Re: [asterisk-users] Spandsp rxfax txtax fails no errors
On Mon, Nov 20, 2006 at 02:25:30PM -0800, daveasterisk wrote: -- Executing [EMAIL PROTECTED]:2] RxFAX(SIP/xxx.xxx.xx.xx-0821ec78, /tmp/recievedfax.tif) in new stack saster*CLI and that is where it just sits. no further messages. the target file does not exist and the directory should have acceptable rights. I'm not well-familiar with rxfax, but I recall it has a 'debug' option. Could you try using it? One other sanity check: make sure you're not out of disk space: df /tmp -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Welcome to Join Asterisk MSN Groups!
:), welcome to join MSN groups: [EMAIL PROTECTED], [EMAIL PROTECTED], and [EMAIL PROTECTED] Add to your msn friend, and /help for help! Have a good time here ! ___ --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] RE: Snom 360 Multiple calls on hold help
On Tue, Nov 21, 2006 at 03:55:53PM -0700, Ron McCarthy wrote: Wow! When using Zap channels it doesnt work!!! So for some reasonm, SIP 2 SIP work great, but 2 ZAP channels on hold get bridged I verified on a 360 and a 320. Not good!!! Any help, any clues One clue from the hitchhiker's guide: don't panic. There is certainly no need for the exclamantion marks (!). that im missing? What we're missing is your zapata.conf . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Diva Server, chan_capi and tone detection
Hi all, I have a Diva Server V-BRI-2 card, which support, as written in reference guide: Extended tone processing (human talker detection, generation and detection of country-specific tones) I would like to detect human speech and fax tone with asterisk. I think that the diva card transmit a DTMF code when detecting voice, but chan_capi doesn't receive this DTMF code. I verbose it more, displaying all DTMF received, and only DTMF code CNG is received. Did you know how I can enable this detection (see DivaReportTones in Diva Server SDK) or how can I receive this DTMF in chan_capi ? Thanks Greg ___ --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] Agent Channel SIP transfer
Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer call using SIP phone's transfer feature, he is always in busy status and cannot answer any more incoming call from queue until the transferee hang up the call. -- Regards! Liangliang ___ --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] snom subscriptions issue on WRT
Hi, I've just installed asterisk 1.2.1 on my openwrt distro ( I own a WRT54GL by Linksys ) . No problem by now, but I can see that my 3 snom 320, once they started they send subscriptions to asterisk, and I can see that running: sip show subscriptions But, after one hour about, OR when I do asterisk reload , asterisk losts all th snom subscriptions. Someone can help me please? Thanks ___ --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-cluster with one database
hello all, somebody know how it is possible to have 2 asterisk and one database which is shared for both * ? i saw a problem with hinting that only 1 asterisk where the call ist made knowing that th eline is in use, the other said line is free. Is there a possibilty to let both * see that the line is inuse? regards rene ___ --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] IAX Codecs IAX Realtime
Hi for all! I have two questions about IAX2 connections ... - is it possible to use codecs used by IAX2 depending on SIP codecs? I have some SIP phones and one FAX using IAXMODEM. SIP users use alaw and IAXMODEM use slin. When i'm calling out thru IAX2 connection the best solution is to use slin codec for fax and GSM for SIP (with asterisk transcode), but i don't know how to configure this, because all connections going thru IAX2 use codecs based on IAX2 peer configuration (all calls go thru GSM) - is it possible to set TRUNK=yes if im using IAX2 realtime? I can't find trunk field in iax_buddies table description. thanks for all comments, cheers Marian -- Marian Rychtecky [EMAIL PROTECTED] Tel. +420 724 397 441 ICQ 76582857 ___ --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] Re: Spandsp rxfax txtax fails no errors
On Tue, Nov 21, 2006 at 03:33:57PM -0800, daveasterisk wrote: Is there anyone who can help with this? rxfax and txfax when called in the extensions do nothing and no error are generated that I can find. I asked something similar on the list a while ago, got no answers and took a look at the code myself and learned a little bit. Before I used System(tiff2pdf) to detect errors, which wasn't so elegant, but worked well anyway. The description looks like this: *CLI show application RxFAX -= Info about application 'RxFAX' =- [Synopsis] Receive a FAX to a file [Description] RxFAX(filename[|caller][|debug]): Receives a FAX from the channel into the given filename. If the file exists it will be overwritten. The file should be in TIFF/F format. The caller option makes the application behave as a calling machine, rather than the answering machine. The default behaviour is to behave as an answering machine. Uses LOCALSTATIONID to identify itself to the remote end. LOCALHEADERINFO to generate a header line on each page. Sets REMOTESTATIONID to the sender CSID. FAXPAGES to the number of pages received. FAXBITRATE to the transmition rate. FAXRESOLUTION to the resolution. Returns -1 when the user hangs up. Returns 0 otherwise. If you read the code, a return value of -1 means error and 0 means success, although not clearly stated so in the message above. So far, that is what you would expect, but return values are not testable in * dial plans, as far as I know. I modified app_rxfax.c to set FAXSTATUS to ERROR or SUCCESS and got it working, but then I discovered that the four return variables listed above are set only on success. I think that FAXPAGES would be the best to use for error checking. But still, you will not get a reason for the failure... There is a line in the code: ast_log(LOG_DEBUG, Fax receive not successful - result (%d) %s.\n, result, t30_completion_code_to_str(result)); that shows us that written information on the type of error *is* available. These message are in the spandsp code I suppose. Regards: Håkan pgp99onlNRg5J.pgp Description: PGP signature ___ --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] Is it easy to route SIP/SDP and SIP/RTP through different routes ?
Hello, The setup is : Asterisk --www --- Router1-LAN - SIP Phones | | |--pstn--Router2| (I hope the sketch is understandable. If not, it could be summarized with : SIP hardphones on a given LAN are connected to an Asterisk server through 2 different routers. one is connected to PSTN, the over to the www) For backup/failover puropose, is it easy to setup 2 seperate routers so that RTP media flow goes through router1, and SIP/SDP goes through router2 ? So that registration, signaling and so on benefit from Internet access flat rates and time sensitive Voice packet benefit from PSTN QoS. This way, you don't even have to deal with RTP and NAT coexistence. So ? ___ --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] qualify=yes
hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? I want to view whitch voip-phone is connected but I don't want to DOS my adsl connection ;) Thanks Enrico P. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto ___ --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] qualify=yes
Enrico Pasqualotto wrote: hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? You have to set qualify=second instead of qualify=yes|no. Eheheheh -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto ___ --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] reduce dialtone volume on zap channel.
On Nov 20, 2006, at 2:21 PM, Don Pobanz wrote: Eric ManxPower Wieling No, you cannot change the volume of ONLY the dialtone on a Zap interface. I was afraid of that. The most common problem with the first digit being missed by the telco is that Asterisk is trying to dial too soon after it goes off hook. If it is an analog port then prefixing your number with w or ww will help. My issue is prior to ever sending the digits somewhere else such as the pstn. It is just having asterisk recognize the dtmf when I press a button on the phone. Don Pobanz Dan, If I have followed this thread correctly, your problem is that, when you pick up a local analog phone connected to asterisk through a zap channel, asterisk generates a dialtone, and everything works fine, except that the echo is intolerable. Then, you install an echo canceller, and then asterisk cannot reliably register your DTMF digits when you pick up the phone and dial. In other words, your problem shows up when you install the echo cancellers. Do you have the echo cans installed between your local extension and the zap channel? I assume so, because otherwise they should have no effect on your DTMF. Maybe I'm missing something, but I was always of the impression that echo cancellers are installed between asterisk and the PSTN, not between the local handset and asterisk. That way, the echo canceller is only in the media stream when you place a call out to the PSTN. I assume that you don't have echo problems calling from one local analog extension to another. If you do, however, I would suggest that maybe your problem is bigger than just a DTMF issue. Just a thought, Tom ___ --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] SOLVED: Digium TE405 card and Matra PBX
Hello asterisk-users, I have solved interconnection between Digium TE405 and Matra PBX. I plug this card to another computer and with the same configuration parameters card now works without problem. First server has 2xPIII/1GHz and 256MB of RAM, Adaptec SCSI adapter and SCSI system disk. I don't know now, what chipset it was, but mainboard was from Supermicro, then I pretend Intel chipset. Now I have this card in the 2x Opteron F 2218, 4GB RAM, mainboard Tyan Thunder h2000M, chipset is ServerWorks. FYI. Sincerely Jan Marek -- Ing. Jan Marek | Nez mi poslete prilohu .doc, .xls University of South Bohemia | nebo .ppt, prectete si, prosim, Academic Computer Centre | WWW stranku uvedenou na poslednim Phone: +420-38-9032080 | radku signatury... http://www.gnu.org/philosophy/no-word-attachments.cs.html ___ --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] Zaptel error
hi all iam using ztdummy driver after my call end , when i look at debug mode in cli i get this errors --- (0 headers 0 lines) Nat keepalive --- -- Reloading module 'chan_agent.so' (Agent Proxy Channel) == Parsing '/etc/asterisk/agents.conf': Found -- Reloading module 'chan_local.so' (Local Proxy Channel) -- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring switchtype Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring signalling Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring rxwink -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP)) -- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP)) -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny)) is this error cause any problem or just ignore this any suggestion Ram ___ --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] about voicemail setting
As I know, the voicemail will be sent using localhost smtp. I want to use another smtp server for sending voicemail to the users. Is it possible to set it, where to set it? ___ --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] Request for working config for DISA
Hi Friends, I have configured DISA. But, its not working. When I dial to my zap channel, its asking to enter pin number. After entering PIN number, its giving continuous engage sound and hangup. Can anybody send me correct working configuration for DISA? Looking forward to your response. Thank you. Regards, Chandra. - Sponsored Link Get an Online or Campus degree - Associate's, Bachelor's, or Master's -in less than one year.___ --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] Ast 1.4 and B410p
I have used the b410p card with Asterisk 1.12 quite successfully. I now want to get the card to work with Asterisk 1.4.0beta3. It however can't seem to get chan_misdn compiled. In menuselect, chan_misdn has this: Depends on: isdnnet, misdn, suppserv Can anyone help? -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Can anyone enlighten me as to what this means?
