[asterisk-users] IAX/NEW delays
Hi, Could someone tell me where are the good places in chan_iax to put trace points when I experience strange delays in NEW processing? I tried to output some debug after every stage of socket_process / case IAX_COMMAND_NEW, but it all takes max 30ms. However, sometimes in a normal call I get an extreme delay though (around 10 seconds), if I can believe the timestamp field... Do you know any good places to look at in ast-1.4? Or maybe someone had this problem before and can tell me what can cause delays that long. Thanks, Stan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to convert *.wav files ?
2009/9/26 hadi motamedi : > I need to convert the original *.wav sound files (their file attribute is > reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice > quality . That's useless. You can do that of course, but even if you reencode the file, the quality of sound itself will not change. Also, unless you use a wideband codec, you won't be able to send more than 8000Hz over the line for most standard codecs (u/alaw, gsm, etc.). Just play the .wav you already have - you cannot get a better quality than that in the original file. -- KTHXBYE, Stanisław Pitucha, Gradwell Voip Engineer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
2009/9/14 Olle E. Johansson : > Make sure that each device has a TRANSFER_CONTEXT dialplan variable. What about a situation where sip devices register at a proxy in front of many asterisks and asterisks authorise all calls from that proxy? I.e. I don't have any devices that asterisk would know about. That way as far as asterisk is concerned, the call is a simple trunk call and the B side (in A->B call) doesn't trigger any TRANSFER_CONTEXT setting when doing a transfer. I hacked together a solution that works for me now, but I'd rather solve this problem properly. My solution was that the A->B call gets out to the device via "rB" context. When A does a transfer current.chan1 (in handle_refer) has CALLERID(num) set to "rB". When B transfers, callerid is obviously A. So I just copy that value to some variable in the new channel and bill based on that in a common transfer context. Still - I'd rather find a solution that doesn't involve patching chan_sip... (and doesn't require me to set up sip users on all asterisks). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
2009/9/14 Matt Riddell : > For every billable item we use a code for the account and store it in... > > accountcode :) I'm not sure that actually answers my question... If you have a A->B call and set accountcode for A on it, then B does a blind transfer, how do you set the correct accountcode then? (assuming B is a different customer and blind-transfers you to pstn) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
2009/9/9 Stanisław Pitucha : > I've got different customers that may use the same asterisk. Each user > can blind-transfer a call to whatever place they want. But of course > the transferring side should be billed for it. > What can I do to see the difference between the channels here? Trying again, since I didn't get any responses... but someone has to know the answer ;) I know I can get the channel name via BLINDTRANSFER, but that doesn't really help me. What I need is either the CALLERID(num) if the calling side initiated the transfer, or either EXTEN or some other custom variable from the calling leg if it was the called party that does the transfer. Has anyone solved this billing problem in any way? (well - any apart from the asterisk-generated cdr-s - I don't really want to start relying on them) Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfers security
Hi, I've got different customers that may use the same asterisk. Each user can blind-transfer a call to whatever place they want. But of course the transferring side should be billed for it. What can I do to see the difference between the channels here? If there is an A->B call going on, I'd like to know which side did the transfer - but whichever side does it, I get back to context 'default'. Any ideas? -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote attended transfer
Hi, I'm having problems with sip remote attended transfer using 2 asterisk boxes (same version, latest 1.4.X). Whenever I transfer from a call from box A to a call on box B, one call leg of the transferring phone is not disconnected (the one that is normally dropped by server side, phone disconnects the other one). The same situation works perfectly with local attended transfer. Is anyone using remote attended transfers with asterisk? Does it work for you? Do you use any workarounds? I'm asking here, because it would be strange if that functionality was broken since 1.4.8 and noone noticed ;) Exact scenario I'm using is described in the bug: https://issues.asterisk.org/view.php?id=15833 Thanks for any help. -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com Gradwell – Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - log rotation [solved]
2009/9/4 Olivier :
> From an off-list comment, I think the explanation is :
> - some apps are opening and closing log files before and after each writing,
> - some are leaving log files open.
