Re: [asterisk-users] AMR codec for Asterisk 1.6.1.X
Dear list, i have re-compiled again the source code of amr patch for 1.6 (https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/asterisk-1.6-AMR.patch) The patch does not compile with the static function into frame.c called : static int amr_samples(unsigned char *data, int datalen) i have removed the static and used like int amr_samples(unsigned char *data, int datalen) Anyone else got this issue ??? In this way the patch compile . It also show right format name when i try lo load codec_amr.so load codec_amr.so The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format amr to slin, cost 2000 == Registered translator 'lintoamr' from format slin to amr, cost 17997 Loaded codec_amr.so = (AMR Coder/Decoder) Also i have into the config file asterisk.conf the following value to filed transcode_via_sln = yes so transcode_via_sln = yes If i try to make a call to echotest by dialing 600 '600' = 1. Answer() [pbx_config] 2. Playback(demo-echotest) [pbx_config] 3. Echo() [pbx_config] 4. Playback(demo-echodone) [pbx_config] with a client that have only enabled amr codec i got this output: [May 6 17:51:11] WARNING[9684]: chan_sip.c:7654 process_sdp: Unsupported SDP media type in offer: audio 4002 RTP/SAVP 114 18 113 0 8 101 Anyone know how to get this AMR codec doing transcoding on asterisk 1.6? Many thanks in advantage Andrea Il 05/05/2010 18:13, Adrian Marsh ha scritto: It says in the readme from that link you provided: This patch adds AMR-NB support to Asterisk 1.4 (for Asterisk 1.6 check out asterisk 1.6 branch and use the asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov)) Did you use the 1.6 branch and patch ?? I'll have to try this myself at some point. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrea Cristofanini Sent: 05 May 2010 14:22 To: asterisk-users@lists.digium.com Subject: [asterisk-users] AMR codec for Asterisk 1.6.1.X Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats
[asterisk-users] AMR codec for Asterisk 1.6.1.X
Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*CLI core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats version debbi*CLI core show co codec codecs config debbi*CLI core show code codecs codec debbi*CLI core show codec codecs codec debbi*CLI core show codec audio Usage: core show codec number Displays codec mapping debbi*CLI core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) debbi*CLI The CLI does not show codec audio or codedc translation for AMR NB. Anyone have any idea ?? Thanks in advantage Andrea -- _ -- 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] 1.6.0 verses 1.6.2
This sound strange, i have running on asterisk 1.4 1090 calls with no problem 24/24 h. (24 giga ram 4 dual core xeon ) Maybe is the configuration or configuration tuning missing in somewhere. Andrea Il 14/04/2010 10:45, Tzafrir Cohen ha scritto: On Tue, Apr 13, 2010 at 04:25:49PM -0600, John Rose wrote: Why do versions 1.6.2 and 1.6.1 use much more CPU resources that 1.6.0? I can get 400+ SIP/G.711 calls running on this dual core box with 1.6.0 but the cpu maxes out and core dumps at approx. 180 calls when version 1.6.1/2 is running. Could you please be more specific? Have you tested those versions on the same system? Can you reproduce those results? What exact versions? On what system? -- --- Andrea Cristofanini Chief Technical Officer ZeroZero39 srlTel: +39 02 61294759 Viale Brianza, 20 Fax: +39 02 87365813 20092 Cinisello Balsamo (MI) Mob:+39 329 1871756 web: www.zerozero39.it PGP Key: http://www.zerozero39.it/cristofanini.asc -- _ -- 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] ISDN Remote HOLD
Hi there, i'm using dahdi to manage a B400P openvox BRI card. All works as expected, i would like to know if there ia a way to put the call in REMOTE HOLD, like pressing R button on ISDN phone. This can be done by CAPI using the proper application , It is implemented on DAHDI ? Regards Andrea -- --- Andrea Cristofanini Chief Technical Officer ZeroZero39 srlTel: +39 02 61294759 Viale Brianza, 20 Fax: +39 02 87365813 20092 Cinisello Balsamo (MI) Mob:+39 329 1871756 web: www.zerozero39.it PGP Key: http://www.zerozero39.it/cristofanini.asc ___ -- 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] Asterisk GSM Gateway Project
Hi There http://www.2n.cz/company/2n_history.html offer this kind of products. they works vey well with asterisk. Ciao Andrea Michael Graves ha scritto: On Mon, 23 Jun 2008 10:09:21 -0400, Steve Totaro wrote: On Mon, Jun 23, 2008 at 9:57 AM, [EMAIL PROTECTED] wrote: The quad-band model is around $250 USD. See Ebay auction here http://tinyurl.com/5tvoa9 Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Do any of these do SMS? Yes, most of them do. See #7 on the following page: http://www.portech.com.tw/eweb/index1.htm Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- 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 ___ -- 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] AGI after Hangup
