Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem
>>>>> "JC" == Joshua C Colp writes: JC> To be specific, this is in PJSIP land. There was no insisting or anything JC> and it wasn't a decision we originally made. It's the way that Teluu JC> implemented the TLS transport in PJSIP and since we use PJSIP then it JC> applies to us. my recall is more likely a bit older than that, before pjsip. there was a thread either in bugs or on one of the lists. but as later notes pointed out (and i really ought to have thought of ☹) it is only relevant, as you noted, if verify is on. at the time i was a fan on wildcards. then le came along, and then added dns01 support. now i prefer a separate cert each plus a 3/1/1 tlsa for each port. but at the time it was anoying. -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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 to Twilio over TLS - wildcard cert problem
>>>>> "KT" == Kingsley Tart writes: KT> I can't get Asterisk to send a SIP call to Twilio over TLS KT> because it complains about Twilio's wildcard certificate. the sip rfc claims that wildcard certs should be invalid for sip. digium insisted on following that advise as set in stone, and so asterisk refuses such certs. i doubt that stance is different under sangoma. the only workaround is to remind twil of the rfc and get them to replace the wildcard with an rfc-copliant cert. at least for the sip ports. -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] T-38 re-invite issue
>>>>> D'Arcy Cain writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] T-38 re-invite issue
>>>>> "DC" == D'Arcy Cain writes: DC> Perhaps someone can explain what t38timeout is supposed to do A 'git grep t38timeout' on the src leads one to res/res_fax.c, where one case see that it is the number of miliseconds to permit for t38 negotiation to complete once it starts. Ie after both sides select t38, until they agree on the t38 terms. -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] Let's encrypt privkey : Specified certificate file could not be used
>>>>> "JK" == Jonas Kellens <jonas.kell...@telenet.be> writes: JK> [Jun 2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441 JK> ast_rtp_dtls_set_configuration: Specified certificate file JK> '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance JK> '0x7f920c538a78' could not be used That error means that openssl's SSL_CTX_use_certificate_file() returned an error. The later error is just a result of that one. Does the uid/gid used for asterisk have access to the key? If the uid you use for asterisk is called asterisk, run this as root: su -c 'cat /etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' - asterisk If it fails, then the problem is permissions. You may need to alter the permissions on /etc/letsencrypt to allow non-root uids to access the symlinks and their targets. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] log incoming calls without answering
>>>>> "FM" == Fabio Moretti <fmore...@tecytal.com> writes: FM> when a call enter, asterisk sense it and store its values (callerid, FM> date and time, etc) somewhere, but nothing more, people will answer FM> using the old analog phone. The goal is to have a log of the inbound FM> calls without touching the old analog system (it's shared between FM> different subjects). IIUC, the pots line has both some number of analog phones a/o fax machines on it, plus a fxo->sip gateway, yes? You can route the sip portion to asterisk and have the dialplan log everything but never answer. You may want to call the Ringing dialplan application, but even that may not be required. OTOH, calling Ringing should prevent the gateway from assuming that the asterisk machine never saw the INVITE. Eventually, when the other extension answers, the fxo->sip gateway will cancel the sip call just like it would if the caller hung up. (There is a possibility that any given gateway may not cancel the sip call until the analog call is completed; you need to test.) -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] Voicemail asking for login
>>>>> "DC" == D'Arcy Cain <da...@vybenetworks.com> writes: DC> I did debug 10 and saved the console output into files which I DC> compared side by side. No material difference. In that case I'd add more debug statements to apps/app_voicemail.c (in vm_exec()), including a log at the start of what is in *data and args. Looking at it, it only plays vm-whichbox when ast_strlen_zero(data), which implies that the args to Voicemail are not making it through. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] Voicemail asking for login
I enable full log and run 'core set debug 9' before doing a pair of tests. (The full log is easier to grep than the console output.) Then compare a working vs stocktrans2 side by side. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] E-911
There are a few services available, in addition to the offerings from (some? most?) upstreams. I'm aware of http://www.bulkvs.com/e911.html It is probably the most affordable option if your upstream does not offer it. I use their origination and termination services, but have not needed to try their e911. They only charge $0.72/month/number for e911. (The way e911 works each number which may be used as callerid for 911 calls must have an address added to the e911 database. You then need to arrange to use a/the number corresponding to the address where the emergency is as the userpart in the SIP From: header.) -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] inbound T38 to email
>>>>> "JL" == Jeff LaCoursiere <j...@jeff.net> writes: JL> Is there any new modern way to take t38 from a (SIP) DID provider and JL> route to email? Thanks for any insight :) With recent versions of asterisk you can use the ReceiveFax and SendFax dialplan applications. They can be configured (in res_fax.conf) to use t38 when available. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?
