such implementation. Looking at Yealink phone Admin guide, it seems possible to set a Yealink phone to handle such event Notify/Subscribe communicating with a dedicated SIP server. It seems rather "expensive" though. Thoughts ? Best regards Le mer. 1 mars 2017 à 21:21, Joshua Colp a écrit : > On Wed, Mar 1, 2017, at 04:02 PM, Trey Hilyard wrote: > > Is there any "easy" way to add a custom subscribe handler? I have a set > > of > > users with Polycom phones that attempt to Events that Asterisk/PJSIP > > doesn't recognize, "call-info" and "as-feature-event". It just generates > > a > > warning, but it got me wondering if I could add my own handlers for those > > that didn't actually do anything but simply responded with a 200 OK. > > > > Yes, I can probably stop the phones from subscribing, but this is more > > academic at this point. I assume there are things that I could do if I > > wanted to make changes and recompile, but is there an easier way to add a > > handler? I am a little confused about whether there is a subscribe > > handler > > in front of res_pjsip_pubsub, or if that is the first place that a > > SUBSCRIBE could get caught. > > The res_pjsip_pubsub module itself provides a framework for registering > support for event types (ast_sip_register_subscription_handler) and > handles the subscription lifetime. Callbacks are invoked on the various > things. Subscription requested, terminated, etc. There are also > functions for sending a NOTIFY and such. > > There's also another framework for handling PUBLISH. > > This all does require writing a C module though and building it with > Asterisk. There's no external mechanism to implement such things. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can SIP domain help to set multiple SIP trunks between two boxes ?
Hello, I've been asked if it is possible or not to set several (10 or so) SIP trunks between two boxes, one beeing an Avaya IPBX, the other being an Asterisk 13 or 16 box (with either chan_sip or pjsip). The reason behind this question come from billing requirements. I'm not convinced yet setting several trunks is the proper answer to specific billing requirements but my above question remains. Before officially answering the root question, I'm planning to set a demonstration between two Asterisk boxes, leaving trials with an Avaya IPBX for a later step. >From previous trials with chan_sip years ago, the main issues was matching incoming calls to appropriate trunk as calls may come from the same IP/port combination. Now, I read very few lines about SIP domains and wondered if this could be a mean to set several trunks without touching IP/port settings. What do you think of this ? Is it possible for Asterisk to send outbound and receive inbound using SIP domain syntax in SIP messages instead of IP values ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk -rx "cmd" truncates cmd's output
Hello, I've edited my diaplan to print some data on screen with statements like: [foobar] exten = foo,1,Verbose(0,Whatever I need to display) exten = bar,1,Verbose(0,Some more text) When using rasterisk and entering "channel originate Local/foo@foobar application Noop", I can read lines such as: ... Verbose(0,Whatever I need to display) When I type, asterisk -rx "channel originate Local/foo@foobar application Noop", only single line "Setting max files open to 1000" is printed . This "Setting max files open to 1000" line is the one asterisk prints when starting a CLI session How can I work around this ? Comments ? Suggestions ? I'm using Asterisk 13.14.1 from Debian Stable repo on Stretch. Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Desktop Click to dial with Google Contacts
Hello, On a linux desktop, Google Contacts web application ties phone numbers with URL such as https://hangouts.google.com/?action=chat=%2B123456789=fr=0. Have you ever tried to redirect or rewrite such URL and replace with something like https://myasteriskdialer.example.com/?tel=%2B123456789 ? How successful ? How did you proceed ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to force Asterisk to reply with floating IP with chan_sip ?
Hello, I'm setting up a new cluster that must replace several old Asterisk instances. For various reasons, this new cluster must use chan_sip (migration to PJSIP is planned in a later phase). This new cluster uses VRRP in active/passive mode: - at any time, only one cluster member is active, - when a member becomes active, it inherits several floating IP addresses, it changes its IP configuration (with ip rule or ip route statements) and it starts Asterisk. Beside tweaking Linux IP configuration, is there a way to teach Asterisk's chan_sip "to always reply using as IP source, the destination IP it got the SIP request from" ? For instance, if Asterisk's chan_sip listen on IPs IP A, Ip B and IP C, all belonging to the same network, then I want Asterisk to reply with IP A for any request it received through IP A. Looking at ASTDB SIP/Registry instances, Asterisk does not save the IP address it heard an incoming REGISTER with. Did I miss something ? Suggestions ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPp scenario file for testing UAC Authentication with Asterisk ?