You have to run DeadAGI, in h . Regards Andrea Cristofanini voip crazy ha scritto: Which is the way to run an AGI after hangup a call? The problem I have is when the call dies the AGI dies too I try the Dial command g option, but it does not work for me Any clue will be welcomed. Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on iPhone
http://www.mgamble.ca/oss/iphone_asterisk/ ___ -- 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] Asterisk on iPhone
Hi I just saw this now ! does the microphone and speaker works ? Can you use it like softphone for recive calls ? Regards Andrea C F ha scritto: TODAY I have managed to hack the iPhone and install Asterisk on it. Detailed instructions to follow. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cosini iAN7s
Cosini SS7 work for 99% of all cases. Femi ha scritto: Hi, Has anyone on this list used the Cosini iAN7s SS7 gateway with Asterisk? Please let me know how it performed and what issues you faced Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 320 Lost Settings
yes i see you have to enable in ADVANCED SETTING Challenge Response on Phone: = OFF Regards /a Matt Riddell ha scritto: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Has anyone ever seen an Snom320 lose settings? It's been working fine for months and then I got a call this morning saying that it was asking for country, timezone etc. I logged in remotely, and it had lost the server address, username, password, mailbox and ringtone. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHl6vCDQNt8rg0Kp4RAospAJ9DUNge64n7u3RkQWsodHgdOS/higCgwNFy VfZUUNJIgzeC4Hy5vg0f+mY= =tpnK -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX softphone
Try zoiper bilal ghayyad ha scritto: Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set CDR userfield in a realtime dialplan
Hello
I'm running the same with no-problem.
CDR(userfield)=INCOMING
We use also a quite nice patch from Matt Riddell
http://bugs.digium.com/view.php?id=9424
that allow to have extra userfield.
CDR(userfield2)=${CODEC-IN}
CDR(userfield3)=${CODEC-OUT}
and so on...
This is quite good for custom report.
/a
Benchev ha scritto:
On Wednesday 09 January 2008 09:54:59 Yves Räber wrote:
Hello,
I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have
some trouble with the CDR userfield that is not changed when using the
SET command in the realtime dialplan.
In my dialplan (extensions.conf, the file) I'm setting the userfield
like this :
exten = s,n,Set(CDR(userfield)=X)
Later, my dialplan switches to the realtime part and this is an extract
for what is inside :
===
id | context | exten | priority | app | appdata
===
12 | script | s | n| SET | CDR(userfield)=Y
===
I can show that the command is executed :
-- Executing Set(SIP/siemens1-081ca290, CDR(userfield) = Y)
But in my CDR, the old value is saved (X in this case).
Into a database the line exten = s,n,Set(CDR(userfield)=X)
should be enterd as:
context | exten | priority | app | appdata
means in your case:
your_context| s|n|Set(CDR(userfield)|X
Boyko
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Pcengines Alix board
Hi there we have astlinux running on alix board, it is awesome. Andrea Giuseppe Barichello ha scritto: Date: Mon, 19 Nov 2007 10:39:31 -0600 From: Bob Pierce [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk on Pcengines Alix board To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Sun, 2007-11-18 at 22:14 +0100, Giuseppe Barichello wrote: I have successfully compiled and installed Asterisk on an Alix board (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian variant). I'm using it at home for a month. That's very interesting! I've been curious about trying this. Did you run across any challenges getting this setup? Two main issues: 1) Understanding how voyage linux configures read-only and rw mounts (I wanted to mount all /var tree as rw) 2) Getting MOH play MP3 sound files with Debian standard packages: I had to recompile Asterisk from source to fix it. Giuseppe ___ --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 __ Informazione NOD32 2674 (20071121) __ Questo messaggio è stato controllato dal Sistema Antivirus NOD32 http://www.nod32.it -- Cheers Andrea Andrea Cristofanini CTO - VoIP Gedam Europe S.r.l. - (Torino,Italy) Gedam Advanced Communication Ltd - (Dunedin,New Zealand) Strada da Bertolla all'abbadia di Stura, 151 - 10156 Torino - IT GSM. +39-329.1871756 - PSTN. +39-011.19824516 - FAX. +39-011.8338622 - http://www.gedameurope.com/ http://freevoip.gedameurope.com/ http://www.faropbx.com/ http://www.pbxsolution.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
Re: [asterisk-users] CDR
___ --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 -- Cheers Andrea Andrea Cristofanini CTO - VoIP Gedam Europe S.r.l. - (Torino,Italy) Gedam Advanced Communication Ltd - (Dunedin,New Zealand) Strada da Bertolla all'abbadia di Stura, 151 - 10156 Torino - IT GSM. +39-329.1871756 - PSTN. +39-011.19824516 - FAX. +39-011.8338622 - http://www.gedameurope.com/ http://freevoip.gedameurope.com/ http://www.faropbx.com/ http://www.pbxsolution.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
Re: [asterisk-users] CDR