I saw this on the bug list first and sent a reply, but for the archives I'll copy it here, too. REMAINDER() calls libm's remainder(3) or remainderl(3), infix % calls fmod(3) or fmodl(3). remainder(3) is defined to round the quotient to the nearest int (always using round-to-even, notsithstanding the rounding mode), fmod(3) is defined to trunc(3)ate the quotient. So the result of x%y will always be in the range [0,x] and the results of remainder(x,y) will be in the range (-y/2,y/2]. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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 Stun/ICE
>>>>> "AS" == A J Stiles <asterisk_l...@earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Stun/ICE
>>>>> "JC" == Joshua Colp <jc...@digium.com> writes: JC> I disagree that it makes it worthless for a large number of JC> users. It's only within the last few days that a few people have run JC> into this particular issue where they have a public IP address that is JC> changing a lot and PJSIP does not support changing it without a JC> restart. If it were a huge sweeping issue we'd be seeing it more JC> often. If it continues to show up a community member or us (heck maybe JC> even myself in my spare time) may look into implementing it. It is only in the last few days that this discussion occurred. This is not the first mention of problems with using pjsip on dynamic ips. Most affected users are probably still using chan_sip. Or haven't even upgraded to 13 yet. I gave up switching my edge asterisk to pjsip at least twice because I couldn't figure out how to configure it properly for a dynamic ip. And I sent a note to one of the lists at least on the 2nd attempt. That install doesn't need nat for sip/rtp since it runs on the router, but it does need to handle dynamic ip. In short, this breaks sip for nearly everyone using asterisk at home and even a lot of businesses. It may not break it every day, but it is enough to drive a lot of people away from asterisk once they learn of it. JC> The support level for chan_sip has already been changed and was JC> announced long ago. had this issue been noted in that announce you'd have received -- I expect -- quite a few complaints. This flies in the face of all of the (very welcome) work which went into supporting reload rather than restart. Getting pjsip to support changes on a reload would be an acceptable first step. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Stun/ICE
>>>>> "JC" == Joshua Colp <jc...@digium.com> writes: JC> This stems from PJSIP not being dynamic with transports (it JC> doesn't like its environment changed to that degree while JC> in use). I'm afraid if your IP changes you'd have to restart JC> Asterisk when you are using PJSIP. Wow. I say this having voted for pjsip over the listed alternatives back when the plan to depricate chan_sip was first floated: That should have excluded pj from the options. Which of course means there were no reasonable options. Can ari get around that bug? Lack of full support for traversing nat makes pjsip worthless for a large number of users. And the whole point of realtime is to have all of the rt config fully dymanic. If ari cannot avoid that limitation, chan_sip should get full ongoing maintainance until pjsip is fixed. -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ARI all subscribe
I wasn't able to make it back to the devcon after lunch or to as many of the talks as I'd have liked (the excessive a/c exacerbated by symptoms enough to be painful), so I probably missed something relevant to this... What is the syntax of an ALL subscription websocket url in ari? I'd like to use wscat with such a sub to get a better idea of what the various events look like. Thanks, -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anonymous SIP calls
Some of us do allow sip from the internet, but just like for smtp email protections are in order. I point my SRV records at dedicated sip proxies (I use kamailio) which check the INVITEd sip uri the same way my MXs check the SMTP Evelope-To addresses, and only allow INVITEs through to authorized destinations. And when those INVITEs make it to asterisk/freeswitch or the like, the dialplan is generally not direct to phone(s), but via an IVR. As an example, calling my email address via sip goes to an Asterisk FollowMe instance. I also provide my clients with dedicated sip addresses which avoid the protections. But the vast majority of the INVITEs coming to my public sip proxies are fraud attempts. My primary sip proxy has blocked over 32k fraudulent INVITEs over the last six months. And about one OPTIONS sip:100@... per hour by something calling itself friendly-scanner. Then again, the number of invalid sip INVITEs per public sip destination are fewer than the number of spam/virus type SMTP attempts per unit time. And all of the telemarking fraud I have had to deal with have come via pstn dids, not via direct sip. A half-gig virtual works fine for such a sip proxy. You may also want to look into getting an ISN number, check out http://freenum.org/ for the details. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s
JBB == James B Byrne byrn...@harte-lyne.ca writes: JBB tcpenable=yes JBB tlsenable=yes JBB tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB tlsdontverifyserver=yes JBB tlscipher=ALL JBB tlsclientmethod=tlsv1 You are missing the tls key. The config name is tlsprivatekey; set that to the filename of your tls key, akin to how tlscertfile is set. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s