Asterisk is connecting to the outside world. I have 2 PRI lines to the telco, and am using them as primary and secondary clock source. span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,2,0,esf,b8zs bchan=25-47 dchan=48 Here's something else interesting. Any calls placed from within the telcos network completed fine it was only calls placed outside the telco's network. To me this yells telco problem... yet a restart of asterisk fixed the issue... On 11/22/06, Paul Hales [EMAIL PROTECTED] wrote: Are you connecting your Asterisk box to the outside world or a PABX? (I got this sort of error connecting an Asterisk box to a pabx..) PaulH On Tue, 2006-11-21 at 22:28 -0500, Matt wrote: We are doing PRIs into T4XXP cards. When I call out things are fine... however tonight sometimes on inbound calls I'd get: chan_zap.c: Duplicate setup requested on channel 0/1 already in use on span 1 in the full debug log followed by a fast busy signal on the calling parties end. Anyone know what would cause that? ___ --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 ___ --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 ___ --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] about voicemail setting
On Wed, 2006-11-22 at 18:17 +0800, rilawich ango wrote: As I know, the voicemail will be sent using localhost smtp. I want to use another smtp server for sending voicemail to the users. Is it possible to set it, where to set it? ___ it does not use smtp. If it did use smtp it would need to handle errors and queuing in app_voicemail. It pipes it to /usr/sbin/sendmail -t. I use exim (which installs a link /usr/sbin/sendmail) and tell exim to route it to another mailserver via smtp. So you need to configure whatever mta you have installed on your system to route it to the other smtp server. Conrad ___ --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] RE: Snom 360 Multiple calls on hold help
I'm having the same problem, pressing a speed dial/extension when 2 calls are on the phone connect the 2 calls together. Typing the number instead of using speed dial works. With older firmware, 6.2.1 or 6.3, it was working... But then other problem with pickup, deadlocking the phone (or slowing it down). Certainly due to the dp bug (fixed in 6.5.1). Regards, Alban. Le Mardi 21 Novembre 2006 15:37, Usman Tahir a écrit : Just don't put the 2nd call on hold if you want to transfer it to a third one i.e. either through speed dial/extension or through manual input. Regards, Usman. -- Message: 10 Date: Tue, 21 Nov 2006 06:44:29 -0700 From: Ron McCarthy [EMAIL PROTECTED] Subject: Re: [asterisk-users] RE: Snom 360 Multiple calls on hold help To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Perhaps this is a user error then. I get the first call, press hold. Second call comes in, I press hold. Then I scroll thru the calls on hold, hit transfer then hit the speed dial or dial extension. Thats wrong I take it? Can you show me the call process then on how to transfer calls if this wrong?? Im setting up a mock lab here in a few minutes to test some more myself! Thanks! ___ --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 ___ --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] VM mail notification and locale
LC_CTYPE or LC_TIME?
LC_TIME, sorry for the typo.
I added
export LC_TIME=fr_FR
(date; locale) /tmp/log
in /etc/init.d/asterisk and restarted by service asterisk restart.
The content of /tmp/log :
linux2:/tmp # cat log
Wed Nov 22 12:43:09 CET 2006
LANG=POSIX
LC_CTYPE=POSIX
LC_NUMERIC=POSIX
LC_TIME=POSIX
LC_COLLATE=POSIX
LC_MONETARY=POSIX
LC_MESSAGES=POSIX
LC_PAPER=POSIX
LC_NAME=POSIX
LC_ADDRESS=POSIX
LC_TELEPHONE=POSIX
LC_MEASUREMENT=POSIX
LC_IDENTIFICATION=POSIX
LC_ALL=POSIX
Obviously the assigment has no effect but a NoOp(LC_TIME: ${ENV(LC_TIME)})
in extension.ael2 displays in the CLI :
-- Executing NoOp(SIP/103-0819f470, LC_TIME: fr_FR) in new stack
!
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Tzafrir Cohen
Envoyé : mardi 21 novembre 2006 23:32
À : asterisk-users@lists.digium.com
Objet : Re: [asterisk-users] VM mail notification and locale
On Tue, Nov 21, 2006 at 10:54:31PM +0100, Dominique Dartois wrote:
I set the ENV variable LC_TYPE=fr_FR
LC_CTYPE or LC_TIME?
in the starting shell of Asterisk.
A NoOp(${ENV(LC_TIME)}) in extension.ael2 shows the right value.
In voicemail.conf, emaildateformat=%A prints the day of week in the
mail body but in English, not in French.
Could you add to that script, just before running asterisk,
(date; locale) /tmp/log
and provide here the generated /tmp/log from an asterisk startup so we can
see that the assignment had the right effect?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Re: Call limits and VoIP providers
I`m impressed. Thanks for the reply, I'll try that! Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen Sent: November 21, 2006 4:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Call limits and VoIP providers M == Mike [EMAIL PROTECTED] writes: M That sounds like the most practical solutionExcept its not clear M to me when the number of group unit is actually decreased. M Does it work automagically when the call is hung up, or is there a M command to decrease it? It's completely automagic. It also knows not to double-count the same call if you put a call in the same group twice. It really is rather clever. /Benny ___ --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 ___ --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] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V
Also the 48v and higher systems can transmit the lower current further than a low voltage with a higher current. On 11/21/06, Julien Goodwin [EMAIL PROTECTED] wrote: On Tue, Nov 21, 2006 at 08:57:44PM -0500, Zeeshan Zakaria arranged a set of bits into the following: Why Aastra phones use more electricity, i.e. 48VDC whereas other phones use much less, e.g. Grandstream and Linksys both use only 5VDC. I first thought it was because of PoE, but the ones with 5VDC also run fine on PoE. What is the difference in power consumption then? The difference due to the different voltages would be 1w. Many of the commercial phones (Aastra, Polycom, Cisco) use 48 volt power supplies as it lets them have a single power circuit for wall-warts and PoE (Standard PoE is 48 volts). Basic electrical theory (for DC) is that power == Watts, and Watts = Volts * Amps, so the only real difference between a 5v input and a 48v is that the 48v will use less current (although it might go through more DC-DC convertors those are highly efficient these days) Thanks, Julien -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFFY7PXBN1Ia7JOLPcRAu01AJ0UPc5dHFj/3gavruQPwD+oOXd+mgCgn/70 5w5Mrgn6JJcjHdMKGW1+ihA= =m+2o -END PGP SIGNATURE- ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] snom subscriptions issue on WRT
Likely a SNOM setting, look in the wiki for info on configuring the phones for asterisk as the are several settings that allow them to work. On 11/22/06, tommaso.carrara [EMAIL PROTECTED] wrote: Hi, I've just installed asterisk 1.2.1 on my openwrt distro ( I own a WRT54GL by Linksys ) . No problem by now, but I can see that my 3 snom 320, once they started they send subscriptions to asterisk, and I can see that running: sip show subscriptions But, after one hour about, OR when I do asterisk reload , asterisk losts all th snom subscriptions. Someone can help me please? Thanks ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] RE: Snom 360 Multiple calls on hold help
On 11/22/06, Alban [EMAIL PROTECTED] wrote: I'm having the same problem, pressing a speed dial/extension when 2 calls are on the phone connect the 2 calls together. Typing the number instead of using speed dial works. With older firmware, 6.2.1 or 6.3, it was working... But then other problem with pickup, deadlocking the phone (or slowing it down). Certainly due to the dp bug (fixed in 6.5.1). Regards, Alban. Has this been reported to snom by anyone? They are generally pretty good at fixing this type of issue and providing beta firmware. Regards, Steve ___ --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] spc.exe
You didn't get the memo? I will have alice send you over a copy Seriously, yes there are laws against what he did but it would be in the best interest of their sales department to send the software with every phone. On 11/21/06, Brian Capouch [EMAIL PROTECTED] wrote: Matt wrote: Good job you've just violated Linksys/Cisco IP laws Good grief; I didn't realize they now have their own lawmaking powers!! B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] Hints no longer working in 1.4beta3 with Polycoms
Mark was working on this, I think it was called sla and it called something line apperance On 11/21/06, John Lange [EMAIL PROTECTED] wrote: Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --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 ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] help in Call parking......
Hello Users I'm Doing working on Both OpenSER and Asterisk ... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=115,1,Park() exten =115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says U parked on 701 extension After When 9001 dial 701 . it Say 483 too many parameters ... in X-lite , Actual it has to ring 9003, -- Executing Park(SIP/9003-085d9e10, ) in new stack == Parked SIP/9003-085d9e10 on 701. Will timeout back to extension [from-sip] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/9003-085d9e10' == Spawn extension (from-sip, s, 1) exited KEEPALIVE on 'SIP/9003-085d9e10' Nov 22 18:18:45 NOTICE[3289]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.2.5 -- Stopped music on hold on SIP/9003-085d9e10 -- Registered extension context 'park-dial' -- Added extension 'SIP/9003' priority 1 to park-dial == Timeout for SIP/9003-085d9e10 parked on 701. Returning to park-dial,SIP/9003,1 -- Executing Dial(SIP/9003-085d9e10, SIP/9003||t) in new stack -- Called 9003 -- Got SIP response 482 Loop Detected back from 192.168.2.76 -- Now forwarding SIP/9003-085d9e10 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/9003-085df878) Nov 22 18:19:20 NOTICE[3949]: chan_local.c:498 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Nov 22 18:19:20 NOTICE[3949]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 0) == Everyone is busy/congested at this time (1:0/0/1) Nov 22 18:19:30 WARNING[3949]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'park-dial' -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.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] Rewriting caller ID from database?
Hi Most of our customers have generic names like Hospital, so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as Reading Hospital so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you. ___ --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] RE: Snom 360 Multiple calls on hold help
Yes, already. Waiting now for a new firmware... Regards, Alban Le Mercredi 22 Novembre 2006 13:50, Steve Davies a écrit : On 11/22/06, Alban [EMAIL PROTECTED] wrote: I'm having the same problem, pressing a speed dial/extension when 2 calls are on the phone connect the 2 calls together. Typing the number instead of using speed dial works. With older firmware, 6.2.1 or 6.3, it was working... But then other problem with pickup, deadlocking the phone (or slowing it down). Certainly due to the dp bug (fixed in 6.5.1). Regards, Alban. Has this been reported to snom by anyone? They are generally pretty good at fixing this type of issue and providing beta firmware. Regards, Steve ___ --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 ___ --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] Recordings.
Hi, We want to build an Asterisk system that needs to be able to record, when in a peak situation, a maximum of twenty calls simultaneously. I could not find any reference to performance and recording. I need to order a new server but need to know the specs I need. Does anyone have experience with recording multiple calls simultaneously on a single system with or without performance trouble? What kind of system do I need? John Vermeeren ___ --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] Welcome to Join Asterisk MSN Groups!
Why would I want to join MSN groups then MS can't get an OS right! Now MS whats to do get into VOIP that will be a total messup. The thing is when MS will try to say that they asterisk. MS has no place anywhere around Asterisk. You will see what I mean just look at the bottom of MY website. I just wanted to put my .02 in about MS and VOIP Servers.. I know some will agree with me some will not. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email@ (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Mayson.Wang wrote: :), welcome to join MSN groups: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED], [EMAIL PROTECTED] mailto:[EMAIL PROTECTED], and [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]! Add to your msn friend, and /help for help! Have a good time here ! ___ --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 Inbound (clean). Database: 0650-0, 11/22/2006 - 11/22/2006 7:52:22 AM ___ --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] qualify=yes
Enrico Pasqualotto wrote: Enrico Pasqualotto wrote: hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? You have to set qualify=second instead of qualify=yes|no. This is WRONG. qualify=500 means consider this device lagged if responses take longer than 500ms I don't know if you can set the frequency of qualify packets. If you can, I assume the option would be listed in sip.conf.sample. ___ --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] reduce dialtone volume on zap channel.