Hello, I'm testing an Asterisk instance. At the moment, I'm focusing on its capability to receive and challenge incoming SIP Registrations. For various reasons, I would prefer to use SIPp instead of Asterisk to act as SIP Client. Has someone successfully done this ? If negative, what explains this ? If positive, can you give an example of a successful SIPp scenario file ? I've played with both embeded branchc and [1] but met no success yet Best regards [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How best to run a SIPp test on a remote host
Hello, I'm curently setting a lab environment for load testing an Asterisk instance. This environment includes: - a management workstation where I would like to run scripts and store test reports - a box hosting SIPp - the Asterisk box I'm load testing (System Under Test) - an other Asterisk box that simply anwers and echoes incoming media The SIPp commande I'm currently using looks like: sipp -bg -nostdin -trace_screen -screen_file /home/foobar/reports.d/campaign_123456789_1_1.txt -sf /home/foobar/custom_with_10_minutes_audio.xml 3.4.49.3 -i 3.4.58.206 -s 123456789 -r 1 -m 1 -d 6 Explainations: - Run in background mode (more on that later) - Last SIPp screen with successful/failed call counters saved in a file - SIPp reads a 10 minutes audio file and stops reading after 6 ms (see -d parameter) - Media comes from SIPs, pass though System Under Test and then comes back 1. How can you run such "sipp -bg ..." command from management workstation on remote SIPp box ? I tried using Fabric (I'm discovering) but it always fails with a 99 exit code, as if SIPp/Fabric required a different options set. 2. This project pysipp [1] seems interesting. Has anyone tried it ? How can you get started ? 3. Suggestions ? Best regards [1] https://github.com/SIPp/pysipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to defer SDP in ACK for unit testing purposes
Le mer. 10 oct. 2018 à 12:26, Joshua Colp a écrit : > On Wed, Oct 10, 2018, at 4:27 AM, Olivier wrote: > > Hello, > > > > I think I met a case similar to the one solved by [1] . Quoting this > case : > > > > * res_pjsip: Handle deferred SDP hold/unhold properly. > > > > Some SIP devices indicate hold/unhold using deferred SDP > reinvites. In > > other words, they provide no SDP in the reinvite. > > > > A typical transaction that starts hold might look something like > this: > > > > * Device sends reinvite with no SDP > > * Asterisk sends 200 OK with SDP indicating sendrecv on streams. > > * Device sends ACK with SDP indicating sendonly on streams. > > > > > > Now, I would like to configure an Asterisk instance to act as those SIP > > devices, ie to defer all SDP signaling in ACK. > > > > This is for testing purpose as I would like to reproduce in a lab an > issue > > with those SIP devices. > > > > 1. Is it possible ? I can use any Asterisk version for implementation. > > It is not possible to configure Asterisk for this. The chan_pjsip module > only does normal reinvites with SDP when configured to pass through MOH > signaling. > This is the answser I feared ;-) Thanks for replying. If someone has a clue for alternatives (softphones, hardphones, ...), I'll curious to know > > -- > Joshua Colp > Digium - A Sangoma Company | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to defer SDP in ACK for unit testing purposes
Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating sendrecv on streams. * Device sends ACK with SDP indicating sendonly on streams. Now, I would like to configure an Asterisk instance to act as those SIP devices, ie to defer all SDP signaling in ACK. This is for testing purpose as I would like to reproduce in a lab an issue with those SIP devices. 1. Is it possible ? I can use any Asterisk version for implementation. 2. Alternatively, do you know any softphone "implementing SDP in ACK" ? 3. Alternatively, do you know any SIP hardphone implementing this ? 4. Suggestions ? [1] http://lists.digium.com/pipermail/asterisk-code-review/2016-April/019483.html Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Makefile target to generate asterisk.service file
Hello, Now that systemd is default init system in several Linux distribution, is there a Makefile entry to generate a local systemd asterisk.service file ? Something like "make asterisk-service" just like "make config". Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] First attempt with statsd
Hello, I've just read this [1] blog entry. I'm completely new with statsd. My questions are: 1. This [1] mentions both res_chan_stats and res_endpoint_stats. I can't find any res_chan_stats.so or res_endpoint_stats.so file in my debian Stretch asterisk box. What does it mean ? 2. On a general point of view, is collectd daemon extended with a statsd plugin able to collect Asterisk Statsd statistics ? [1] https://blogs.asterisk.org/2016/02/03/integrating-asterisk-with-statsd/ Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Explain module reloading error message
Hello, On a freshly update Debian Stretch packaged-Asterisk (13.14.1) box, I'm reading this: asterisktuto*CLI> module load res_statsd.so Unable to load module res_statsd.so Command 'module load res_statsd.so' failed. [Oct 9 12:53:26] WARNING[488]: loader.c:1077 load_resource: Module 'res_statsd.so' already exists. [asterisktuto*CLI> module reload res_statsd.so No such module 'res_statsd.so' Can you explain this second message as res_statsd.so module seems to exist ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WebRTC as Softphone substitute ?