Other things to consider: The transport config, which can be in [general] or in a peer's [] block. if you want tls-only, use transport=tls it also accepts tcp, udp or a comma-separated list. if given a list, it tries them in order If you need ast to register over tls, use something like this: register = tls://username:xxx...@sip-tls-proxy.example.org (copied from the example sip.conf). Set tlsbindaddr to the address to which to bind(2) the tls socket. tlsbindaddr=0.0.0.0 is typical in ipv4-only configs. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird SIP stuff
EW == Eric Wieling ewiel...@nyigc.com writes: EW Basically, the traffic is coming from 65.211.180.237 but the header is: EW f: sip:+1347545xxx@199.173.94.80:5060;user=phone;tag=4-45026-159e4a6-995949f-159e4a6 The From: header and the socket ip are often different. Check for Via: headers. Those show the path of the SIP. In your example, look for a Via which mentions 65.211.180.237. Note that the From does not necessary have to be a sip address reachable from the outside. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi fxo vs sip blf
It has been may years since I've done anything with a dahdi fxo; much has changed in the interim and I havne't found answers googling. The fxo hw is installed on the pots line in parallel to existing pots phones. My goal is to have a blf on the sip phone which lights whenever any of the devices on the pots line are off hook and which, when pressed, INVITEs the asterisk box such that it takes the fxo off hook and joins the existing conversation. Or just passes the dialtone over the rtp if eerything was on hook when the blf was hit. Clearly Hint is part of the solution, but my attempts so far have failed. At the moment I'm on 12.7, but still using chan_sip. Converting the chan_pjsip will be the next project for this box. What is the proper way to set this up? -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-eSpeak and Asterisk 12
O == Olivier oza.4...@gmail.com writes: O Hijacking this thread, [the espeak] default voice ... produces a O quite rebotic voice. Is this something that can be improved...? O exten = 1234,n,Espeak(This is a simple espeak test in english.,any) Try the other voices. Run espeak from the command line, try out the various voices and the other options which also are available in the Asterisk-eSpeak espeak.conf file until you find a combination which sounds OK. Depending on your hardware and settings, you may need to have espeak write its output to a file, and use another applicaion to play that. The option space for espeak has large variability. Flite also needs such tuning for nice output. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FollowMe reinvites
For a sip-only application, what exactly is required to ensure that calls completed via followme are reinvited? Can it at all? The code after outbound = findmeexec(targs, chan) calls ast_bridge_ call(). I don't see anything there which can cause a reinvite, yes? When the same peer is used for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the legs are associated with different remotes, I'd prefer to proxy only when NATs a/or v4-v6 gatewaying are involved. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available
BF == Barry Flanagan barryf-li...@flanagan.ie writes: BF Something like: BF [peer_inbound] BF context=peercontext BF type=peer BF host=192.168.1.1 BF ...should do the job. That of course was something I tested, although I used the hostname; perhaps the dual-v4/v6 got in the way? -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] srtp/dtls when sip is clear over lo
JColp == Joshua Colp jc...@digium.com writes: Are you saying that asterisk doesn't care whether the sip is secure and will happily negotiate srtp depending only on whether the remote is willing to do so? (That may come off as harsh; I do not mean it to be so, since it is what I want. :) JColp Yes. Thanks! -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] srtp/dtls when sip is clear over lo
JColp == Joshua Colp jc...@digium.com writes: JColp The media is not carried over the SIP signaling, Please give some credit, eh? Given the sdp-negotiated srtp is not secure unless the sip is carried over tls, the Best Practice is to require tls (or even sips: uris) to agree to srtp. Are you saying that asterisk doesn't care whether the sip is secure and will happily negotiate srtp depending only on whether the remote is willing to do so? (That may come off as harsh; I do not mean it to be so, since it is what I want. :) And does anyone here have any operational experience on the matter of what other endpoints are willing to do in such cases? Thanks, -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available
And related thereto: What needs to be done on kama and ast to ensure that all incoming calls which route through a given kama box always matches a sip.conf [section] based on the socket(7)'s remote address, w/o any consideration of the INVITE's sip headers or body? I tried a several variations, but nothing quite worked. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] srtp/dtls when sip is clear over lo
Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly, will ast negotiate srtp or dtls even ast and the proxy speak sip in the clear over the lo interface? Avoiding encryption over lo can aid debugging, but will doing so also block secure media? -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXModem or T38Modem?