Tom Rymes wrote: Dan, If I have followed this thread correctly, your problem is that, when you pick up a local analog phone connected to asterisk through a zap channel, asterisk generates a dialtone, and everything works fine, except that the echo is intolerable. Then, you install an echo canceller, and then asterisk cannot reliably register your DTMF digits when you pick up the phone and dial. In other words, your problem shows up when you install the echo cancellers. Yes you understand correctly. Do you have the echo cans installed between your local extension and the zap channel? I assume so, because otherwise they should have no effect on your DTMF. Yes analog phone (Aastra PT390) | channel bank (Adtran TA750) | T1 echo can (Orion Telecom) | Asterisk 1.2.13 (Dell poweredge 1750) Maybe I'm missing something, but I was always of the impression that echo cancellers are installed between asterisk and the PSTN, not between the local handset and asterisk. I also have a T1 echo canceller between asterisk and the pstn. That will help with the echo for SIP phones. That way, the echo canceller is only in the media stream when you place a call out to the PSTN. I assume that you don't have echo problems calling from one local analog extension to another. I do have echo extension to extension! That is the issue I am trying to eliminate. If you do, however, I would suggest that maybe your problem is bigger than just a DTMF issue. This may be bigger than DTMF. However, since this is in service, I need to keep it running. I have worked with Digium and with Orion to try and resolve this. Digium has ssh into the system on 4 different occasions. Orion assures me that they have thousands of these same echo cancellers in service but have never seen the issues I am having. Some of the things I have done are. 1 - moved TE410P cards to their own interrupts 2 - modified levels in Asterisk, echo cancellers and channel bank. (I have a few pages of test results.) I have set levels to Digium recommended levels and to Orion recommended levels. I have also tried a whole bunch of other levels. By changing levels, I can get better results (8% errors instead of 20 or 25%). However, an error rate above 2% is unacceptable. 3 - turned off vpm support in TE412P card 4 - turned off frame buffer 5 - moved to another server (Dell 1650 poweredge) with different TE410P cards. 6 - loaded different versions of zaptel and of asterisk (all in the 1.2.x version) 7 - swapped Digium cards with a spare Once asterisk recognizes a pressed digit, all of the rest of the digits are recognized. The only different for the first digit is the presence of dialtone. That was why I was trying to reduce the dialtone volume. Maybe this is not the correct solution. But hey, I need to try something! The TE412P card with echo cancellers enabled did not offer satisfactory results. (too much echo). Don Pobanz winmail.dat___ --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] DTMF detection during Call
Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian ___ --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] Zaptel error
On Wed, 2006-11-22 at 15:45 +0530, ram wrote: [snip] Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring switchtype Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring signalling Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring rxwink [snip] is this error cause any problem or just ignore this ^ Error? Where does it say error? Read the messages carefully and you will see that it says.. WARNING. If it was an error it would have said ERROR. But it didn't. Phew. Just a harmless warning. And to figure out what the warnings mean, I suggest you buy/get the Asterisk book. It's very helpful to learn about these basic things. Regards, Patrick ___ --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] IP601 Expansion Module HELP!!!
Ron McCarthy wrote: Hey list, Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion modules to work. I need the XML config files I guess. Does anyone have these I can have? Im trying to get this phone up and running, and haveing MUCHO problems, can someone help me out!! Im sure if I see the configs I can see how it works, just need those XML files!! The ones from the 501 that I have dont seem to work. Or do you have any help/configs on this? Any help would be GREAT on this, im in a huge crunch Thanks in advance!!! If you are running the same firmware on the 501 and the 601 the config files are the same. If you are looking for the firmware and bare bones configs you have three options: -Ask the vendor you purchased the phones from for it. -Download the 1.6.x series from polycom themselves. -Go here: http://www.freedomphones.net/polycom/files/ As for the expansion modules, to my knowledge there isn't anything special that needs to be done to configure them. Simply plugging them in to the phone activates them and gives you the extra set of line keys/speed dials. If you are still having problems, it might be helpful if you described the specific task you are trying to accomplish and what it is not working. -Dave ___ --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] Recordings.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: Does anyone have experience with recording multiple calls simultaneously on a single system with or without performance trouble? What kind of system do I need? Well, isnt this just a simple calculation? Do a record of one of your lines for about a minute. Look at the size of the created file and divide the kb by 60 and multiply by 20 and you have an first overview about how much data will get written down to harddisk per second. But I think you should be fairly well if you use state-of-the-art server disks. They should be fast enough for this. Marcus -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZFjMqwWWw48OFWoRAvhtAKC58l2WXpK/RmzWB2FtRDbHFxsJWQCgp3OI o9zojHnurfaMtAOjLytHFUs= =upqP -END PGP SIGNATURE- ___ --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 incoming call behaviour
I am using asterisk to receive call from a DID provider . In configured everything in freepbx properly and its working . I forwarded incoming calls from did to a certain extension . Now i tried calling from another sip provider to this box , when i call from other provider to my DID number then call reaches asterisk and is sent to configured extension .. however if the extension hangs up without picking then also i am being billed at sip provider ( outgoing one ) . In simple words when people call me then they ( other people ) are billed even if configured extension isnt picked up and hangs the phone. Normally when you call a person and they hang up then you arent charged . Is this asterisk behaviour or is it freepbx dialplan the culprit here ? ___ --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] Hairping calls and Originating CLI
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: 21 November 2006 19:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hairping calls and Originating CLI On 21 Nov 2006, at 10:08, Adrian Marsh wrote: Hi, I'm trying to track down what happened to some calls to a mobile today between 9:00 and 9:15 (I've modified the log to mask IPs/Passwds/Phone #, etc. 127.111.200.* is our PSTN provider - Gradwell, the extension is configured to ring 3 SIP connections, then divert to a mobile). Below is our IAX log from our Asterisk box. The log has raised 2 questions. heres the call-flow as I understand it: First question: 1) A call comes in from 127.111.200.135, we try contacting various SIP clients, and timeout 2) After the timeout we then place an outbound call to a mobile, via 127.111.200.135, but the call is accepted by 127.111.201.75 (is this a cluster?) 3) The mobile is answered, and both call-legs are marked as ready to transfer.. At this point: Who hairpins the call? i.e., is the call handed back to Gradwell ? The next message in the log is Releasing the two calls, rather than any message about joining them, and then we see hangup messages. Second question: In the outbound leg (our A*k - mobile), we set the CallerID number to the public PSTN of the local extension here.. Is there any reason why we can't set this to be the originating CLI from the first leg (the incoming call) ? Example scenario: Bob calls Tracy, Bobs CLI gives his originating number.. The A*k box makes an outbound call to Tracys mobile. Theres little point in putting the second-leg originating CLI as Tracy's office number (herself calling herself). Instead, we need to put Bobs CLI as the originating digits, and the join the two legs of the call. I'm going to post this out to the A*k maillist, and see what comes back, but I thought I'd get your view.. ubiphone*CLI -- Accepting AUTHENTICATED call from 127.111.200.135: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw), priority = mine ubiphone*CLI -- Executing GotoIf(IAX2/127.111.200.135:4569-4, 0?20) in new stack -- Executing Dial(IAX2/127.111.200.135:4569-4, SIP/204SIP/ 404IAX2/20004:[EMAIL PROTECTED]/20004|15|r) in new stack -- Called 204 Nov 21 09:07:14 NOTICE[1333]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) ubiphone*CLI -- Called 20004:[EMAIL PROTECTED]/20004 ubiphone*CLI -- SIP/204-088ac6b0 is ringing ubiphone*CLI -- Call accepted by 194.192.14.200 (format alaw) -- Format for call is alaw ubiphone*CLI -- IAX2/194.192.14.200:4569-6 is ringing ubiphone*CLI -- Nobody picked up in 15000 ms -- Hungup 'IAX2/194.192.14.200:4569-6' -- Executing Macro(IAX2/127.111.200.135:4569-4, call-mobile| 2004|x25) in new stack -- Executing Set(IAX2/127.111.200.135:4569-4, CALLERID (number)=404) in new stack -- Executing Dial(IAX2/127.111.200.135:4569-4, IAX2/ iaxout:[EMAIL PROTECTED]/x25|30|r) in new stack ubiphone*CLI -- Called iaxout:[EMAIL PROTECTED]/x25 ubiphone*CLI -- Call accepted by 127.111.201.75 (format ulaw) -- Format for call is ulaw ubiphone*CLI Nov 21 09:07:32 ERROR[13825]: chan_sip.c:10990 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.1.2, but there is no hint for that extension ubiphone*CLI -- IAX2/127.111.201.75:4569-3 is ringing ubiphone*CLI -- IAX2/127.111.201.75:4569-3 is making progress passing it to IAX2/127.111.200.135:4569-4 ubiphone*CLI -- IAX2/127.111.201.75:4569-3 answered IAX2/127.111.200.135:4569-4 -- Attempting native bridge of IAX2/127.111.200.135:4569-4 and IAX2/127.111.201.75:4569-3 ubiphone*CLI -- Channel 'IAX2/127.111.200.135:4569-4' ready to transfer ubiphone*CLI -- Channel 'IAX2/127.111.201.75:4569-3' ready to transfer -- Releasing IAX2/127.111.201.75:4569-3 and IAX2/127.111.200.135:4569-4 ubiphone*CLI -- Hungup 'IAX2/127.111.201.75:4569-3' == Spawn extension (macro-call-mobile, s, 2) exited non-zero on 'IAX2/127.111.200.135:4569-4' in macro 'call-mobile' == Spawn extension (macro-call-mobile, s, 2) exited non-zero on 'IAX2/127.111.200.135:4569-4' -- Hungup 'IAX2/127.111.200.135:4569-4' To answer your questions: At this point: Who hairpins the call? i.e., is the call handed back to Gradwell ? Your asterisk box hairpins it. Unless you have notransfer=yes in iax.conf asterisk will always _try_ and transfer a call where both endpoints are IAX so that it
Re: [asterisk-users] qualify=yes
I doubt that . I think qualify=500 means asterisk checks every 500 ms if the other extension is available or not . Because when qualify=( value in ms ) is set and you do a sip show peers in console asterisk whos how much latency is there between extension and asterisk . If i set qualify = no then it shows UNKNOWN . If i set qualify=10 then it doesnt mean asterisk shows extension lagged if latency is less than 10 ms ... It just checks every 10 ms for extension . I am not very sure though :) On 22/11/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Enrico Pasqualotto wrote: Enrico Pasqualotto wrote: hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? You have to set qualify=second instead of qualify=yes|no. This is WRONG. qualify=500 means consider this device lagged if responses take longer than 500ms I don't know if you can set the frequency of qualify packets. If you can, I assume the option would be listed in sip.conf.sample. ___ --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 ___ --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] Recordings.
Has anyone tried recording to a ramdisk? To an NFS mount? Was there a benefit? [EMAIL PROTECTED] wrote: Hi, We want to build an Asterisk system that needs to be able to record, when in a peak situation, a maximum of twenty calls simultaneously. I could not find any reference to performance and recording. I need to order a new server but need to know the specs I need. Does anyone have experience with recording multiple calls simultaneously on a single system with or without performance trouble? What kind of system do I need? John Vermeeren ___ --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 ___ --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] Recordings.