@Nasir: Thanks for replying here. Did you met in your deployments, the kind of stability issues Carlos reported earlier ? Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal a écrit : > Hi Olivior, > > We have recently worked on a WebRTC based agent panel. As based on my > experience I think that WebRTC based phones are far better and cheaper then > those soft / sip phone. the big plus is that they are easy to customize and > developer can use the power of browser and web to build / offer features > which are not possible with regular phones. > > Regarding your concern about BLF or call history, for me as a being > developer it is just a matter of customization. > > Regards > > Nasir Iqbal > > ICTBroadcast - an Auto Dialer software for ITSP > <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> > SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns > http://www.ictbroadcast.com/ > > > On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez wrote: > >> On 9/26/18 10:20 AM, Matthew Fredrickson wrote: >> >> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez >> wrote: >> >> On 9/26/2018 4:46 AM, Olivier wrote: >> >> >> >>> Hello, >> >>> >> >>> This morning, I asked myself if WebRTC could be a viable alternative >> >>> to softphone deployment. >> >>> >> >>> For me, main issue with Softphones is the amount of work needed for >> >>> installation and configuration. >> >>> Also, Softphones must be carefully choosen if Deskphone-like quality >> >>> is expected. >> >>> >> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade >> >>> Softphone features (call history, BLF, ...) for WebRTC deployment >> >>> simplicity. >> >>> >> >>> What do you think of this ? >> >>> What kind of experience did you met with such WebRTC deployments ? >> >>> What about classic telephony features (CallTransfer) ? >> >>> Have you tried Cyber Maga Phone 2K ? >> >>> >> >> If you can get it to work WebRTC is a good option. The problem >> is >> >> that any changes in your network may disrupt it and even trying to >> >> replicate your installation is difficult. I have it working fine on my >> >> website so customers can call us directly from our web page but I never >> >> could get Cyber Mega Phone 2K to work on the same server. We used >> JSSIP >> >> to create the webrtc phone on our website. >> > We just updated the documentation for how to get CMP2K working on the >> > wiki [1]. We'd love some feedback if you still have issues getting it >> > setup so that we can improve the docs. >> > >> > [1] >> https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone >> > >> > Best wishes, >> > Matthew Fredrickson >> > >> I followed the procedure indicated in the link but I cannot get >> remote video. I can only see my own feed. We do have audio for a >> little while. For some reason the users get disconnected after a few >> minutes even though you can still see your video feed on screen. This >> was done with Asterisk 15.6.0 >> >> -- >> Telecomunicaciones Abiertas de México S.A. de C.V. >> Carlos Chávez >> +52 (55)8116-9161 >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Astricon is coming up October 9-11! Signup is available at: >> https://www.asterisk.org/community/astricon-user-conference >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> 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 -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WebRTC as Softphone substitute ?
Le mer. 26 sept. 2018 à 16:40, Carlos Chavez a écrit : > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and configuration. > > Also, Softphones must be carefully choosen if Deskphone-like quality > > is expected. > > > > Now that WebRTC becomes ubiquitous, it might make sense to trade > > Softphone features (call history, BLF, ...) for WebRTC deployment > > simplicity. > > > > What do you think of this ? > > What kind of experience did you met with such WebRTC deployments ? > > What about classic telephony features (CallTransfer) ? > > Have you tried Cyber Maga Phone 2K ? > > > > If you can get it to work WebRTC is a good option. The problem is > that any changes in your network Very interesting ! Can you elaborate a bit further ? Are we talking about network settings (IP, routing table) or rather changes in browser themselves or both ? > may disrupt it and even trying to > replicate your installation is difficult. Yes, I also think replicating someone else's browser is nearly impossible if you don't control end devices (and even so ...). If tiny differences make a whole experience succeed or fail ... I thought WebRTC were stable now. If you have to dig in tiny details and can't even rely that your past will prevent most issues, it is rather discouraging. > I have it working fine on my > website so customers can call us directly from our web page but I never > could get Cyber Mega Phone 2K to work on the same server. We used JSSIP > to create the webrtc phone on our website. > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez > +52 (55)8116-9161 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WebRTC as Softphone substitute ?
Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for WebRTC deployment simplicity. What do you think of this ? What kind of experience did you met with such WebRTC deployments ? What about classic telephony features (CallTransfer) ? Have you tried Cyber Maga Phone 2K ? [1] https://github.com/asterisk/cyber_mega_phone_2k Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to implement an ENUM mock database ?
Hello, For personal lab testing, I would like a mock database, replacing a legacy ENUM database. More precisely, I would like to: - play with Asterisk's ENUMLOOKUP, ENUMQUERY and so on functions - populate mock db with a couple of fake numbers or ranges of numbers - test common use-cases (unknown number, ...) I'm entirely discovering ENUM, itself. How can I easily implement such db ? With dnmasq (don't laugh) ? With things like [1] ? Best regards [1] https://github.com/hadrienk/enum-dns -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to remove launching shell command from Asterisk CLI