j == joakimsen joakim...@gmail.com writes: j I wouldn't mind if someone posted on the list a known working provider j with the proper configuration to use T.38. In my case I don't consider j it an issue with the provider because they sent the proper T.38 j Invite, but Asterisk IMO does not know how to handle it. Are you using a single credential-tuple with the provider? If the provider supports T.38 and if you can separate out fax lines, there is no need to stick asterisk between them and t38modem. Just have t38modem access the provider directly. Hylafax will handle the rest. (Look for things like sub-account, peer and/or trunk configs.) -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail variables on email subject
RdSS == Rafael dos Santos Saraiva rafaels...@gmail.com writes:
RdSS emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR}
RdSS Return:
RdSS Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?=
That is a proper encoding for an SMTP mail header which is in utf8.
RdSS Expected:
RdSS Subject: 1504|12|Teste - Rafael 1570|16
The sent header decodes to this string:
Subect: 1504|12|Teste_-_Rafael_1570|0:16
Note the colon from $VM_DUR (minutes:seconds).
MUAs are supposed to decode that.
-JimC
--
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6
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Re: [asterisk-users] Asterisk + iaxmodem + hylafax makes sometimes wedged for hylafax
GF == Gianni Fioretta gianni.fiore...@yetopen.it writes: GF disallow=all GF allow=alaw Given that, the sip leg will only permit alaw. The fact that the log showed no common codec means the other side did not include alaw in the codecs it offered. Perhaps for some destinations it will only offer ulaw? A sip debug or packet trace would show what it offered. I don't see anything in the iax.conf to explain the deadlock(?). Perhaps iax debug output, and/or a packet trace of the iax might explain it? I do recall some posts in the past about issues with sip-iax conversion. I don't remember *how* far in the past, or whether debian/ ubuntu's 1.8 might be affected. I'd run rasterisk -n in a script session, run 'sip set debug on' and 'iax2 set debug on' and wait for a modem to stop responding. (It might be a good idea to run script on a larger box and ssh into the asterisk box w/in the script session if the asterisk box has limited storage. The debug output could get LARGE before a modem stops.) The script command is in the bsdutils package (apt-get install bsdutils). -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + iaxmodem + hylafax makes sometimes wedged for hylafax
GF == Gianni Fioretta gianni.fiore...@yetopen.it writes: GF [Jul 4 17:45:03] NOTICE[22995]: chan_iax2.c:8775 update_registry: Restricting registration for peer 'modem99' to 60 seconds (requested 300) GF is that right? When iaxmodem registers with asterisk, it defaults to asking that the registration last 300 seconds. Asterisk defaults to permiting only 60 seconds per registration. You can eliminate that notice by configuring both sides with the same duration. In the iaxmodem config files, add a line: refresh 60 to make iaxmodem ask for 60 seconds. Alternatively, change the maxregexpire setting in asterisk's iax.conf (in the [general] section) topermit values at least as large as 300 seconds. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + iaxmodem + hylafax makes sometimes wedged for hylafax