Yeh even a simple UDMA 5 enabled hard drive can handle 30 calls recording easily . Sata hard drives are even better . On 22/11/06, Marcus Franke [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: Does anyone have experience with recording multiple calls simultaneously on a single system with or without performance trouble? What kind of system do I need? Well, isnt this just a simple calculation? Do a record of one of your lines for about a minute. Look at the size of the created file and divide the kb by 60 and multiply by 20 and you have an first overview about how much data will get written down to harddisk per second. But I think you should be fairly well if you use state-of-the-art server disks. They should be fast enough for this. Marcus -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZFjMqwWWw48OFWoRAvhtAKC58l2WXpK/RmzWB2FtRDbHFxsJWQCgp3OI o9zojHnurfaMtAOjLytHFUs= =upqP -END PGP SIGNATURE- ___ --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 ___ --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] iax2 - wildiax phone myself puzzled
I know in advance maybe I'm overlooking something stupid, however I'm really lost and cannot find the solution... Situation: - asterisk-1.2.13 on a linux box with no iptables active. - one iax2 peer defined - one wildiax phone running on my laptop the soft phone is configured to connect register on asterisk, however, I cannot get it working. What am I missing? Please help!! I'm going crazy... Here's my config: iax.conf: [general] bindport=4569 bindaddr=dynamic nochecksums=yes delayreject=yes language=it bandwidth=high disallow=all allow=alaw jitterbuffer=no forcejitterbuffer=no minregexpire=300 maxregexpire=600 tos=lowdelay autokill=yes [support] type=friend auth=plaintext,md5 username=support secret=support host=dynamic qualify=no context=incoming-iax extensions.conf (excerpt): ... [incoming-iax] exten = 1,1,Playback(congratulations-its-working) exten = 1,2,Hangup() ... The wildiax client is properly configured (ip addr of asterisk, username=support, password=support, register=yes) Here's what I get with iax2 debug + console debug active: output of iax2 show peers: Name/UsernameHost Mask Port Status support/support (Unspecified) (D) 255.255.255.255 0 Unmonitored 1 iax2 peers [0 online, 0 offline, 1 unmonitored] log output: 2006-11-22 15:14:06 VERBOSE[6863] logger.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ 2006-11-22 15:14:06 VERBOSE[6863] logger.c:Timestamp: 3ms SCall: 20971 DCall: 0 [10.0.10.160:4569] 2006-11-22 15:14:06 VERBOSE[6863] logger.c:USERNAME: support 2006-11-22 15:14:06 VERBOSE[6863] logger.c:REFRESH : 30 2006-11-22 15:14:06 VERBOSE[6863] logger.c: 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: New max nontrunk callno is 4 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Creating new call structure 3 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Received packet 0, (6, 13) 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: IAX subclass 13 received 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: For call=3, set last=3 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Sending 3 on 3/20971 to 10.0.10.160:4569 2006-11-22 15:14:06 VERBOSE[6863] logger.c: Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK 2006-11-22 15:14:06 VERBOSE[6863] logger.c:Timestamp: 3ms SCall: 3 DCall: 20971 [10.0.10.160:4569] 2006-11-22 15:14:06 DEBUG[6863] acl.c: # Testing 10.0.10.160 with 0.0.0.0 2006-11-22 15:14:06 DEBUG[6855] chan_iax2.c: Checking device state for device support 2006-11-22 15:14:06 DEBUG[6855] chan_iax2.c: iax2_devicestate: Found peer. What's device state of support? addr=0, defaddr=0 maxms=0, lastms=0 2006-11-22 15:14:06 DEBUG[6855] devicestate.c: Changing state for IAX2/support - state 5 (Unavailable) 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Sending 3 on 3/20971 to 10.0.10.160:4569 2006-11-22 15:14:06 VERBOSE[6863] logger.c: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH 2006-11-22 15:14:06 VERBOSE[6863] logger.c:Timestamp: 3ms SCall: 3 DCall: 20971 [10.0.10.160:4569] 2006-11-22 15:14:06 VERBOSE[6863] logger.c:AUTHMETHODS : 1 2006-11-22 15:14:06 VERBOSE[6863] logger.c:USERNAME: support 2006-11-22 15:14:06 VERBOSE[6863] logger.c: 2006-11-22 15:14:06 DEBUG[733] app_queue.c: Device 'IAX2/support' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. 2006-11-22 15:14:06 VERBOSE[6863] logger.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ 2006-11-22 15:14:06 VERBOSE[6863] logger.c:Timestamp: 3ms SCall: 20965 DCall: 0 [10.0.10.160:4569] 2006-11-22 15:14:06 VERBOSE[6863] logger.c:USERNAME: support 2006-11-22 15:14:06 VERBOSE[6863] logger.c:REFRESH : 30 2006-11-22 15:14:06 VERBOSE[6863] logger.c: 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: New max nontrunk callno is 5 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Creating new call structure 4 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Received packet 0, (6, 13) 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: IAX subclass 13 received 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: For call=4, set last=3 2006-11-22 15:14:06 DEBUG[6863] chan_iax2.c: Sending 3 on 4/20965 to 10.0.10.160:4569 2006-11-22 15:14:06 VERBOSE[6863] logger.c: Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK 2006-11-22 15:14:06 VERBOSE[6863] logger.c:Timestamp: 3ms SCall: 4 DCall: 20965 [10.0.10.160:4569] 2006-11-22 15:14:06 DEBUG[6863] acl.c: # Testing 10.0.10.160 with 0.0.0.0 2006-11-22 15:14:06 DEBUG[6855] chan_iax2.c: Checking device state for device support 2006-11-22 15:14:06 DEBUG[6855] chan_iax2.c: iax2_devicestate: Found peer. What's device state of support? addr=0, defaddr=0 maxms=0, lastms=0 2006-11-22 15:14:06 DEBUG[6855] devicestate.c: Changing state for
[asterisk-users] Send event from dialplan
Hi all, Another question for today, hope an answer for this one. I have a program talking with asterisk via the AMI. I receive events, and I would like to insert some events in the dialplan, which could be catch by my program. Any idea how to do this ? Greg ___ --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] Answer Machine Detection
I had a hell of a time getting AMD to work correctly on 1.4. If I didn't
compile asterisk correctly (I'm not sure how I was doing it incorrectly), it
wouldn't work (and would just stop the dialplan execution). Try recompiling
everything (make clean make install) and see if that helps. (It's working
fine for me now...)
FYI, AMD will not Wait() for the answering machine to finish talking. It
will only set some status variables, with which you should use
WaitForSilence() afterwards to wait for everything to get 'quiet'
AMDSTATUS - This is the status of the answering machine detection.
Possible values are:
MACHINE | HUMAN | NOTSURE | HANGUP
AMDCAUSE - Indicates the cause that led to the conclusion.
Possible values are:
TOOLONG-%d total_time
INITIALSILENCE-%d silenceDuration-%d initialSilence
HUMAN-%d silenceDuration-%d afterGreetingSilence
MAXWORDS-%d wordsCount-%d maximumNumberOfWords
LONGGREETING-%d voiceDuration-%d greeting
--
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matheus Rossato
Sent: Tuesday, November 21, 2006 3:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Answer Machine Detection
Hi all,
i'm trying to make AMD, Answer Machine Detection, to work on my
outbound context but i can't get it to work, just on inbound context
like whe i use the application Answer before AMD, but i need to make AMD
to do the detection on an outbound predictive dialer integration. Follow
are the inbound and outbound examples. My current environment is
Asterisk 1.4beta3 and a Digum TE105P with ISDN E1. Have any one managed
to do answer machine detection already?
[outbound]
exten = _x.,1,AMD
exten = _x.,2,Dial(SIP/[EMAIL PROTECTED],,tT)
exten = _x.,3,Wait(2)
exten = _x.,4,Set(RECORDEDFILE=${CALLERID(num)}.wav)
exten = _x.,5,Record(${RECORDEDFILE},,,skip)
exten = _x.,6,Hangup
[inbound]
exten = _x.,1,Answer
exten = _x.,2,AMD
exten = _x.,3,Wait(2)
exten = _x.,4,Set(RECORDEDFILE=${CALLERID(num)}.wav)
exten = _x.,5,Record(${RECORDEDFILE},,,skip)
exten = _x.,6,Hangup
My AMD conf
;
; Answering Machine Detection Configuration
;
[general]
initial_silence = 2500 ; Maximum silence duration before
the greeting.
; If exceeded then MACHINE.
greeting = 1500 ; Maximum length of a greeting. If
exceeded then MACHINE.
after_greeting_silence = 300 ; Silence after detecting a greeting.
; If exceeded then HUMAN
total_analysis_time = 5000 ; Maximum time allowed for the algorithm
to decide
; on a HUMAN or MACHINE
min_word_length = 120 ; Minimum duration of Voice to considered
as a word
between_words_silence = 50 ; Minimum duration of silence after a word
to consider
; the audio what follows
as a new word
maximum_number_of_words = 3 ; Maximum number of words in the greeting.
; If exceeded
then MACHINE
silence_threshold = 256
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[asterisk-users] Recordings for VR analysis
Is there a programmatic to to trim the silence from the beginning and end of a recording? From a .wav file? From a .ulaw file? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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] Recordings.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Vicky wrote: Yeh even a simple UDMA 5 enabled hard drive can handle 30 calls recording easily . Sata hard drives are even better . Hehe, UDMA sounds like EIDE drives.. nice to see they are fast enough, but I do not recommend those as server hardware. ;-) But, if John is going to buy a extra new server, he could use two drives in a mirror setup extra for recordings of these files. As it is not only the frequency of reading/writing these files but other accesses of the media like starting programs or reading/writing of logfiles that slowes down the access to the recorded audio files. Marcus -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZGUUqwWWw48OFWoRAvidAJwPSpTSuY6nwxKTDKI8fZDmshmbUgCgtWAp 27akzsEDv03q5CmlGMObo50= =2jAI -END PGP SIGNATURE- ___ --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] Send event from dialplan
Sorry, asking too quickly, thats what im looking for : http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Eve nts Greg _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gregory Duchatelet Envoyé : mercredi 22 novembre 2006 15:33 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Send event from dialplan Hi all, Another question for today, hope an answer for this one I have a program talking with asterisk via the AMI. I receive events, and I would like to insert some events in the dialplan, which could be catch by my program. Any idea how to do this ? Greg ___ --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] Is this possible?
Dont Use Call Progress Instead Use The M Option In App Dial That Asks The User To Press A Button To Accept The Call On 11/21/06, shadowym [EMAIL PROTECTED] wrote: Anyone tried this, I put in an Asterisk/FreePBX phone system to replace one of those el cheapo Bizfon analog key systems. The Bizfon was able to do follow me to analog lines but I do not believe the capability is in Asterisk. Of course now it becomes the one feature they cannot do without. Somehow the Bizfon is able to detect if an analog line is answered (without answer supervision). If not it goes to the Bizfon voicemail. I have not figured out a way to do this in Asterisk/FreePBX. Anyone know how Bizfon does it? Is there a way to do it in Asterisk? The analog lines are standard loop start lines in a hunt group provided by Bell Canada. They do not have any type of answer supervision such as polarity reversal. I think maybe the Bizfon asks for button press confirmation on the other end and if not it times out. That's just a theory though. With Asterisk, as soon as an analog line goes off hook and starts ringing Asterisk set's that line to answer state unless it has polarity reversal (answer supervision for loop start lines). Once Asterisk thinks the Analog line is answered you cannot do anything further. callprogress=yes provides the functionality I believe but this is a production environment and that feature does not work reliably enough. Is this how bizfon does it maybe? Maybe they can detect the ringing indicating the call was not answered? ___ --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 ___ --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] Recordings.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael Welter wrote: Has anyone tried recording to a ramdisk? To an NFS mount? Was there a benefit? RAM disk? Interesting idea, but what to do in case of a server crash loosing these recorded files? You will get very angry customers if you have to explain them, that your server, where you did record their complaints, crashed and lost their problems :) Id recommend this as a cache drive where you would move the files away from, when the call is finished. But thats extra cpu cycles and it would be kind of an effort to trigger the move the files after call is finished.. Marcus -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZGnpqwWWw48OFWoRAoxqAJ0WIynm41jcxh3WT2GM1C/8bMj1KACg8vZJ gkKQtIlykyJO6ZQwmvggUm4= =JC+B -END PGP SIGNATURE- ___ --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] How ecord all calls?