Yes: I never thought of using sudo to also forbid access some apps. Using it for that is very smart ! Thank you for sharing it here. I'll experiment with this and report here my findings. Thanks again 2018-08-14 19:50 GMT+02:00 John Kiniston : > I use sudo to limit this. > > Cmnd_Alias CAPTAGENT = /sbin/service captagent stop, /sbin/service > captagent start, /sbin/service captagent restart > Cmnd_Alias ASTERISK = /sbin/service asterisk stop, /sbin/service asterisk > start, /sbin/service asterisk restart, /usr/sbin/rasterisk, > /usr/sbin/asterisk, /usr/sbin/tcpdump > Cmnd_Alias EDITORS = /bin/nano, /etc/asterisk/[A-z]*, /usr/bin/vim > /etc/asterisk/[A-z]* > > %pbxadmin ALL = (root) NOEXEC: EDITORS, ASTERISK, CAPTAGENT > > This prevents my admin users from being able to spawn a shell or > subprocess from vim, nano, and the asterisk console. > > On Tue, Aug 14, 2018 at 7:43 AM Olivier wrote: > >> Hello, >> >> Is there a way to let someone access to Asterisk CLI and type whatever >> command (s)he likes but the shell command (the ones started by !) ? >> >> Ideally, it could be an argument to rasterisk: >> >> rasterisk --no-shell >> >> When done, a session could be like this: >> >> > pjsip show endpoints >> ... >> > core reload >> ... >> > !rm /etc/foobar >> Forbidden >> >> Suggestions ? >> >> Best regards >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > 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 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to remove launching shell command from Asterisk CLI
Hello, Is there a way to let someone access to Asterisk CLI and type whatever command (s)he likes but the shell command (the ones started by !) ? Ideally, it could be an argument to rasterisk: rasterisk --no-shell When done, a session could be like this: > pjsip show endpoints ... > core reload ... > !rm /etc/foobar Forbidden Suggestions ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to properly execute rasterisk over SSH ? [SOLVED]
2018-08-14 15:53 GMT+02:00 Barry Flanagan : > On Tue, 14 Aug 2018 at 14:34, Olivier wrote: > >> Hello, >> >> I've got Asterisk installed on a Debian Stretch host. >> From another Debian Stretch host on which Asterisk is not installed, I >> want to run rasterisk over SSH in one step with: >> >> ssh root@foobar rasterisk >> >> The above command "rougly works" but some non-printable characters cause >> undesirable issue during CLI session. >> >> > Maybe try > > ssh -t root@foobar rasterisk > > Hope this helps. > > -Barry Flanagan > > Any suggestion ? >> >> Regards >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> 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 -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Yas, adding -t solved this issue ! Thank yoy very much Barry for replying ! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to properly execute rasterisk over SSH ?
Hello, I've got Asterisk installed on a Debian Stretch host. >From another Debian Stretch host on which Asterisk is not installed, I want to run rasterisk over SSH in one step with: ssh root@foobar rasterisk The above command "rougly works" but some non-printable characters cause undesirable issue during CLI session. Any suggestion ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Do you set chan_sip's ignoresdpversion to true ?
Hello, I've just discovered chan_sip's ignoresdpversion setting. Do you use it ? If positive which kinnd of issue could you solve with it ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to ignore REFER entirely with chan_sip or PJSIP ?
Hello, In my testing, I saw that Asterisk always included a REFER value in each INVITE's Allow header, no matter how allowtransfer/allow_tranfer was set. Is there a way to remove this REFER value entirely either globally or specifically for a given peer/endpoint ? Which telephony feature would loose without REFER method ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?
2018-06-05 20:29 GMT+02:00 George Joseph : > > > On Tue, Jun 5, 2018 at 10:59 AM Olivier wrote: > >> >> >> 2018-06-05 15:27 GMT+02:00 George Joseph : >> Thank you very much, George for replying. >> >>> >>> >>> On Tue, Jun 5, 2018 at 3:35 AM Olivier wrote: >>> >>>> Hi, >>>> >>>> After a long discussion with a friend, I would like to ask here: >>>> >>>> 1.According SIP RFCs, is possible/recommended to have different values >>>> in From and P-Asserted-Id fields ? >>>> For instance, From field showing 123456789 and P-Asserted-Id showing >>>> 987654321 (beside privacy considerations) ? >>>> >>> >>> Possible? yes absolutely. >>> >> >> How would you then configure both headers, respectively ? >> >> From memory, in previous testings, whenever CALLERID was set to >> WHATEVER, P-Asserted-Id was also set to WHATEVER and vice versa, so that I >> inferred from this that P-Asserted-Id was meant for Privacy >> considerations and nothing else (see [1]) >> > > PAI should be used to indicate the calling party's identification > regardless of privacy concerns. In the dialplan you can use the CALLERPRES > function to control privacy on a call by call basis. > > > I'm sorry but I still have a doubt ... Let me re-phrase my question: My setup is: Asterisk <--- PJSIP ---> Bob For a reason, I want Bob's phone to receive a call with the following headers: From: "Foo" ;tag=as75ee8c7c P-Asserted-Id: "Foo" >;whatever My dialplan is: same = n,Set(CALLERID(num)=999) XXX same = n,Dial(PJSIP/123456@bob) What shall I replace XXX with to allow me to set 8 in the user part of P-Asserted-Id URI (see example above) ? CALLERPRES would change From or P-Asserted-Id but not having different user parts in URI, would it ? To my knowledge, a possible way to implement what I'm after is to "turn off" P-Asserted-Id feature, add a custom P-Asserted-Id header with PJSIP_HEADER. Am I missing something ? > > >> >> >> [1] https://www.voip-info.org/p-asserted-identity-and-remote-par >> ty-id-header/ >> >> >>> Recommended? who knows? Implementations are all over the place. I've >>> always thought of the From header as identifying the user agent making the >>> request which kinda agrees with RFC3261. The PAI header should contain >>> the identity of the original caller. >>> >>> >>>> >>>> 2. When Bob forwards to Cory a call coming from Alice, would expect >>>> Diversion/History-Info header to include Alice's number ? >>>> >>> >>> No. The diversion header shows who the diverter is. >>> https://tools.ietf.org/html/rfc5806 >>> >> Thank for this reference: I think I confused diverter/caller/callee roles when I first read this document. So, if Bob forwards to Cory a call from Alice, in which headers would you expect Alice and Bob numbers to respectively appear ? > > Best regards > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> -- >>> George Joseph >>> Digium, Inc. | Software Developer >>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US >>> Check us out at: www.digium.com & www.asterisk.org >>> >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Check out the new Asterisk community forum at: >>> https://community.asterisk.org/ >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> 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 -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ &
Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?