GF == Gianni Fioretta gianni.fiore...@yetopen.it writes: GF -- Executing [0224300258@fax:1] Dial(IAX2/modem2-3460, SIP/centralino/0224300258) in new stack GF == Using SIP RTP CoS mark 5 GF -- Called SIP/centralino/0224300258 GF -- SIP/centralino-0284 is making progress passing it to IAX2/modem2-3460 GF -- SIP/centralino-0284 is ringing GF -- SIP/centralino-0284 is making progress passing it to IAX2/modem2-3460 GF -- SIP/centralino-0283 is making progress passing it to IAX2/modem4-8449 GF -- SIP/centralino-0283 is ringing GF -- SIP/centralino-0283 is making progress passing it to IAX2/modem4-8449 GF -- SIP/centralino-0284 answered IAX2/modem2-3460 GF [Jul 4 16:49:55] WARNING[22988]: chan_sip.c:9123 process_sdp: Failing due to no acceptable offer found That last line above shows that an outgoing fax attempt failed because the sip end wasn't able to negotaiate a codec for that part of the call. It looks like it was modem2's call which failed; modem4's call seems not yet to have been answered. I don't know whether that is what triggers the wedge, but the failure to negotiate a codec for the sip/rtp leg probably is a configuration bug. Which version of asterisk? Self compiled or a distribution's version? The sip.conf and iax.conf might help debug it. (Elide passwords, of course.) If you run the conf files through something like: :; egrep -v '^[[:blank:]]*;' iax.conf|egrep -v '^$' /tmp/short-iax.conf before editing the password lines it will be easier to read them. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing a fax with mutt does not succeed
D == Daniel - Asterisk earohua...@gmail.com writes: D I'm trying to send a received fax with mutt, when I try it from the Linux D shel it works, but when trying with Asterisk's System command it doesn't. Sending mail from a daemon is best done by calling /usr/sbin/sendmail directly, without relying on an MUA. The default mailcmd for app_voicemail is '/usr/sbin/sendmail -t' You might also want to use the -oi flag. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
If they will do atm over oc-n, perhaps that would work better. Ie, a perm virt circ for SS7 and as-needed vc's for ulaw. Atm oc-n cards with linux sw support are widely available, according to goog. libss7 and and ast *might* need a bit of patching to work with it, but it shoudn't take too much. Sip/rtp over private ptp ethernet is an option with at least some of the ILECs. They may call it virtual-pri or some such. Of course, if they are installing an actual sonet ring, and not just a spur, that can have built-in redundancy, depedning on physical routing. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem to install asterisk on vps digitalocean
t == troxlinux xserverli...@gmail.com writes: t I try to install asterisk on vps server , but fails when I want to t install dahdi There is no hardware for dahdi to use; you shouldn't need to install it. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Performance Asterisk large installation on Vmware/Xen
RdSS == Rafael dos Santos Saraiva rafaels...@gmail.com writes: RdSS I would like the opinion of you and if anyone has a similar scenario. I RdSS have a project for installation of a Asterisk server in a client with about RdSS 400 extensions. My question is whether this scenario carry an Asterisk RdSS virtualized. Will be used only extensions and trunks sip sip, 1 queue with RdSS 2 agents, without call recording. It is best to use XEN or VMware? Which RdSS best version of Asterisk for this scenario? I would use xen. Be sure to allocate enough resources to the asterisk domU. (In this case, the fact that the bare iron is partitioned by xen is no different, from the point of view of asterisk's performance, than running it on an unpartioned box along with other services. The more one squeezes onto a given box the more everything suffers. But partitioning the hardware with something like xen does not automatically break stuff. Were the box doing ISDN, on the other hand, routing the pci card to the right partition can be an issue. But for 100% sip it should just work.) -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and hylafax: how to debug ...