Hi All!! Prompt how to record all calls passing through certain span? --- Thanks... ___ --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] qualify=yes
Vicky wrote: I doubt that . I think qualify=500 means asterisk checks every 500 ms if the other extension is available or not . Because when qualify=( value in ms ) is set and you do a sip show peers in console asterisk whos how much latency is there between extension and asterisk . If i set qualify = no then it shows UNKNOWN . If i set qualify=10 then it doesnt mean asterisk shows extension lagged if latency is less than 10 ms ... It just checks every 10 ms for extension . I am not very sure though :) Try it. Set qualify=1 in sip.conf. ___ --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] How ecord all calls?
Eugeniy Khvastunov wrote: Hi All!! Prompt how to record all calls passing through certain span? pbx-1*CLI show application record pbx-1*CLI -= Info about application 'Record' =- [Synopsis] Record to a file [Description] Record(filename.format|silence[|maxduration][|options]) Records from the channel into a given filename. If the file exists it will be overwritten. - 'format' is the format of the file type to be recorded (wav, gsm, etc). - 'silence' is the number of seconds of silence to allow before returning. - 'maxduration' is the maximum recording duration in seconds. If missing or 0 there is no maximum. - 'options' may contain any of the following letters: 'a' : append to existing recording rather than replacing 'n' : do not answer, but record anyway if line not yet answered 'q' : quiet (do not play a beep tone) 's' : skip recording if the line is not yet answered 't' : use alternate '*' terminator key instead of default '#' If filename contains '%d', these characters will be replaced with a number incremented by one each time the file is recorded. Use 'show file formats' to see the available formats on your system User can press '#' to terminate the recording and continue to the next priority. If the user should hangup during a recording, all data will be lost and the application will teminate. pbx-1*CLI ___ --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] DTMF detection during Call
[EMAIL PROTECTED] wrote: Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. pbx-1*CLI show application dial pbx-1*CLI -= Info about application 'Dial' =- [Synopsis] Place a call and connect to the current channel [Description] Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL]): This applicaiton will place calls to one or more specified channels. As soon as one of the requested channels answers, the originating channel will be answered, if it has not already been answered. These two channels will then be active in a bridged call. All other channels that were requested will then be hung up. Unless there is a timeout specified, the Dial application will wait indefinitely until one of the called channels answers, the user hangs up, or if all of the called channels are busy or unavailable. Dialplan executing will continue if no requested channels can be called, or if the timeout expires. This application sets the following channel variables upon completion: DIALEDTIME - This is the time from dialing a channel until when it is disconnected. ANSWEREDTIME - This is the amount of time for actual call. DIALSTATUS - This is the status of the call: CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL DONTCALL | TORTURE For the Privacy and Screening Modes, the DIALSTATUS variable will be set to DONTCALL if the called party chooses to send the calling party to the 'Go Away' script. The DIALSTATUS variable will be set to TORTURE if the called party wants to send the caller to the 'torture' script. This application will report normal termination if the originating channel hangs up, or if the call is bridged and either of the parties in the bridge ends the call. The optional URL will be sent to the called party if the channel supports it. If the OUTBOUND_GROUP variable is set, all peer channels created by this application will be put into that group (as in Set(GROUP()=...). Options: A(x) - Play an announcement to the called party, using 'x' as the file. C- Reset the CDR for this call. d- Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current context, or the context defined in the EXITCONTEXT variable, if it exists. D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. The 'called' DTMF string is sent to the called party, and the 'calling' DTMF string is sent to the calling party. Both parameters can be used alone. f- Force the callerid of the *calling* channel to be set as the extension associated with the channel using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the number assigned to the caller. g- Proceed with dialplan execution at the current extension if the destination channel hangs up. G(context^exten^pri) - If the call is answered, transfer both parties to the specified priority. Optionally, an extension, or extension and context may be specified. Otherwise, the current extension is used. h- Allow the called party to hang up by sending the '*' DTMF digit. H- Allow the calling party to hang up by hitting the '*' DTMF digit. j- Jump to priority n+101 if all of the requested channels were busy. L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are left. Repeat the warning every 'z' ms. The following special variables can be used with this option: * LIMIT_PLAYAUDIO_CALLER yes|no (default yes) Play sounds to the caller. * LIMIT_PLAYAUDIO_CALLEE yes|no Play sounds to the callee. * LIMIT_TIMEOUT_FILE File to play when time is up. * LIMIT_CONNECT_FILE File to play when call begins. * LIMIT_WARNING_FILE File to play as warning if 'y' is defined. The default is to say the time remaining. m([class]) - Provide hold music to the calling party until a requested channel answers. A specific MusicOnHold class can be specified. M(x[^arg]) - Execute the Macro for the *called* channel before connecting to the calling channel. Arguments can be specified to the Macro using '^' as a delimeter.
Re: [asterisk-users] Recordings for VR analysis
On Wednesday 22 November 2006 08:43 am, Michael Welter wrote: Is there a programmatic to to trim the silence from the beginning and end of a recording? From a .wav file? From a .ulaw file? Thanks, try man sox - look for 'silence' Brett ___ --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] qualify=yes
qualify=xxx in sip means, consider peer as OK if delay reply is bellow xxx (ms) qualify checks (POKE) is every 60s (and is not configurable in sip.conf) qualify setting in iax.conf is working differently, this is how frequently to check peer (and is not possible to set some POKE delay threshlold as working qualify in sip) this is quite misleading and inconsistent and should be improved ;-) PJ Vicky wrote: I doubt that . I think qualify=500 means asterisk checks every 500 ms if the other extension is available or not . Because when qualify=( value in ms ) is set and you do a sip show peers in console asterisk whos how much latency is there between extension and asterisk . If i set qualify = no then it shows UNKNOWN . If i set qualify=10 then it doesnt mean asterisk shows extension lagged if latency is less than 10 ms ... It just checks every 10 ms for extension . I am not very sure though :) On 22/11/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Enrico Pasqualotto wrote: Enrico Pasqualotto wrote: hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? You have to set qualify=second instead of qualify=yes|no. This is WRONG. qualify=500 means consider this device lagged if responses take longer than 500ms I don't know if you can set the frequency of qualify packets. If you can, I assume the option would be listed in sip.conf.sample. ___ --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 ___ --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 ___ --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] Cisco media gateways in general
is possible to control ci$co gateway from asterisk via mgcp? i.e. asterisk as mgcp call agent? PJ Bas van der Veen wrote: Scott, Thanks for the reply. I am experiencing the following with a 2801: - user mistypes a phone number, so the number becomes non-existent - asterisk sends the call to the cisco - the cisco 2801 tries to connect to the non-existent number - the cisco sends a SIP 404 error to asterisk and the call is terminated This behaviour in itself is not weird, but the 2651 and 2821 routers at other branch offices for the same customer DO connect the user to the PSTN and they'd hear a message from the PSTN provider like this number is not in use. I'd like that with the 2801 as well. Would you happen to have the possibility to dial a non-existent number on this setup you mentioned and let me know what the result is? Regards, Bas On Tue, Nov 21, 2006 at 12:03:19PM -0500, Scott Keagy wrote: In my last job I set up Cisco 3845 with PRI cards, talking SIP to Asterisk. No problems there... main trick to get it working for me was to make sure Asterisk was not doing any authentication... add this to a line of the [peer] setup in sip.conf file on Asterisk: insecure=invite,port In terms of IOS side, if you are familiar with enabling sip UA and setting up dial peers, there is nothing special. Regards, Scott From: [EMAIL PROTECTED] on behalf of Bas van der Veen Sent: Tue 11/21/2006 10:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco media gateways in general Greetings, After the 0 respones I had on my previous mail regarding the Cisco 2801, I thought I'd be more general. Is anybody using Cisco media gateways at all? If so, how is it working for you? -- Kind regards, Meilleures salutations, Bas van der Veen GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x9E890160 The question of whether a computer can think is no more interesting than the question of whether a submarine can swim. --Edsger Dijkstra ___ --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 ___ --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] How ecord all calls?