2018-06-05 15:27 GMT+02:00 George Joseph : Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in >> From and P-Asserted-Id fields ? >> For instance, From field showing 123456789 and P-Asserted-Id showing >> 987654321 (beside privacy considerations) ? >> > > Possible? yes absolutely. > How would you then configure both headers, respectively ? >From memory, in previous testings, whenever CALLERID was set to WHATEVER, P-Asserted-Id was also set to WHATEVER and vice versa, so that I inferred from this that P-Asserted-Id was meant for Privacy considerations and nothing else (see [1]) [1] https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/ > Recommended? who knows? Implementations are all over the place. I've > always thought of the From header as identifying the user agent making the > request which kinda agrees with RFC3261. The PAI header should contain > the identity of the original caller. > > >> >> 2. When Bob forwards to Cory a call coming from Alice, would expect >> Diversion/History-Info header to include Alice's number ? >> > > No. The diversion header shows who the diverter is. > https://tools.ietf.org/html/rfc5806 > > > > >> >> Best regards >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > George Joseph > Digium, Inc. | Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?
Hi, After a long discussion with a friend, I would like to ask here: 1.According SIP RFCs, is possible/recommended to have different values in >From and P-Asserted-Id fields ? For instance, From field showing 123456789 and P-Asserted-Id showing 987654321 (beside privacy considerations) ? 2. When Bob forwards to Cory a call coming from Alice, would expect Diversion/History-Info header to include Alice's number ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] When should a Progress or Ringing be used in a today's telephony ?
Hello, Thinking back to my current practices, I would be very curious to share here about when should applications such as Congestion, Progress or Ringing be used in today's telephony. I would define today's telephony with: - SIP phones - Asterisk - a SIP trunk to an ITSP - fixed or mobile lines reachable through this ITSP 1. When Asterisk receives a SIP call coming from PSTN, is there a time frame within which Asterisk must reply something to keep caller from canceling the call ? Where does this limit come from ? From SIP RFC ? From local regulation bodies ? 2. Which SIP signal is required to stop call cancellation in the previous case ? 3. When Asterisk receives a call, either from PSTN or from a SIP phone) it cannot process (unkown callee, whatever reason, ...), should you stop processing with Hangup or Congestion ? Hangup application allow for exit code customization, if I'm not mistaken, but Congestion exists for a reason. 4. Is it a good practise to send a 180/183 when you don't get one ? 5. I observed I sometimes got a 100 Trying then a 183 session Progress when outcalling some (mobile) phones while simpy getting 100 Trying when some other (mobile) phone through the same carrier (most probably, end devices were not managed by the same (mobile) telephony provider). What explains such difference ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on PJSIP's endpoint section in wiki
Hello, I don't know if this list is the best place to ask such question but here it is, anyway. In page [1], I can read in PJSIP's endpoint section configuration reference: identify_by username,location Way(s) for Endpoint to be identified Then clicking over identify_by text, you can read: identify_by ... supported options are username, ... and auth_username How do yopu read it ? I would expect the first line to written as: dentify_by username,auth_username Way(s) for Endpoint to be identified Thoughts ? Best regards [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_identify_by -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP global section ignored in Asterisk 13.14.1 [SOLVED]
2018-04-27 14:59 GMT+02:00 Joshua Colp <jc...@digium.com>: > On Fri, Apr 27, 2018, at 9:57 AM, Olivier wrote: > > Hello > > > > I've just discovered this [1] invaluable blog post (thank you very much > > Richard for writing it) and its reference to PJSIP's > > endpoint_identifier_order setting. > > > > On my Debian Stretch box powered with a packaged Asterisk 13.14.1, I > edited > > a pjsip.conf file with the following content (and nothing more): > > [global] > > endpoint_identifier_order=auth_username,ip,username > > max_forwards=50 > > This is incomplete. You need to also have "type=global". > Adding a "type=global" line solved this issue as I missed the comment ";type= ; Must be of type global (default: "")" two lines bellow. Thank you very much for correcting me. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP global section ignored in Asterisk 13.14.1
Hello I've just discovered this [1] invaluable blog post (thank you very much Richard for writing it) and its reference to PJSIP's endpoint_identifier_order setting. On my Debian Stretch box powered with a packaged Asterisk 13.14.1, I edited a pjsip.conf file with the following content (and nothing more): [global] endpoint_identifier_order=auth_username,ip,username max_forwards=50 Then: # service asterisk restart # asterisk -rx "pjsip show settings" | grep forwa max_forwards: 70 # asterisk -rx "pjsip show identifiers" Identifier Names: name not specified ip username anonymous auth_username # grep -v '^;' modules.conf [modules] autoload=yes noload => pbx_gtkconsole.so noload => pbx_kdeconsole.so noload => app_intercom.so noload => chan_modem.so noload => chan_modem_aopen.so noload => chan_modem_bestdata.so noload => chan_modem_i4l.so noload => chan_capi.so load => res_musiconhold.so noload => chan_alsa.so noload => cdr_sqlite.so noload => app_directory_odbc.so noload => res_config_odbc.so noload => res_config_pgsql.so [global] What do you think of this ? Best regards [1] http://blogs.asterisk.org/2018/02/07/identifying-endpoint-pjsip/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to check modules loading order or force such order ?