SN == Sebastian Niehaus nieh...@web.de writes: SN Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the SN same maschine: Hylafax fax server. I want hylafax to use t38modem (a SN virtual T.38 modem) for sending faxes. t38modem schould connect to SN asterisk on the same host. SN If hylafax sends a fax it should use t38modem which ist connected to SN asterisk. Asterik is expected to establish an outbound connection to my SN SIP provider which supports T38. The asterisk box is behind nat. Silly question: If you want to use T38 to the remote provider, and have t38modem, do you /need/ the asterisk in the middle? And if you /do/ need something between the two, might a sip proxy work better than a pbx? -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calendar: cert mismatch
PD == Phil Daws ux...@splatnix.net writes: PD It does generate a validity warning, as its self-signed, though I have PD added it to the PBX ca-bundle.crt. Am I right in assuming that PD Asterisk will use the default OpenSSL paths for where certificates are PD stored ? The error said that the hostname in the uri does not match (any of) the hostname(s) in the cert. Does the self-signed cert have the hostname in either the CN or in (any of) the dnsName(s) in the subjectAltName section? It might work better if you created a local CA and used that to sign an end-entity cert for each server which needs one. Then add that CA cert to the bundle. Recent versions of tls (claim to have) deprecated the idea of using self-signed certs for anything other than root ca certs, but you can always create your own CA. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fast AGI library/support for C C++
KD == Kashyap Darji kashyap@gmail.com writes: KD Thanks for your comment, but if possible then could you please give some KD reference link or example of the using the same what you have suggested. wikipedia has an OK page about xinetd: http://en.wikipedia.org/wiki/Xinetd and its homepage is at: http://xinetd.org/ Your distribution will have documented. And as an aside, the agi and fastagi frameworks for other languages document what one needs to do well enough to code a similar library in c, c++ or any other language. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fast AGI library/support for C C++
AJS == A J Stiles asterisk_l...@earthshod.co.uk writes: Please let us know how to write FastAGI using C/C++. AJS You don't need it! C, being a compiled language, doesn't suffer from AJS interpreter overheads and therefore doesn't require such bodgery. You can also use inetd(8) or xinetd(8) to handle the tcp side of things; it can call your AGI app whenever asterisk makes the tcp connection, and keep it open for future calls. Then, just use stdin and stdout as you would for a normal AGI app. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove the call waiting tone without disabling callwaiting?
NB == Niccolò Belli darkba...@linuxsystems.it writes: NB If someone knows how to COMPLETELY REMOVE the fucking beep please NB let me know: there are already tons of phones ringing everywhere so NB there is no need for an annoying beep. Edit chan_dahdi.c. The my_callwait() and/or dahdi_callwait() functions may be enough. You want to skip the silence and tone creation steps. Or perhaps #defining CALLWAITING_REPEAT_SAMPLES to 0 might work. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk won't start - trap invalid opcode
DT == Duncan Turnbull dun...@e-simple.co.nz writes: DT I have a new install of asterisk 1.8.8.1 on ubuntu server DT 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux DT The only errors I can see are limited - I also stopped wan router and dahdi and I still get DT ~# asterisk -cvv DT Illegal instruction What does /proc/cpuinfo say? (Just the first chunk is enough.) Try running asterisk is gdb: :; gdb asterisk (gdb) run -cvvddd When it dies, try: (gdb) bt full (gdb) disasemble /m You may also want to recompile asterisk after turing on: DONT_OPTIMIZE DEBUG_THREADS BETTER_BACKTRACES in the Compiler Flags section of make menuselect. The gdb output if you do that may be more comprehensible. Either way run gdb from the asterisk src directory. When you find the point where it crashed, you can discover what the illegal instruction is. I suspect your compile may expect a more recent cpu than you have, and may use sse instructions which it doesn't support. A disassembly around the failing instruction will confirm whether that is true and which instruction it is. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX over HTTP
Rob == Rob Scott [EMAIL PROTECTED] writes: Rob For work environments where you only get HTTP or HTTPS access, Rob what is the feasibility of doing IAX over HTTP? Tunnelling tcp/ip over http/(tls/)?tcp/ip is viable, tunnelling rtp/udb/ip or iax/udp/ip over http/(tls/)?tcp/ip however will only work reliably if the tcp doesn't see any packet loss. Else it will retransmit lost packets and the voice quality will suck. That said, if you can get a http or https socket going you can probably also tunnel over dns. So you may want to look into ip over dns/udp/ip tunnels for rtp or iax. -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM to g729 Conversion