Eric ManxPower Wieling wrote: Eugeniy Khvastunov wrote: Hi All!! Prompt how to record all calls passing through certain span? Next time I'll have coffee before hitting Reply. pbx-1*CLI show application monitor pbx-1*CLI -= Info about application 'Monitor' =- [Synopsis] Monitor a channel [Description] Monitor([file_format[:urlbase]|[fname_base]|[options]]): Used to start monitoring a channel. The channel's input and output voice packets are logged to files until the channel hangs up or monitoring is stopped by the StopMonitor application. file_format optional, if not set, defaults to wav fname_baseif set, changes the filename used to the one specified. options: m - when the recording ends mix the two leg files into one and delete the two leg files. If the variable MONITOR_EXEC is set, the application referenced in it will be executed instead of soxmix and the raw leg files will NOT be deleted automatically. soxmix or MONITOR_EXEC is handed 3 arguments, the two leg files and a target mixed file name which is the same as the leg file names only without the in/out designator. If MONITOR_EXEC_ARGS is set, the contents will be passed on as additional arguements to MONITOR_EXEC Both MONITOR_EXEC and the Mix flag can be set from the administrator interface b - Don't begin recording unless a call is bridged to another channel Returns -1 if monitor files can't be opened or if the channel is already monitored, otherwise 0. ___ --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] RE: Snom 360 Multiple calls on hold help
Yeah, doing more testing shows that the speed keys are broken, but dialing it works!!! Ugg!!! can you let me know if you get a new firmware? Im going to try and downgrade... Thanks! On 11/22/06, Alban [EMAIL PROTECTED] wrote: Yes, already. Waiting now for a new firmware... Regards, Alban Le Mercredi 22 Novembre 2006 13:50, Steve Davies a écrit: On 11/22/06, Alban [EMAIL PROTECTED] wrote: I'm having the same problem, pressing a speed dial/extension when 2 calls are on the phone connect the 2 calls together. Typing the number instead of using speed dial works. With older firmware, 6.2.1 or 6.3, it was working... But then other problem with pickup, deadlocking the phone (or slowing it down). Certainly due to the dp bug (fixed in 6.5.1). Regards, Alban. Has this been reported to snom by anyone? They are generally pretty good at fixing this type of issue and providing beta firmware. Regards, Steve ___ --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 ___ --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 ___ --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] G729 issues on 1.4 beta 3
Hello Everyone, I just upgraded to the latest beta version and I am running into one problem. We purchased g729a licenses from digium and they aren't loading anymore. If I roll back asterisk to 1.2.10 the codecs work fine. I've downloaded the new 1.4 version of the codec from their website and re-registerd everything with no luck. Here is the error message: error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: undefined symbol: ast_translator_activate I have tried i686, i386, athlon, and athlon-xp versions of the codec but none of them have loaded. Any help would be appreciated. Thanks, Jason Jason Adams Sumo Systems 4694 Cemetery Road Suite 310 Hilliard, OH 43026 Phone | 614.433.9906 ext: 102 Fax | 614.433.9931 E-mail | [EMAIL PROTECTED] ___ --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] Cisco media gateways in general
I've never used Asterisk MGCP, and I've only used MGCP gateway on Cisco IOS when controlled from Cisco CallManager (with PRI D-channels backhauled to CallManager). In terms of making an invalid number dialed via Asterisk to Cisco... behavior on Cisco side is entirely subject to how you've programmed the router. If you have no matching dialplan entry, then router will reject the call. If you put a catch-call type of dial peer (i.e. with a destination pattern that matches everything under the sun) and point it out a PSTN connection, then if the call is set up and plays an announcement back to the router with a please hang up and try again type of message, then it may be played back in-band in the audio stream if the router interprets it as an answered call, or it may reject the call toward Asterisk if the PSTN call leg is never established. I suspect this only works as you described when using analog PSTN connections to Cisco gateway, because the loop has to be closed to play back the audio announcement. If you had a PRI on the Cisco gateway, you'd probably get a reject message from the telco and send back a similar reject message toward Asterisk, even if you had the dial peer pointing toward the PSTN for the invalid number. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Wednesday, November 22, 2006 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco media gateways in general is possible to control ci$co gateway from asterisk via mgcp? i.e. asterisk as mgcp call agent? PJ Bas van der Veen wrote: Scott, Thanks for the reply. I am experiencing the following with a 2801: - user mistypes a phone number, so the number becomes non-existent - asterisk sends the call to the cisco - the cisco 2801 tries to connect to the non-existent number - the cisco sends a SIP 404 error to asterisk and the call is terminated This behaviour in itself is not weird, but the 2651 and 2821 routers at other branch offices for the same customer DO connect the user to the PSTN and they'd hear a message from the PSTN provider like this number is not in use. I'd like that with the 2801 as well. Would you happen to have the possibility to dial a non-existent number on this setup you mentioned and let me know what the result is? Regards, Bas On Tue, Nov 21, 2006 at 12:03:19PM -0500, Scott Keagy wrote: In my last job I set up Cisco 3845 with PRI cards, talking SIP to Asterisk. No problems there... main trick to get it working for me was to make sure Asterisk was not doing any authentication... add this to a line of the [peer] setup in sip.conf file on Asterisk: insecure=invite,port In terms of IOS side, if you are familiar with enabling sip UA and setting up dial peers, there is nothing special. Regards, Scott From: [EMAIL PROTECTED] on behalf of Bas van der Veen Sent: Tue 11/21/2006 10:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco media gateways in general Greetings, After the 0 respones I had on my previous mail regarding the Cisco 2801, I thought I'd be more general. Is anybody using Cisco media gateways at all? If so, how is it working for you? -- Kind regards, Meilleures salutations, Bas van der Veen GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x9E890160 The question of whether a computer can think is no more interesting than the question of whether a submarine can swim. --Edsger Dijkstra ___ --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 ___ --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 ___ --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] Siemens Gigaset SL75
I bought the phone in Germany. Except another wlan phone from Siemens which was not available any more, I did not find any alternatives to it. -Original Message- From: Olivier [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 22, 2006 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Siemens Gigaset SL75 This phone seems attractive but is not distributed in France. I wondered the reasons behind that. Just for curiosity, in which country did you buy it ? How would you compare it to alternatives ? Cheers 2006/11/21, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] : Hi, yes I tested this one week ago and it worked without problems. It is a nice wlan-phone with some (in my opinion) unnecessary features. Regards, Jens -Original Message- From: Olivier [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Friday, November 17, 2006 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Siemens Gigaset SL75 Hi, Has anyone tested Siemens Gigaset SL75 with Asterisk ? How would you rate its performances ? Cheers ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Hints no longer working in 1.4beta3 with Polycoms
Thanks, John - this confirms what we are seeing. 'show hints' output isn't changing, so it looks like a bug. I'll open one and see what happens. A. On Nov 21, 2006, at 5:44 PM, John Lange wrote: Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --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 ___ --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 ___ --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] Re: Rewriting caller ID from database?
There are two I can think of. Hoodahek and asterdex (or asteridex) We used hoodahek at first, but now use asterdex(sp?) It has a web interface to enter the new names into. We use it to fixup, corp. cell phones and used to use it for our leagcy PBX extensions. -- -- Steven http://www.glimasoutheast.org Vincent Delporte [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi Most of our customers have generic names like Hospital, so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as Reading Hospital so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you. ___ --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 ___ --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] Re: Rewriting caller ID from database?
Hi, You can do it using AstDB, just load the database with callerid names and numbers and then include a lookup on database in all incoming calls, so you can override whatever you wanted:) On 11/22/06, Steven [EMAIL PROTECTED] wrote: There are two I can think of. Hoodahek and asterdex (or asteridex) We used hoodahek at first, but now use asterdex(sp?) It has a web interface to enter the new names into. We use it to fixup, corp. cell phones and used to use it for our leagcy PBX extensions. -- -- Steven http://www.glimasoutheast.org Vincent Delporte [EMAIL PROTECTED] wrote in message news: [EMAIL PROTECTED] Hi Most of our customers have generic names like Hospital, so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as Reading Hospital so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you. ___ --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 ___ --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 -- Best regards, Marco Mouta ___ --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] channel_find_locked: Avoided deadlock ... messages - What to do?
What are these? Nov 22 09:35:23 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:25 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:25 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:26 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:26 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! ___ --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] Send event from dialplan
Gregory Duchatelet wrote:
Hi all,
Another question for today, hope an answer for this one…
I have a program talking with asterisk via the AMI. I receive events,
and I would like to insert some events in the dialplan, which could be
catch by my program.
Any idea how to do this ?
Greg
this easiest way is to use UserEvent below is an example
exten =
8,n,UserEvent(${IF(${ISNULL(${AGENTNUM})}?Queue:Schedule)}${ISTRANSFER}|CallerIDName:
${CALLERIDNAME})
the above from a dialplan would show up on the AMI as events that look like
(in this example AGENTNUM and ISTRANSFER are empty)
Event: Newexten
Privilege: call,all
Channel: Zap/17-1
Context: gdincoming
Extension: talk
Priority: 4
Application: UserEvent
AppData: Queue|CallerIDName: ~315CLD02-6945-true~
Uniqueid: 1156985743.5540
Event: UserEventQueue
Privilege: user,all
Channel: Zap/17-1
Uniqueid: 1156985743.5540
CallerIDName: ~315CLD02-6945-true~
side note: if you use ael then you will need to add a space after the |
otherwise it will fail
example
Playback(pls-hold-while-try);
UserEvent(Queue${ISTRANSFER}| CallerIDName: ${CALLERID(name)});
^^^ note the space in here
goodluck
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Re: [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms
http://bugs.digium.com/view.php?id=8405 On Nov 22, 2006, at 9:11 AM, Anthony Rodgers wrote: Thanks, John - this confirms what we are seeing. 'show hints' output isn't changing, so it looks like a bug. I'll open one and see what happens. A. On Nov 21, 2006, at 5:44 PM, John Lange wrote: Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --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] Cisco media gateways in general
I'm using a 2811 to transfer 4 digits to another Cisco gateway that connects to a NEC pbx. Working great when calls are originating from the Asterisk. When I try to call the Asterisk it is answering the calls, but not transferring them to the appropriate extension. Sam Little Me Childrenswear Bas van der Veen [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 11/21/2006 10:02 AM Please respond to Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To asterisk-users@lists.digium.com cc Subject [asterisk-users] Cisco media gateways in general Greetings, After the 0 respones I had on my previous mail regarding the Cisco 2801, I thought I'd be more general. Is anybody using Cisco media gateways at all? If so, how is it working for you? -- Kind regards, Meilleures salutations, Bas van der Veen GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x9E890160 The question of whether a computer can think is no more interesting than the question of whether a submarine can swim. --Edsger Dijkstra ___ --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 attlzi6b.dat Description: Binary data ___ --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] Request for working config for DISA
Here you go:
[Custom-CLID]
exten = s,1,Answer
exten = s,2,Authenticate(12345)
exten = s,15,Playback(after-the-tone)
exten = s,16,Playback(pls-entr-num-uwish2-call)
exten = s,18,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = s,19,Monitor(wav,${CALLFILENAME},m)
exten = s,20,DISA(no-password|from-internal|${CLIDArea})
On 11/22/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi Friends,
I have configured DISA. But, its not working. When I dial to my zap
channel, its asking to enter pin number. After entering PIN number, its
giving continuous engage sound and hangup. Can anybody send me correct
working configuration for DISA? Looking forward to your response. Thank you.
Regards,
Chandra.
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[asterisk-users] Asterisk On FreeBSD
Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task ___ --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] TE110P and TDM400P
Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four X100P - FXS). Both boards are recognized by the operating system as showed above: :08:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 64, IRQ 169 I/O ports at e800 [size=256] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 :08:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 193 I/O ports at e400 [size=256] Memory at febfe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 The problem is that I cant make the both cards to work together in the same server. Here is my /etc/zaptel.conf: ### fxsks=1-4 loadzone = us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 ### When I load the wctdm module, I get this error: ZT_CHANCONFIG failed on channel 5: No such device or address (6). Its sounds like the FXS module its tring to configure the channels 5 to 35 (E1 - ISDN Channels - TE110P card). Anybody already saw this ? Its possible to use this two cards in the same computer ? There is any separator that I can use in zaptel.conf to make the load of the modules dont mistakes itself ? Here is my versions: Debian kernel - 2.6.8 asterisk-1.2.12.1 libpri-1.2.4 zaptel-1.2.11 Thanks Lincoln___ --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 On FreeBSD
yep. email me offlist. I can help you. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid Exeplish Sent: Wednesday, November 22, 2006 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk On FreeBSD Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task ___ --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 On FreeBSD
On Wed, 22 Nov 2006, Dumpolid Exeplish wrote: Has anyone installed Asterisk on FreeBSD? i need help/steps on this task Im running Asterisk on FreeBSD, use the port in /usr/ports/net/asterisk /e -- http://hostname.nu/~emil ___ --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 On FreeBSD
Dumpolid Exeplish wrote: Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task ___ --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 http://archives.free.net.ph/message/20060618.125548.f385ddf1.en.html Complete how to -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Can anyone enlighten me as to what this means?