Thank you very much ! Andrea Atis Lezdins ha scritto: You have to revert patch for issue http://bugs.digium.com/view.php?id=10659 More specific - that is - remove those two lines from main/cdr.c if (cdr-disposition AST_CDR_ANSWERED (ast_strlen_zero(cdr-channel) || ast_strlen_zero(cdr-dstchannel))) continue; /* people don't want to see unanswered single-channel events */ Regards, Atis Andrea Cristofanini -- [GedamEurope] wrote: Hi there ! So it is possible to have logged a single unanswered channels ? What sould be stted into the cdr.conf ? Regards Andrea Steve Murphy ha scritto: Sorry! I've gotten some complaints on this; I will try this week to mod 1.4 so that you can choose to see the single-channel unanswered CDR's, in a new config file option. I've gotten complaints both ways, tho, so pardon me if I get a little confused about what users out there want from CDR's. My biggest trouble is that by forcing all channels to have a CDR at creation time solves problems with missing CDR's, but creates a problem by generating extra unanswered CDR's that weren't generated before... for instance, when you ring three phones via a dial command, you then get 3 CDR's, including the two phones that were rung, but not answered. Another problem is with Zap-based phones; you take the handset off-hook, and a channel is created and dialtone generated. If you hang up, you get a CDR there, also. I have not found an easy way to detect and drop these kinds of CDR's, as most folks really do not find them very useful. And, I've gotten a complaint that you end up with 'duplicate' CDR's, which is also an artifact of forcing all channels to have a CDR associated. If anybody thinks they have a magic spell that will calm down the CDR's, I will not mind the information at all! I worked all last week to try to iron out the 1.4 zap-transfer CDR issues. I have 12 cases I test with involving hook-flash and #-blindxfers, and so far, I've got 9 of the 12 working OK (as far as I'm concerned.), but I have 3 cases that come up with problems. For instance, if you hookflash, and dial a number, the CDR's will be different, if you hang up before the dialed party answers, versus hanging up afterwards... The diff between a blind xfer and an attended xfer (without the 3 way), I guess, but I lose the calling channel name... I'll try to sort all this out, and then I'll attack this problem. Hopefully, I get it all into svn before the next release of 1.4...! As far as xfers in 1.4 go, I'm trying to make sure that the source and destination channel names reflect the true dialing party, as this makes more sense from a billing perspective, at least to me. So, if A calls B, and B forwards the call to C, then the CDR's need to reflect a call from A to B, and a call from B to C, which you may or may not be seeing right now. AFAICT, transfers pretty much result in confused CDR's. I gave up totally on generating separate CDR's for any 3-ways that might occur. Such 3-ways will end up being billed to the dialing parties. Here's an interesting situation: A calls B, A then hookflashes, and then A calls C, and hookflashes again. It's now a 3-way call, between A, B, and C. A then drops out and B and C converse. My goal with this situation was to have two CDR's, one for A-B and one for A-c. Since A placed both calls, it seems only just that A pay for B's and C's conversation. Especially if A is in the US, for example, and B is in Uganda, and C is in Bangladesh! Getting the right info in the right field from the driver level is pretty tricky, and you can add the fact that there's definite timing issues at play. If I make changes to CDR's in channels A and B, near to the time a hangup occurs (and it's very commonly the case), I can end up with some pretty strange stuff happening! I found that adding debug logging statements to the driver can affect the way the CDR's are generated! I solved some of this by locking channels (which to me is pretty ugly, considering the number of locks involved). So, please, cut me some slack... and keep me informed about your happiness levels. I want this stuff to be good, solid, and useful to the majority. But also keep in mind that I fix something, someone out there is going to be unhappy, because they have code/backends/whatever that depends on the borked behavior! 1.4 ***IS a release, and I dare not do ANYTHING but fix bugs. But, wow, fixing a bug changes behavior, and my goal is minimal impact, but real and useful fixes. murf On Mon, 2007-10-15 at 10:40 +0200, Andrew Nowrot wrote: Hi Thanks for reply Yes, there's a change. For me it's completely unacceptable, so i reverted the patch (http://bugs.digium.com/view.php?id=10659). For me too. This bug occur in September. Is it still present in asterisk 1.4.12.1. I also have asterisk 1.4.4 on a different box
Re: [asterisk-users] Queue PROBLEMS
Dear all
I have found this on Queue :
I send calls to PSTN numbers, i set some a variable in the channel, like
CALLERID(name)=${recordid}, and i send the answered calls to a Queue .