Matthew == Matthew Boehm [EMAIL PROTECTED] writes: Matthew There is no way to convert existing files to g729? The reference codec has a cli to do that. It converts from raw 16-bit signed linear files (sox filetype sw) to g729 files that should work with *'s format_g729. I beleive it is even licenced w/o royalties for non-real-time one-off conversions of files, but check the docs in the archive to be sure. Cf: http://www.itu.int/rec/recommendation.asp?type=itemslang=eparent=T-REC-G.729-22-I!AnnC%2B Price is CHF 137 but I beleive you can register to dl a couple of their standards for free, possibly including G.729 Annex C+. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
Bill == Bill Seddon [EMAIL PROTECTED] writes: Bill My use of sox for down sampling is limited to Bill this kind of command: Bill sox in.wav -r 8000 out.gsm You really want to use the polyphase app in sox for resampling. It is significantly slower than the other options, but that is irrelevant here. So try: sox in.wav -r 8000 out.gsm polyphase Use out.sl for a slinear file (then rename it to whatever.snl), out.ul for a mu-law file, out.al for an alaw file, etc. Cf show file formats at the * cli. If you are mostly sending the audio out over a zap channel, you may as well use ulaw or alaw -- whichever the pstn in your area uses. If primarily voip, you can match whichever codec you use or just make it slinear (.sw in sox; .sln for *) and convert it on the fly. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VM access
Larry == Larry Shields [EMAIL PROTECTED] writes: Larry On most VM systems you can press the * key or # key to get a Larry login prompt during your greeting. Is that not possible with Larry this system? If you hist * during the outgoing message you'll get sent to the a extension, if that exists in the current context. Eg: OPERATOR = 2121 ; dump them to vm exten = s,1,VoiceMail2(2345) ; if they enter 0 exten = o,1,Dial(OPERATOR) ; if they enter * exten = a,1,Goto(ivr|2345|1) -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More on Broadvox
Rich == Rich Adamson [EMAIL PROTECTED] writes: Rich If they are suggesting the sip negotiation process is trying to Rich negotiate something like silence-suppression=off, and their Rich equipment won't handle _anything_ other then Rich silence-suppression=on, then that sounds like a short-coming of Rich their equipment. This is incidently a know issue with cisco gateways. They can only configure vad per box rather than per call. And given that most customers of sip providers willb e using sip equipment that supports it, and most of thost customers will want to bandwidth savings it is as sure a bet as anything that sip providers using cisco will always have vad turned on. Asterisk just has to cope in these cases. To the OP: grep for silence in chan_sip.c; try commenting that line out and see whether they like that better -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound file quality
Christoph == Christoph Rothe [EMAIL PROTECTED] writes: Christoph Which Formats will * accept and what extensions may Christoph be used? Is there a page in the wiki about that ? Look in the formats dir in the asterisk src. Each of those formats can be used. They are well documented in terms of what file types and filenames they require. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound file quality
David == David Gurr [EMAIL PROTECTED] writes: David As a result, I'd like to ensure that the voice prompts I'm David using have the best possible audio quality. David My callers will be coming in over PSTN to a VoIP gateway and David then to me by uLaw/aLaw ... The optimal quality in the case where pstn is involved would be from: Using pro-quality (which these days does not necessarily mean pro $$$) recording equipment Recording in DAT quality (16 bit 48 kHz) the recording equipment is more likely to support that than 16 kHz or 32 kHz 32bit float rather than 16 bit int is ok, too Edit the files at this point for lead time, trail time, equal volume, et al. Use a high quality resampling algorithm (in sox use polyphase) to resample to signed-16bit 8 kHz. Optionally use a band-pass filter here to drop stuff outside of the PSTN frequenc range. If you only do one of alaw/ulaw, you might as well convert the files to that, else leave them as signed-16bit You can still get things like phase distortion if the path has jitter and the receiver does not jitter-buffer. You will also need to do some experimenting to determine the optimal amplitude to avoid both clipping and too-little use of the available u/a-law bandwidth. -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfering incoming message from app_queue
Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing anything like that?) -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