This happens when a call is offered to asterisk on a B-Channel that's already marked as used, I had the problem with one of my PRI provider, not hanging up calls but instead giving network congestion when users hung up... Trouble was solved at their side... Regards, Tristan Paul Hales a écrit : Are you connecting your Asterisk box to the outside world or a PABX? (I got this sort of error connecting an Asterisk box to a pabx..) PaulH On Tue, 2006-11-21 at 22:28 -0500, Matt wrote: We are doing PRIs into T4XXP cards. When I call out things are fine... however tonight sometimes on inbound calls I'd get: chan_zap.c: Duplicate setup requested on channel 0/1 already in use on span 1 in the full debug log followed by a fast busy signal on the calling parties end. Anyone know what would cause that? ___ --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 ___ --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 ___ --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] TE110P and TDM400P
I think that you are loading the drivers in the wrong order. You can change the order of loading are first define the E1 followed by the TDM400 Hope this helps, Henk _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln Zuljewic Silva Sent: woensdag 22 november 2006 20:51 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TE110P and TDM400P Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four X100P - FXS). Both boards are recognized by the operating system as showed above: :08:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 64, IRQ 169 I/O ports at e800 [size=256] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 :08:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 193 I/O ports at e400 [size=256] Memory at febfe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 The problem is that I cant make the both cards to work together in the same server. Here is my /etc/zaptel.conf: ### fxsks=1-4 loadzone = us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 ### When I load the wctdm module, I get this error: ZT_CHANCONFIG failed on channel 5: No such device or address (6). Its sounds like the FXS module its tring to configure the channels 5 to 35 (E1 - ISDN Channels - TE110P card). Anybody already saw this ? Its possible to use this two cards in the same computer ? There is any separator that I can use in zaptel.conf to make the load of the modules dont mistakes itself ? Here is my versions: Debian kernel - 2.6.8 asterisk-1.2.12.1 libpri-1.2.4 zaptel-1.2.11 Thanks Lincoln ___ --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] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V
On Wed, Nov 22, 2006 at 06:58:13PM +0100, Huib van Wees wrote: On 11/22/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Why Aastra phones use more electricity, i.e. 48VDC whereas other phones use much less, e.g. Grandstream and Linksys both use only 5VDC. I first thought it was because of PoE, but the ones with 5VDC also run fine on PoE. What is the difference in power consumption then? 48V is also a sort of standard for telco devices if I remember it correctly... Power is nothing to do with voltage (well it is, but not alone), you need the current too i.e. V * A. Pylon electricity lines run at very high voltage (several hundred thousand volts) or the current going down the lines would heat the cables and you'd lose a lot of power. 48V is just a telco standard, and most telco equipment (that runs in racks) is 48V. Probably because 110 (or 220/240 here in EU) is enough to electrocute an engineer, and 5V/12V would require too many Amps so wiring would have to be huge to carry the current. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] TE110P and TDM400P
This is the scenarios: 1 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 fxsks=1-4 loadzone = us defaultzone=us ### modprobe wcte11xp ZT_CHANCONFIG failed on channel 32: No such device or address (6) FATAL: Error running install command for wcte11xp 2 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 fxsks=1-4 loadzone = us defaultzone=us ### modprobe wctdm ZT_CHANCONFIG failed on channel 5: No such device or address (6) FATAL: Error running install command for wctdm 3 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxsks=32-35 loadzone = us defaultzone=us ### modprobe wcte11xpok modprobe wctdmok modprobe wcfxook modprobe wct4xxpok modprobe zaptelok ### /etc/asterisk/zapata.conf [channels] context=corsidian overlapdial=yes immediate=no callprogress=yes busydetect=no switchtype=euroisdn signalling=pri_net channel = 1-15,17-31 group=2 group=1 callgroup=1 pickupgroup=1 signalling=fxs_ks channel = 32-35 ### tail -f /var/log/asterisk/messages Nov 22 15:11:43 ERROR[5524] chan_zap.c: Channel 16 is reserved for D-channel. Nov 22 15:11:43 ERROR[5524] chan_zap.c: Unable to register channel '1-15' Nov 22 15:11:43 WARNING[5524] loader.c: chan_zap.so: load_module failed, returning -1 Nov 22 15:11:43 WARNING[5524] loader.c: Loading module chan_zap.so failed! - Original Message - From: Henk Dick To: 'Lincoln Zuljewic Silva' ; 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, November 22, 2006 4:08 PM Subject: RE: [asterisk-users] TE110P and TDM400P I think that you are loading the drivers in the wrong order. You can change the order of loading are first define the E1 followed by the TDM400 Hope this helps, Henk -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln Zuljewic Silva Sent: woensdag 22 november 2006 20:51 To: asterisk-users@lists.digium.com Subject: [asterisk-users] TE110P and TDM400P Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four X100P - FXS). Both boards are recognized by the operating system as showed above: :08:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 64, IRQ 169 I/O ports at e800 [size=256] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 :08:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 193 I/O ports at e400 [size=256] Memory at febfe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 The problem is that I cant make the both cards to work together in the same server. Here is my /etc/zaptel.conf: ### fxsks=1-4 loadzone = us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 ### When I load the wctdm module, I get this error: ZT_CHANCONFIG failed on channel 5: No such device or address (6). Its sounds like the FXS module its tring to configure the channels 5 to 35 (E1 - ISDN Channels - TE110P card). Anybody already saw this ? Its possible to use this two cards in the same computer ? There is any separator that I can use in zaptel.conf to make the load of the modules dont mistakes itself ? Here is my versions: Debian kernel - 2.6.8 asterisk-1.2.12.1 libpri-1.2.4 zaptel-1.2.11 Thanks Lincoln ___ --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] More than one asterisk process
Hi, Can somebody in the list tell me why sometimes when I do the TOP command I see more than one asterisk process ? Sometimes it appears and desappears again... Thanks, Ard. ___ --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] TE110P and TDM400P
I would suggest the following - remove the drivers - load them manually (zaptel, wcte11xp, wctdm) Run: Zttools - should show unconfigured cards. Take: /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxsks=32-35 loadzone = us defaultzone=us run: ztcfg -vv See what it is saying Hope this helps ___ --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] Diva Server, chan_capi and tone detection
On Tue, 21 Nov 2006, Gregory Duchatelet wrote: Hi all, I have a Diva Server V-BRI-2 card, which support, as written in reference guide: Extended tone processing (human talker detection, generation and detection of country-specific tones) I would like to detect human speech and fax tone with asterisk. I think that the diva card transmit a DTMF code when detecting voice, but chan_capi doesn't receive this DTMF code. I verbose it more, displaying all DTMF received, and only DTMF code CNG is received. Did you know how I can enable this detection (see DivaReportTones in Diva Server SDK) or how can I receive this DTMF in chan_capi ? This would require a change in chan-capi. To get the extended tone detection indications, additional request/parameter via CAPI must be issued. Another thing is, how do you want to get these indications for use in your dialplan? Armin ___ --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] Call park on Linksys 922 and similar phones?
I'm having an issue with call park on my new Linksys 922. It has soft menu keys for doing call transfer (which I always think is a good idea because it's amazing how every phone has a different xfer interface and people always get confused). However, I can't get a good call park working on it. It doesn't respond to the use of # for transfer (nor should I want it to, since it has soft transfer keys). If I hit xfer and call 700, the parker does announce the call being parked at 701, but then instead of disconnecting me I hear hold music on the 722 (and continue to hear hold music on the calling phone.) If I hit resume, I am back talking to the calling phone. If I hit xfer again (which is normally how to complete a transfer) both phones disconnect, and the console says that the 922 got tired of parking. --- I must admit, on a side note, I have never been particularly happy with the parking interface. I know a number of other people feel the same since there have been calls and development efforts for ways to improve it, including hints for BLF, shared/bridged line functionality etc. For the SOHO application, ie. a home pbx, the idea of a parking lot with numbered slots is generally overkill. Such a home is extremely unlikely to ever have more than one call parked in a pickup group, or per PBX frankly. I think a much nicer interface would be to have the first phone simply put the call on hold (which is the typical approach in many key systems) and then dial an extension to pick up the call that's on hold in my pickup group. If, as will rarely be the case, more than one call is on hold, I think the best way to deal with it would be to present an IVR that says: 3 Calls are on hold. Please enter the extension that placed the call on hold. Available extensions are 305, 49 and 902. But 99 times out of 100, the interface would amount to putting the call on hold, going to another phone and hitting the pick up held call speed dial. Which is what people tend to like in SOHO settings. You can sort of do this if you just insist there is only one parking slot, but it won't handle the rare double-hold case and it's much more to do when putting the call on hold. Any effort been made in this direction? ___ --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] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V
Power is nothing to do with voltage (well it is, but not alone), you need the current too i.e. V * A. Pylon electricity lines run at very high voltage (several hundred thousand volts) or the current going down the lines would heat the cables and you'd lose a lot of power. 48V is just a telco standard, and most telco equipment (that runs in racks) is 48V. Probably because 110 (or 220/240 here in EU) is enough to electrocute an engineer, and 5V/12V would require too many Amps so wiring would have to be huge to carry the current. The 48V standard came from what was the station battery. This dates back to very early telephone standards (think operators at desks with patch chords). The station battery is hooked up to power the equipment with the positive terminal at ground. On hook voltages (between tip and ring) were derived from this battery. Once a phone goes off hook with 600 Ohms of resistance the voltage across tip and ring drops to roughly 6 V. The power coming over the line is expected to be sufficient to power the phone. Station batteries were intended to be stable permanent power sources much an UPS except without the conversion back to AC. As a matter of further information regarding voltages: Ring voltage is double the 48 volts alternating at 20Hz. I believe that the number of amps that can be driven determines the maximum ringer equivalence rating for a circuit. Ringer equivalence is a non issue for most modern phones. Especially phones with external power sources. This is reproduced from memory. If I recalled incorrectly corrections are welcome. Shaun -- Visit my blog at http://hackerlog.blogspot.com = If more of us valued food and cheer and song above hoarded gold, it would be a merrier world. -- J.R.R. Tolkien ___ --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] Recordings.