Hello, >From [1], you can read: "If you don't have an identify section defined, or else you have res_pjsip_endpoint_*identifier_ip* loading *after* res_pjsip_endpoint_ *identifier_user*, then ..." To remove the above uncertainty coming from modules loading order, how can you either or both : - check modules loading order - force such order. Would it be welcomed addition to the wiki pages ? Best regards [1] https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Explain PJSIP user matching within inbound SIP trunks
May I add the following to my previous question: - When I wrote "only looking at From header", I meant looking at domain part of From header, leaving user part of From header for other purposes. - With a From header such as From: "John Doe" <sip:123456789@foobar>, it would match foobar with a [foobar] endpoint, and then set CALLERID(name) to John Doe and CLAARID(num) to 123456789 2018-04-27 12:00 GMT+02:00 Olivier <oza.4...@gmail.com>: > Hello, > > I'm setting an Asterisk 13.14.1 box (Debian Stretch with packaged > Asterisk) to implement SIP trunking services ie to both trunk with carrier > trunks and IPBX trunks from various brands. > > For various reasons, I was inclined to implement this services with > pjsip_wizard.conf and I'm realizing I still have some remaining questions. > > For the moment, letting registration questions aside, which of the > following sentences is correct for Asterisk's PJSIP stack: > > 1. it would identify an incoming call only looking at From header ignoring > IP settings (both IP address and port), > 2. it would identify an incoming call only looking both at From header and > IP settings (both IP address and port), > 3. it would identify an incoming call only looking both at IP settings, > ignoring From header for identification but using it for other things > (setting CallerID, ...). > > Best regards > > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Explain PJSIP user matching within inbound SIP trunks
Hello, I'm setting an Asterisk 13.14.1 box (Debian Stretch with packaged Asterisk) to implement SIP trunking services ie to both trunk with carrier trunks and IPBX trunks from various brands. For various reasons, I was inclined to implement this services with pjsip_wizard.conf and I'm realizing I still have some remaining questions. For the moment, letting registration questions aside, which of the following sentences is correct for Asterisk's PJSIP stack: 1. it would identify an incoming call only looking at From header ignoring IP settings (both IP address and port), 2. it would identify an incoming call only looking both at From header and IP settings (both IP address and port), 3. it would identify an incoming call only looking both at IP settings, ignoring From header for identification but using it for other things (setting CallerID, ...). Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VMWare guest crash with ^@ character in logs. Where to look at to find root cause ?
Hello, Today, one Asterisk instance of mine crashed. This instance is only providing SIP trunking (from IPBXs to carriers, no transcoding, playing of voice prompts and fancy dialplan tricks, ). The instance is built : - as a VMWare 6.5 guest, - with Debian Stretch (9) - and asterisk 13.20.0 built from source with DONT_OPTIMIZE, BETTER_BACKTRACES options At the time the machine crashed, I can see a long list of null characters (represented as ^@, if I'm not mistaken) in various log files (/var/log/syslog, /var/log/daemon.log, /var/log/asterisk/full) but not in /var/log/kern.log. When the machine crashed, it was not even responding to SSH queries and did not produce any core file. I would like to find the root cause of the issue. My reasonning is : - as Null character is present in /var/log/asterisk/full, then necessarily, this output comes from Asterisk application so Asterisk is on the "cause path" - Asterisk may or may not be the root cause the issue. What would you suggest me to investigate ? What should I infer from coredump absence ? Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is ASTDB /pbx/UUID for ? Can I duplicate whole ASTDB from cluster active member to passive member ?