>
> When a log file is currently rotated, both apps can't append anything
> anymore to log files.
That's not correct (on a *nix system anyways). In the second case, if
some app holds a descriptor of an open file and you move the file
around in the same filesystem, nothing changes. You can rename 'log'
to 'log.1', but it doesn't change the descriptor and the application
will still write to the file (it's just named differently).
The logger restart just does a { close(); open(); } and you're logging
to a new 'log' again. This way you don't lose any messages during the
rotation.
--
Kind regards,
Stanisław Pitucha, Gradwell Voip Engineer
T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com
Gradwell – Internet for Business People
Phone Services | Business Broadband | Email & Website Hosting
Can switching to VoIP today put some change in your pocket?
Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company
Number: 3673235
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Re: [asterisk-users] Bria / eyebeam: no RTCP while on hold
2009/8/26 Paul Herman : > I use Bria and eyebeam and it seems that asterisk doesn't send RCTP > keepalives when a SIP channel is on hold. Slightly related: https://issues.asterisk.org/view.php?id=15466 It also affects integration with OCS for me. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?
Probably the easiest way: put an opensips box in front of asterisk. It can handle multiple registrations on the same username. If you have multiple registrations, it will do a parallel fork and work just like you wanted. You just have to make sure that phones register on opensips, not asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to match "no callerid" in 1.6 ?
2009/7/24 Louis-David Mitterrand :
> This used to work fine in 1.4:
>
> exten => 2131/,1,NoOp(reject3: ${CALLERID(num)})
> exten => 2131/,n,Playback(no_unknow_callerid_here)
> exten => 2131/,n,Hangup
>
> And now, after upgrading to 1.6.1.x it matches every callerid.
I'm not sure if it's the same reason, but have a look at this bug
(exists in 1.6.1.1):
https://issues.asterisk.org/view.php?id=15476
Whether you want to use the function, or a pattern match in version
1.6, you might want to upgrade past revision 206705. Especially if
you're trying to detect no callerid. Otherwise you'll get a wrong
result. (assuming you're using SIP)
HTH
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[asterisk-users] Rtp keepalive
Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all combinations of nat and qualify for the peer that has problems - rtp comfort noise is simply not sent. After trying to make it work for a day or so, I reported it as a bug (https://issues.asterisk.org/view.php?id=15466) but maybe someone here has some ideas how to make it work? -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 851 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com Gradwell - Internet for Business People Phone Services | Business Broadband | Email & Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idle threads
Hi, I noticed something bad happening on our systems lately. We have lots of asterisk threads running, but most of them are completely idle - strace doesn't show anything happening there. The only thread doing work seems to do everything. I see it sending mysql queries, writing logs, sending both SIP and RTP, etc. etc. That doesn't seem right. This system uses a lot of AGI scripts, mysql registration and dialplan - sip only. What can I do to split that load? And what might be the reason that all processing is done by one thread? Thanks, Stan (sorry if this mail arrives twice, my mistake) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec not in channel variables
- "michel freiha" <[EMAIL PROTECTED]> wrote: > Did you try "show translation" That shows a table of times taken by translation... I'm asking about codecs used by a channel on a certain call. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] codec not in channel variables
Hi,
I'm trying to access audionativeformat / other codec variables in the hangup
handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response.
Also 'core show channel ...' doesn't list those variables. Are they always set
by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio
passing through asterisk, same codecs on both sides.
I see that with ast-1.4.11.