Hey i said that as per his requirement as an example :) . His requirement is just around 20 calls . For a moderate server i think sata raid should be fine ..Heres some result posted by someone for recording calls on ram disk . http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/118497 On 22/11/06, Marcus Franke [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Vicky wrote: Yeh even a simple UDMA 5 enabled hard drive can handle 30 calls recording easily . Sata hard drives are even better . Hehe, UDMA sounds like EIDE drives.. nice to see they are fast enough, but I do not recommend those as server hardware. ;-) But, if John is going to buy a extra new server, he could use two drives in a mirror setup extra for recordings of these files. As it is not only the frequency of reading/writing these files but other accesses of the media like starting programs or reading/writing of logfiles that slowes down the access to the recorded audio files. Marcus -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZGUUqwWWw48OFWoRAvidAJwPSpTSuY6nwxKTDKI8fZDmshmbUgCgtWAp 27akzsEDv03q5CmlGMObo50= =2jAI -END PGP SIGNATURE- ___ --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 ___ --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] aastra 480i configuration help
I'm having problems getting my aastra 480i to register with the asterisk server. I can inititate calls from the phone, but sip show peers does not show any IP address registered for this phone. I am probably missing something stupidly simple. Anyone have an example config to share or corrections for my configuration? Asterisk 1.2 aastra 480i CT has the 1.4 firmware sip.conf [general] port = 5060 bindaddr = phone.pbzinc.loc disallow=all allow=ilbc allow=ulaw allow=alaw allow=gsm context=default canreinvite=no nat=no reinvite=no dtmfmode=info tos=0xB8 [tracey] type=friend disallow=all allow=ulaw allow=alaw dtmfmode=info host=dynamic username=tracey mailbox= context=internal callerid=tracey 868 aastra.cfg live dialpad: 1 suppress dtmf playback: 1 time server1: pro5.pbzinc.loc sip dial plan: X+#|XX+* sip proxy ip: phone.pbzinc.loc sip proxy port: 5060 sip dtmf method: 1 sip out-of-band dtmf: 0 sip line1 auth name: tracey sip line1 user name: 868 sip line1 display name: Tracey sip line1 screen name: Tracey sip line1 proxy ip: phone.pbzinc.loc sip line1 proxy port: 5060 sip line1 dtmf method: 1 sip use basic codecs: 1 handset list version: 1 handset1 name: Tracey's Cordless ___ --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] Call to disconnected number on PRI just rings
[EMAIL PROTECTED] wrote: Hi, Running Asterisk 1.2.12.1 when dialing a known disconnected number the calls just rings and rings. We never get the The number you are trying to reach If we dial the same number from an Asterisk 1.0.11 server again over PRI, we get the message on the 1st ring. Here is the PRI debug of such a call that just rings and rings. Any ideas? PRI debug sur CPL: -- Executing Dial(IAX2/3000-3, ZAP/g0/19056463061|120|r) in new stack You probably already got a response. I think the lists and myself are both a little behind. Take off the r option to Dial. Kevin ___ --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] cmd Record
Michael Welter wrote: When I record to a .wav file, I get gsm encoding. Is there a way to record using u-law encoding? The extension for ulaw is .ul Kevin ___ --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] Re: Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V
On Wed, 2006-11-22 at 12:01 -0700, [EMAIL PROTECTED] wrote: On 11/22/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi, Why Aastra phones use more electricity, i.e. 48VDC whereas other phones use much less, e.g. Grandstream and Linksys both use only 5VDC. I first thought it was because of PoE, but the ones with 5VDC also run fine on PoE. What is the difference in power consumption then? 48V is also a sort of standard for telco devices if I remember it correctly... IIRC, It's not just that 48 is a popular source. Most POE taps will regulate the voltage down to whatever they need, which often is just 5V, or 12V. But we are talking DC voltage here, and there are significant voltage drops due to the [small, but not zero] resistance of copper. The longer the cable from the injector to the tap, the bigger the resistance, and the more the voltage drop. The amount of current figures in there, also. So, 48V is a safer voltage in general to inject, as long lines will usually still see hopefully more than 5V at the tap end. If you are going to design networks with POE, you'd best pull out your calculator, multimeter, and V=IR equations, and see if you'll get the required voltage at the other end of the wire, given the current the devices will use. murf smime.p7s Description: S/MIME cryptographic signature ___ --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 On FreeBSD
I've installed on 6.1 it from ports with ztdummy without an issue. I've never used zaptel hardware on it though. Had some issues with meetme and ztdummy but all worked out. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of J. Oquendo Sent: Wednesday, November 22, 2006 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk On FreeBSD Dumpolid Exeplish wrote: Hi, Has anyone installed Asterisk on FreeBSD? i need help/steps on this task ___ --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 http://archives.free.net.ph/message/20060618.125548.f385ddf1.en.html Complete how to -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --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] snom subscriptions issue on WRT
We had the same problem with WRT54G with no Linksys Linux firmware. At that time the problem was WRT54G modified the devices IP address, i.e. Asterisk received the WRT54G IP address instead of device address. Solution was selecting NAT=yes. Hope this help Jorge tommaso.carrara wrote: Hi, I've just installed asterisk 1.2.1 on my openwrt distro ( I own a WRT54GL by Linksys ) . No problem by now, but I can see that my 3 snom 320, once they started they send subscriptions to asterisk, and I can see that running: sip show subscriptions But, after one hour about, OR when I do asterisk reload , asterisk losts all th snom subscriptions. Someone can help me please? Thanks ___ --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 ___ --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] Dialing from Placed Calls on Polycom IP501 doesn't always work
We narrowed this down to when the 'New Call' softkey was used to initiate the call. When this key was used, the corresponding 'Placed Calls' entry wouldn't work. Any other method of placing the call does work. An upgrade to 1.6.7 fixes the issue. CP On Nov 16, 2006, at 4:34 AM, John Marvin wrote: Noah Miller wrote: I never ran 1.6.6 for any length of time. 1.6.7 and 2.0.1 don't seem to suffer this issue. 2.0.1 has some buddy watch problems, so you may not want to use it, but 1.6.7 should be OK. I've been running 1.6.6 for quite a while, and I have been quite annoyed by this bug. However, the release notes for 1.6.7 did not mention fixing this problem, so I did not have any motivation for upgrading. But, since you said that you did not see the problem on 1.6.7 I decided to upgrade and see if the problem was fixed. It appears to have fixed it, although I can't be sure yet, because sometimes a call placed from the placed calls list did work on 1.6.6, so I don't have enough of a sample size yet to be sure the bug is gone. I sure hope it is. Thanks for the info. John ___ --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 ___ --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] Terrible, horrible firewall issues in * to * setup
My mission is to get one * box to dial another * box' extensions. I have set this up previously without any issues by making a simple IAX trunk/extension pair on the two boxes and create a dial plan with a prefix like 9|XXX to select an extension on the other box. My problem is that I now have to do this with extremely restrictive firewalls thrown into the mix - firewalls I have no control over. Basically, the setup is: *1 --- FW1 --- (Internet) --- FW2 --- FW3 --- *2 I have control over firewall 1 and 3, but not 2. Using port forwarding (4569 UDP) on FW1, I have been able to make calls from *2 to *1. My problem lies with making calls the other way, as I have no way of port forwarding on FW2. My initial thought was to set up a reverse SSH tunnel from *2 to *1, which would have worked fine if SSH would tunnel UDP (latency is a different matter altogether). I found a software called Zebedee (http://www.winton.org.uk/zebedee/) which claims to do UDP tunneling, and is able to do it in reverse, but I can't for the life of me get it to work. Before I try further with Zebedee, I thought it wise to ask the * community if there is a standard solution in this particular case, or perhaps if I'm attempting the impossible. Any input is greatly appreciated. ___ --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] Rewriting caller ID from database?
Vincent Delporte wrote:
Hi
Most of our customers have generic names like Hospital, so I need to
rewrite their caller ID name by looking up the number in a database on
the Asterisk server, and rewriting the name such as Reading Hospital
so that we know who's calling.
Any idea if this can be done with Asterisk, and how to do it?
Entries are stored in the internal database. Then at the incoming
context I have:
exten = _XX,n,Set(CALLERID(name)=${DB(CIDNAME/${CALLERIDNUM})})
exten = _XX,n,Set(CALLERID(number)=91${CALLERIDNUM})
Doug
-- Ben Franklin quote: Those who would give up Essential Liberty to
purchase a little Temporary Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Rewriting caller ID from database?
Most of our customers have generic names like Hospital, so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as Reading Hospital so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? I made a simple PHP AGI that takes the phone number and query a MySQL table to find the name assigned to this number. I still need to make a web interface to enter/modify the list but phpMyAdmin do the job for now. If you want it, just let me know. ___ --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] Zaptel - make b410p fails on Ubuntu 6.10
Hi, I've been able to make make install the Zaptel drivers (1.2). I'm using a b410p so I executed the following command make b410p. I tried this on multiple machines, but it always failes: [EMAIL PROTECTED]:/usr/src/zaptel-1.2.11# make b410p [ -f misdn-b410p.tar.bz ] || wget ftp://ftp.digium.com/pub/zaptel/b410p/misdn-b410p.tar.gz --23:59:54-- ftp://ftp.digium.com/pub/zaptel/b410p/misdn-b410p.tar.gz = `misdn-b410p.tar.gz' Resolving ftp.digium.com... 216.27.40.102, 69.16.138.164 Connecting to ftp.digium.com|216.27.40.102|:21... connected. Logging in as anonymous ... Logged in! == SYST ... done.== PWD ... done. == TYPE I ... done. == CWD /pub/zaptel/b410p ... done. == PASV ... done.== RETR misdn-b410p.tar.gz ... done. Length: 572,153 (559K) (unauthoritative) 100%[==] 572,153 61.22K/sETA 00:00 00:00:11 (38.14 KB/s) - `misdn-b410p.tar.gz' saved [572153] tar -zxf misdn-b410p.tar.gz make -C misdn install make[1]: Entering directory `/usr/src/zaptel-1.2.11/misdn' Makeing mISDN = cp /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/Makefile.v2.6 /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/Makefile export MINCLUDES=/usr/src/zaptel-1.2.11/misdn/include ; make -C /lib/modules/2.6.17-10-server/build SUBDIRS=/usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN modules CONFIG_MISDN_DRV=m CONFIG_MISDN_DSP=m CONFIG_MISDN_HFCMULTI=m make[2]: Entering directory `/usr/src/linux-headers-2.6.17-10-server' CC [M] /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.o In file included from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/channel.h:13, from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:20: include/linux/mISDNif.h:570: warning: âpackedâ attribute ignored for field of type âu_charâ include/linux/mISDNif.h:571: warning: âpackedâ attribute ignored for field of type âu_charâ In file included from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/channel.h:16, from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:20: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h: In function âmISDN_queueup_newheadâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:189: warning: implicit declaration of function âmISDN_queue_messageâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:189: error: âFLG_MSG_UPâ undeclared (first use in this function) /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:189: error: (Each undeclared identifier is reported only once /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:189: error: for each function it appears in.) /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h: In function âmISDN_queuedown_newheadâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:199: error: âFLG_MSG_DOWNâ undeclared (first use in this function) /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h: At top level: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:280: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â*â token In file included from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:20: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/channel.h: In function âqueue_ch_frameâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/channel.h:108: error: âFLG_MSG_UPâ undeclared (first use in this function) /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c: In function âwrite_ctrlâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:275: error: âmISDNinstance_tâ has no member named âprivatâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c: In function âhdlc_empty_fifoâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:409: error: âmISDNinstance_tâ has no member named âprivatâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c: In function âhdlc_fill_fifoâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:478: error: âmISDNinstance_tâ has no member named âprivatâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c: In function âhdlc_downâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:769: error: âmISDNinstance_tâ has no member named âhwlockâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:775: error: âmISDNinstance_tâ has no member named âhwlockâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:781: error: âmISDNinstance_tâ has no member named âhwlockâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:784: error: âmISDNinstance_tâ has no member named âhwlockâ
[asterisk-users] How to park calls on a specific extension
Currently at our office, if I want someone else to pick up a call, I have to transfer the call to them. So I'm looking into call parking, which is ALMOST perfect. The missing piece of the puzzle: I'm extension 203. I want any call I park to get parked at extension 2203. I want a call my boss parks to park at 2205, since he's ext. 205. In other words, I want calls parked FROM extension XYZ to be parked AT extension (XYZ+2000). I don't see a way to force parked calls to a specific extension. I'm probably just missing the answer, but I've googled for it and I can't find it. TIA for any help you can offer. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --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] Powering SNOM 200 phones?
A follow up on my message about my SNOM 200 phones now powering from my 802.3af Netgear FS108p PoE box. To follow up for those finding this thread on searches... I purchased some PowerDSine 6001 units (very cheap on ebay) and they power the SNOM 200 fine. Some Buffalo units also did this. So it seems that either the Netgear is too picky about its detection, or the SNOM 200 not fully compliant. The powerdsines are big and require an extra cable as all external injectors will, but they work. ___ --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