In the softphone i read callerid(name) and do some action on CRM.
All is sweet till here...
The problem come when i have CALLS WAITING in the queue, all the agent
are busy, after some times some Agent began available again and the
call is passed to an Agent, in this case callerid(name) began blank, so
look like that i have lost my variable.
Any idea ??
--
Cheers Andrea
Andrea Cristofanini
Gedam Europe Srl
Gedam Advanced Communication Ltd
Torino, Italy
C.so Re Umberto 21
Mobile : + 39 329 1871756
PSTN: + 39 011 19824516
FreeVoip: 6838601
http://www.gedameurope.com
http://freevoip.gedameurope.com
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MaxRetries:10 - Problems Dialout Call files grabbing end of RetryTime
Hi List I use calls file with WaitTime and RetryTime I need to understand when last retry is reached, this because some other work have to start. Is there any variable containig it in the Channels? Regards Doug Lytle wrote: Marco Mouta wrote: Hi all, I'm working with dialout call files and i've noticed that with MaxRetries: 1 ,many times the call is already established successfully and asterisk dials a second call. I was having the same issue until I schedule the dialout a few minutes after the .call file creation. I use the following: cat vm-callout.sh #!/bin/sh cd /usr/local/bin /bin/touch /usr/local/bin/$1.call /bin/touch -r /usr/local/bin/$1.call -m -F 150 /usr/local/bin/$1.call cp --preserve=timestamps /usr/local/bin/$1.call /var/spool/asterisk/outgoing/ Doug -- Cheers Andrea Andrea Cristofanini Gedam Europe Srl Gedam Advanced Communication Ltd Torino, Italy C.so Re Umberto 21 Mobile : + 39 329 1871756 PSTN: + 39 011 19824516 FreeVoip: 6838601 http://www.gedameurope.com http://freevoip.gedameurope.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Outgoing Spool Failed with ViciDial (MattF?)
lol Matt Riddell (IT) wrote: Arun Kumar wrote: hi thanks for reply. I'm using vicidial to make calls at 2.0 dial level it is able to make calls but when I see the asterisk -r most of the time it shows Outgoing Spool Failed. Which Spool File ? Er, probably the best place to ask would be the VICIdial forum on mattf's website, unless he wants to chime in? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers Andrea Andrea Cristofanini Gedam Europe Srl Gedam Advanced Communication Ltd Torino, Italy C.so Re Umberto 21 Mobile : + 39 329 1871756 PSTN: + 39 011 19824516 FreeVoip: 6838601 http://www.gedameurope.com http://freevoip.gedameurope.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime sip_buddies does not store ip address
Hi list i use SVN branch , i have real time working good with IAX2 The problem i have is for sip_buddies , any SIP acount register does not store ip addres inside the table. This only for SIP iax2 works great. i also have in sip.conf rtupdate=yes any ideas ? -- Cheers Andrea Andrea Cristofanini Gedam Europe S.r.l. Gedam Advanced Communication LTD mobile : +39 3291871756 office : +39 011 5694900 freevoip : 6838602 MSN : [EMAIL PROTECTED] http://www.gedameurope.com http://www.asterisknews.it http://freevoip.gedameurope.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CrystalFontz LCD display
Hi there our company can provide custom integration with every kind of LCD display Andrea Giovanni Miano wrote: http://lcdsmartie.sourceforge.net/ Cheers, Giovanni Miano 2006/1/2, Matt Riddell [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Yes, we do development under Linux for this. Was there some particular support you were after? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cheers Andrea Andrea Cristofanini Gedam Europe S.r.l. Gedam Advanced Communication LTD mobile : +39 3291871756 office : +39 011 5694900 freevoip : 6838602 MSN : [EMAIL PROTECTED] http://www.gedameurope.com http://www.asterisknews.it http://freevoip.gedameurope.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Asterisk-User] linux/Asterisk change ip address