Hello, I'm working on an Asterisk active/passive cluster where the following applies: - members are both VM - /etc/asterisk files are copied from one provisonning server to both VM - asterisk is running on active member - asterisk is not running on passive member - members share floating IP addresses - I don't remember whether Asterisk was already installed or not when one VM image was copied to serve as a base for the other VM image - system config of both members are identical except host names and some IP addresses. Looking at ASTDB content I'm seeing the following entries: /dundi/secret /dundi/secretexpiry /pbx/UUID I'm also seeing entries this like: /SIP/Registry/foo /registrar/contact/bar;@da9064cc4962b6a60d4 To minimize service interruption when a passive member become active, I was thinking of regularly copying ASTDB file from active to passive cluster member. As I'm not confident that my own testing would cover all cases, I would like to ask here: 1. What is /pbx/UUID for ? 2. Is it safe to copy entries like /registrar/contact/bar;@da9064cc4962b6a60d4 ? 3. Suggestions ? Comments ? Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Load testing with SIPp
All mentioned boxes are VMWare VMs probably configured with default drivers and settings. Unfortunately, I don't know which network equipements are used when those boxes communicate with each other. So yes, maybe I banged into an unexpected limit because of this ignorance. Anyway, from my testing, I was mostly surprised by 2 things: 1- I consistently hit a 200 calls when I let run an Asterisk instance on the SIPp (calls passing locally from SIPp to a first Asterisk instance before hitting SUT): I thought that communications inside a given VM was "unlimited and cheap" and I was apparently wrong, Disabling Asterisk in my first VM allowed me to hit a 500 limit at the price of loosing Asterisk flexibility to log RTCP stats, select exotic codec and so on. 2- Increasing asterisk.conf's maxfiles value (from 1 to 4, for instance), allowed me to get more than 695 successfull call out of 700 but I could succeed to get, even once, 700 successfull calls, even when I tried with my current 40 maxfiles limit. 2018-03-06 23:35 GMT+01:00 Bruce Ferrell <bferr...@baywinds.org>: > > On 03/06/2018 01:58 PM, Olivier wrote: > >> Hello, >> >> I'm running load testing sessions. >> My System Under Test is an asterisk 13 with 16GB, configured with >> maxfiles set to 400 000. >> This system is supposed do produce simple SIP trunking services without >> transcoding. >> >> >> The box sending call to my System Under Test is anabled with SIPp. >> I'm banging on a 700 concurrent calls/50 CAPS limit I would like to >> improve, if possible. >> >> Tests are done with both signaling and media like this: >> >> SIPp <---> SUT (asterisk 13) <---> Asterisk box echoing media >> >> I checked bandwidth first and got 930 Mb/s on each leg (from SIPp to SUT >> or SUT to echoing box) using iperf3 TCP testing though my target relies on >> UDP >> >> >> My questions are: >> >> 1. Have you ever noticed a better scalability using UDP or TCP ? >> >> 2. Where do Retransmission I'm observing on SIPp console most probably >> come from ? Network issues ? My SIPp not beeing correctly tuned ? Lack of >> resources somewhere ? >> >> 3. Recommandations ? Suggestions ? >> >> Best >> >> >> I do network management for a living. > > In your description, I see nothing to describe the network other than an > observed 930Mb/s. > > What is the network configuration; What NIC(s), switches etc. > > Treating these as effectively "unlimited" is a certain recipe for banging > into unexpected limits. > > Different NICs and switchs can and do provide differing levels of > performance. > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Load testing with SIPp
Hello, I'm running load testing sessions. My System Under Test is an asterisk 13 with 16GB, configured with maxfiles set to 400 000. This system is supposed do produce simple SIP trunking services without transcoding. The box sending call to my System Under Test is anabled with SIPp. I'm banging on a 700 concurrent calls/50 CAPS limit I would like to improve, if possible. Tests are done with both signaling and media like this: SIPp <---> SUT (asterisk 13) <---> Asterisk box echoing media I checked bandwidth first and got 930 Mb/s on each leg (from SIPp to SUT or SUT to echoing box) using iperf3 TCP testing though my target relies on UDP My questions are: 1. Have you ever noticed a better scalability using UDP or TCP ? 2. Where do Retransmission I'm observing on SIPp console most probably come from ? Network issues ? My SIPp not beeing correctly tuned ? Lack of resources somewhere ? 3. Recommandations ? Suggestions ? Best -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Comparison of PJSIP and SIP in Asterisk database
Hello,
I'm currently trying to configure a passive Asterisk instance that must
backup an active Asterisk instance.
Each instance is connected this way:
PSTN <---> Gateway <-- SIP --> Asterisk <-- SIP --> endpoints or IPBXs
Most endpoints connect through registration.
With chan_sip, Asterisk saved registration data in its database with lines
such as:
/SIP/Registry/spa3102 : 192.168.64.207:5060:
3600:7013:sip:spa3102@192.168.64.207:5060
Reading such lines in active instance and copying them back in passive
instance, I think you had a mean to have a passive instance ready to treat
calls coming from PSTN as soon as it would become active (I never
experimented with this).
Now, with PJSIP, Asterisk saves registration data with lines such as :
/registrar/contact/foobar: {"via_addr": ... }
Have you tried to copy such registration data from one instance to an
aother one ?
What happened then ?