Thanks for ideas,
Stan
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[asterisk-users] meetme + jitter buffer
Hi, I was wondering if there's any sense in increasing audiobuffer above the minimal '2' in meetme, if every channel is already dejittered before (Local/.../nj - as described at: http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) Will it help in anything, or just increase delay? Thanks, Stan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] C450 broken rtp handling
Hello, I've got a problem with rtp handling by siemens c450 and similar. I experience a couple seconds of silence between early media and normal call (normal call's rtp is dropped by phone). This is caused by SSRC changing (even though marker bit is set). I have all relevant patches applied - it still happens on 1.4.21.1 and every version before that. Especially http://bugs.digium.com/view.php?id=12570 doesn't change anything, because call is p2p bridged. Issue can be fixed by forcing use of the same ssrc in ast_raw_write and bridged rtp writes and my custom patch works, but I don't want to use it if it can be done in some other way. Is there a way to force treating outgoing rtp as one stream, instead of switching source after early media? Is there a way to do it without resigning from p2p bridging? Thanks for ideas, Stan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Asterisk on to CentOS 4
Add /usr/sbin to your PATH, or run /usr/sbin/asterisk. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sipp scenario for asterisk sip
Hey I'm looking for an advanced scenario for sipp, that can be used for testing asterisk. Mainly I'm interested in making random calls between sipp pseudo-users. Did anyone try to do something like this? Or has anyone got an example scenario with working loops? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan-capi in 1.4.10.1
Hey I just got chan-capi "ported" to *-1.4.10.1 by changing channel allocation in capi_net to: --->8--- tmp = ast_channel_alloc(0, state, i->cid, "", "", "", "", 0, "CAPI/%s/%s-%x", i->vname, i->dnid, capi_counter++); --->8--- Calls seem to work - did I break something big? Anyone else tried that before? Stanisław Pitucha Gradwell Dot Com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why is nonce="584760da" used in sip packets?
- "Rizwan Hisham" <[EMAIL PROTECTED]> wrote:
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="584760da"
> Authorization: Digest username="bernart48", realm="asterisk", algorithm=MD5,
> uri="sip:[EMAIL PROTECTED]:9060", nonce="584760da",
> response="948d3923bf2df47eca17c572713af2c7", opaque=""
> What i dont know, and would very much like to know, is what is the
> purpose of this parameter in sip packets?
It's kind of challenge algorithm. What you see in "response" is not
MD5(password), but MD5('password', 'realm', ..., 'nonce'). Nonce is generated
by server so that you don't get the same hash for for every authorization by
that user. It prevents someone who can see only one way communication from
breaking your sip session + makes breaking hash a little bit harder.
Nonce should be unique per authorization.
If nonce wasn't used you could reuse the same response in next connection even
if you don't know the real password.
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Re: [asterisk-users] Lightweight IAX balancer
Very low chances for that module if any. I haven't been using OpenSER much and I don't think I'll be using it soon - but who knows. Let's hope that implementation will be clean enough to turn it into a library easily if someone else wants to do it one day. So far another pack is available at previous link (http://www.gradwell.com/tmp/iax_proxy.tar.gz) with correct license banners (MIT). Plans for near future are: - stability - speed - POKE'ing and automatic dis/en-abling servers for balancing - for now you can reload list at runtime with SIGUSR1 without dropping calls - maybe some proper site for project... so I won't spam this maillist --- Stanisław Pitucha Gradwell Dot Com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lightweight IAX balancer
- "Tzafrir Cohen" <[EMAIL PROTECTED]> wrote: > Interesting. One thing thoough: what's the license of your code? It's MIT - I forgot to add that. I'll stick the banners to files soon, with next update to the package. (along with some fixes, etc) Stanisław Pitucha Gradwell Dot Com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Lightweight IAX balancer
Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on each machine. It does balancing not per IAX connection, but per call - rewriting call numbers and keeping track of connection status. It's going to be optimized for speed - doesn't do any other modification or audiostream translation - only message passing. If someone's interested -- code + short doc is available at http://www.gradwell.com/tmp/iax_proxy.tar.gz Development will continue - any opinions / comments / contributions are appreciated. Stanisław Pitucha Gradwell Dot Com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