Hi list i have a Asterisk box that use 10 phone with sccp, and some iax2 Every 8 10 hours , my linux machine change ip address and route, and the cisco and iax phone cannot see the server ... What can do that? there are no other linux box , no any pc that provide DHCP -- Cheers Andrea Andrea Cristofanini Gedam Europe S.r.l. Gedam Advanced Communication LTD mobile : +39 3291871756 office : +39 011 5694900 MSN : [EMAIL PROTECTED] http://www.gedameurope.com http://www.asterisknews.it http://freevoip.gedameurope.com ___ --Bandwidth and Colocation sponsored by Easynews.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] IAX2 Softphone Quality Network Cards
tourn of AGC , and mybe use GSM. for usb device that use iax2 prtocol there are this one that have nice http://www.gedameurope.com/us/002servizi_e_prodotti%5Bus%5D.htm this usb device doe not need external sound card. Philipp von Klitzing wrote: Hi! We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. Have you tried a different sound card and/or a USB handset (which includes an external sound card)? And what exactly do you mean with terrible sound? Philipp ___ --Bandwidth and Colocation sponsored by Easynews.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 __ Informazione NOD32 1.1202 (20050825) __ Questo messaggio è stato controllato dal Sistema Antivirus NOD32 http://www.nod32.it -- Cheers Andrea Andrea Cristofanini Gedam Europe S.r.l. Gedam Advanced Communication LTD mobile : +39 3291871756 office : +39 011 5694900 MSN : [EMAIL PROTECTED] www.gedameurope.com www.asterisknews.it ___ --Bandwidth and Colocation sponsored by Easynews.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] Uk Caller id
that's correct :-) Bruno De Luca wrote: this is an italian code and works... try it. [channels] ; -- canale 4 -- language=en faxdetect=both musiconhold=default group=2 canpark=yes context=inbound signalling=fxs_ks usecallerid=no ; echo cancel echocancel=128 ; range from 32 to 256(=echo 100%) echocancelwhenbridged=yes ; yes = 400 msec echotraining=200 channel=4 Bruno De Luca Graziosi Chris Thompson wrote: Hi I have my new TDM400P installed and working. I'm running from cvs HEAD with a 2.6.12 kernel on debian. I can't seem to get Caller id working (in uk with clid supplied and working to line) but am a bit unclear on the docs and hence assume it is something I am doing wrong. I would really* appreciate if anyone could take a look below at my zapata.conf and see is there anything incorrect. I am least convinced on the usecallerid=uk option, but if set to 'yes' i get Jul 22 15:38:47 ERROR[19569]: callerid.c:266 callerid_feed: fsk_serie made mylen 0 (-20) Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5796 ss_thread: CallerID feed failed: Success Jul 22 15:38:47 WARNING[19569]: chan_zap.c:5840 ss_thread: CallerID returned with error on channel 'Zap/2-1' :: zapata.conf :: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=uk callerid=asreceived cidsignalling=v23 cidstart=usehist callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 immediate=no progzone=uk musiconhold=default ; incoming channels signalling=fxs_ks group=2 context=incoming channel = 1-2 ; outgoing channels signalling=fxo_ks group=1 context=outgoing channel = 3 Thanks loads Chris ___ 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 BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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 __ Informazione NOD32 1.1175 (20050721) __ Questo messaggio è stato controllato dal Sistema Antivirus NOD32 http://www.nod32.it -- Cheers Andrea Andrea Cristofanini CEO Gedam Europe S.r.l. CEO Gedam Advanced Communication LTD mobile : +39 3291871756 office : +39 011 5694900 www.gedameurope.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] Dell Hardware
We use Supermicro and we have NO problem at all :-) Bruno De Luca wrote: We are using this combination. we are thinking about change the DELL computers! Bruno De Luca Graziosi Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ 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 BRUNO DE LUCA GRAZIOSI Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ 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 __ Informazione NOD32 1.1175 (20050721) __ Questo messaggio è stato controllato dal Sistema Antivirus NOD32 http://www.nod32.it -- Cheers Andrea Andrea Cristofanini CEO Gedam Europe S.r.l. CEO Gedam Advanced Communication LTD mobile : +39 3291871756 office : +39 011 5694900 www.gedameurope.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