Best regards
--
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Re: [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13
Hello, Digging a bit further, having a local cdr_custom CSV seems to make updatings work ! I did have enough time to properly test this and become more affirmative but it seems to depend on active CDR backend; 2018-02-21 22:19 GMT+01:00 Olivier <oza.4...@gmail.com>: > As a complement to my previous post, may I add that I observed the > following behaviours: > > 1. On one system (Debian Stretch/asterisk 13.19 compiled from source), > hangup causes are correctly saved in a custom CDR column. > > 2. On an other system (Debian Stretch/packaged asterisk), some rtcp stats > are not-correctly saved in a custom CDR column (I also tried unsuccessfully > with userfield column). > > In both cases: > - CDR updates are triggered by a hangup handler pushed with > CHANNEL(hangup-handler-push). > - CDR are saved through ODBC i, aMariaDB or Postgres database. > > Toughts ? > > 2018-02-21 0:07 GMT+01:00 Olivier <oza.4...@gmail.com>: > >> Hi, >> >> Reading this old thread, may I ask if keeping hangup handlers from >> updating CDR values still enforced in Asterisk 15 ? >> If positive, would it be very complex to add in Asterisk, a configuration >> option allowing a system administrator to list in cdr.conf, the CDR fields >> allowed to be updated in hangup handlers ? >> >> I'm planning to store some RTCP stats. >> Saving them in CDR(userfield) would be perfect. >> >> Cheers >> >> >> 2015-08-10 15:05 GMT+02:00 Matthew Jordan <mjor...@digium.com>: >> >>> >>> On Tue, Aug 4, 2015 at 9:16 AM, Filip Jenicek <fjeni...@kerio.com> >>> wrote: >>> >>>> With endbeforehexten=no I actually get two CDR entries. One for the >>>> call and a second one for the "h" extension. >>>> "","13","10","sip-locals","""13"" <13>","SIP/13-0006","SIP/1 >>>> 0-0007","Dial","SIP/10","2015-08-04 06:28:44","2015-08-04 >>>> 06:28:45","2015-08-04 06:28:47",3,1,"ANSWERED","DOCU >>>> MENTATION","1438669724.6","empty" >>>> "","13","h","sip-locals","""13"" >>>> <13>","SIP/13-0006","","NoOp","changed","2015-08-04 >>>> 06:28:47","2015-08-04 06:28:47","2015-08-04 06:28:47",0,0,"ANSWERED","DOCU >>>> MENTATION","1438669724.6","changed" >>>> The first one contains the call itself. There are durations, CDR >>>> variables set during the call, etc. >>>> The second one contains only things configured in the "h" extension. >>>> >>>> With endbeforehexten=yes, the cdr contains: >>>> "","13","10","sip-locals","""13"" <13>","SIP/13-0006","SIP/1 >>>> 0-0007","Dial","SIP/10","2015-08-04 06:28:44","2015-08-04 >>>> 06:28:45","2015-08-04 06:28:47",3,1,"ANSWERED","DOCU >>>> MENTATION","1438669724.6","empty" >>>> There is only the call, nothing from the "h" extension. >>>> >>>> I forgot to mention that I'm using Asterisk 13.1-cert2. Modifying CDR >>>> records in the "h" extension used to work fine with Asterisk 1.8. >>>> >>>> By analyzing the code I must confirm that the endbeforehexten option >>>> behaves exactly according to its description: >>>> As each CDR for a channel is finished, its end time is updated >>>> and the CDR is finalized. When a channel is hung up and hangup >>>> logic is present (in the form of a hangup handler or the >>>> h extension), a new CDR is generated for the >>>> channel. Any statistics are gathered from this new CDR. By enabling >>>> this option, no new CDR is created for the dialplan logic that is >>>> executed in h extensions or attached hangup handler >>>> subroutines. The default value is yes, indicating >>>> that a CDR will be generated during hangup logic. >>>> >>>> I tried to delay the "h" extension by several seconds and I found out, >>>> that the CDR record is sent to the cdr backend later. Unfortunately, it is >>>> not modifiable
Re: [asterisk-users] Which CDR processing for high load ?
1. Would say CDR_ODBC has a greater chance than CDR_CUSTOM (if I may call them both as such) to become a bottleneck under pressure ? 2. I didn't know about batch mode existence. Thanks for presenting it. In such batch mode, where are stored CDRs before being written to files or database ? Does it also apply to data later found in cdr-scv/Master.csv file ? 2018-02-22 18:14 GMT+01:00 Richard Mudgett <rmudg...@digium.com>: > > > On Thu, Feb 22, 2018 at 5:23 AM, Olivier <oza.4...@gmail.com> wrote: > >> Hello, >> >> I'm load testing a new Asterisk 13 system (Debian Stretch, packaged >> asterisk). >> One system writes CDR though an ODBC connection to a local Postgres >> database over the LAN. >> >> >> When sending 50 new calls per second with SIPp, I'm seeing one system >> outputs : >> taskprocessor.c: The 'subm:cdr_engine-0003' task processor queue >> reached 5000 scheduled tasks again. >> >> This [1] thread mentions such limit. >> >> I was thinking of simply writing CDR entries to a local file before >> centralizing them into my database. >> >> What would you suggest on this topic ? >> > > You should get better CDR performance enabling batch mode which is > disabled by default for legacy reasons. > > The CDR code processes all call events in one thread to create CDRs. This > thread is what processes the > 'subm:cdr_engine-0003' task processor queue you mention above. If you > don't have batch mode enabled > then that same thread also has to immediately write the CDRs to each back > end configured. If you do have > batch mode enabled then the CDRs are passed to another thread to write to > the back ends. > > You need to be using at least v13.19.1 or v15.2.1 to also have some CDR > performance enhancements to > help CDR processing of call events. > > For information about ODBC connection pooling performance problems see [2]. > > Richard > > [2] http://blogs.asterisk.org/2016/06/15/asterisk-odbc-connections/ > > >> >> >> Best regards >> >> [1] http://lists.digium.com/pipermail/asterisk-dev/2016-June/075607.html >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> 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 -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
