Re: [asterisk-users] Very simple callback application needed
No, I do want call back. I want the caller to call a number, then hang up without it being answered. They then get a call-back and a dialtone, so they are now an extension on the PBX and can make calls. Danny Nicholas wrote: As I read this, it's not truly a callback; it's more of a notify; you call 555-1212 and want asterisk to call 555-1313? If this is actually the case, you would just do this in your dialplan: - exten = 5551212,1,dial(DAHDI/g1/5551313,60) This would effectively make asterisk do a new call to bridge A to B. If you wanted a non-bridged call, you could set up a call file and do this: - exten = 5551212,1,System(/bin/cp newcall.call /var/spool/asterisk/outgoing) - exten = 5551212,2,hangup Just my .02 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: Wednesday, September 02, 2009 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Very simple callback application needed I have need of a very simple callback function - when any call is made to a special SIP DID, the call is not answered but Asterisk then calls a pre-determined number - no need for CallerID to capture the calling number. Does anyone have a simple script to do this? Chris -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Payload size of 30ms
2 sep 2009 kl. 22.40 skrev Fred Posner: Here's the story... Nortel system set to use g711 @ 30ms payload ... Asterisk box would need to communicate to that box @ 30 ms and another end point at 20 ms. I've seen discussions of setting this to a different size, but seems to be limited to the entire codec and not on a per peer basis. Anyone have luck with this? The Asterisk can be 1.4 or 1.6.x... I've a preference for 1.6.0.x but it's not set in stone :) You can set it per peer in sip.conf when you add the allow= option. There's a readme that documents the payload sizes acceptable for different codecs. From sip.conf.sample: ;allow=ilbc ; see doc/rtp-packetization for framing options /O --- * Olle E Johansson - o...@edvina.net * Open Unified communication - Asterisk, Kamailio, Sip-router projects ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?
Meetme() is the way to go. Running it on a virtual machine might not be such a good idea bacause dahdi_dummy, needed for Meetme() might not run. Google on Meetme() cmd asterisk and check the parameters available. There is one for listen only mode. Don't forget to add a conference room to /etc/asterisk/meetme.conf All you need is a did from a sip dial tone provider with enough incoming lines, a straight asterisk install with dadhi_dummy loading, enough bandwidth for the connections (85 kbs per line) and a handfull of lines in /etc/asterisk/extensions.conf Make sure you use ulaw with the connection to you sip dialtone provider otherwise the asterisk server has to transcode all the channels from ulaw (used by Meetme() ) to whatever codec is used. Erik de Wild Tripple-o Your Asterisk migration partner the Netherlands Op 2 sep 2009 om 20:03 heeft li...@mgreg.com li...@mgreg.com het volgende geschreven:\ On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote: Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j Thanks for the posts thus far! In all honesty I'm looking for a complete in house solution. I don't mind spending up to $500-600 on equipment if necessary. I just want to know that when I'm done there are no residual costs, etc. Is Asterisk capable of this kind of setup/ management? As for labor, I'm willing to donate as much as is necessary. Thanks again, Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing still
3 sep 2009 kl. 00.27 skrev John A. Sullivan III: On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote: i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133...@216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport From:8592192438sip:8592192...@64.191.130.78;tag=as0707d433 To:sip:+185993133...@216.82.224.202 Contact:sip:8592192...@216.82.224.202 Call-ID: 0f3bdcc9171ef53148e7bab413aea...@64.191.130.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Sep 2009 21:10:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 412 v=0 o=root 3831 3831 IN IP4 216.82.224.202 s=session c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/9 a=rtpmap:34 H263/9 a=rtpmap:103 h263-1998/9 a=sendrecv snip I know very little about how ringing works but are they providing any kind of status information to you? Do you need to furnish the ring if they are not? It seems to me I saw quite a few articles about providing ring tone, what causes it to fail, and how to work around it. I assume you've searched for those already. Just a few thoughts - John It's very hard to say much without your configurations at hand. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very simple callback application needed
On 3/09/09 6:24 PM, Chris Mason (Lists) wrote: No, I do want call back. I want the caller to call a number, then hang up without it being answered. They then get a call-back and a dialtone, so they are now an extension on the PBX and can make calls. His second example will do that for you - although your callfile should probably go to a context with a wait before dial (so they have time to hang up) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] passing commands asterisk cli and getting output using PHP AGI
Hellos, I know this might be an easy one but either way I am stuck...I need to execute asterisk cli commands using php agi and get the output via the same script. How to I execute let's say show hints and get the output back to the script? I have tried $agi-exec(show hints); but I am getting the output below on the cli debug AGI Rx EXEC show hints AGI Tx 200 result=-2 AGI Rx VERBOSE EXEC show hints returned -2 1 AGI Tx 200 result=1 From My understanding -2 means failure to find application What am I doing wrong? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] passing commands asterisk cli and getting output using PHP AGI
On 3 Sep 2009, at 08:01, James Mutuku wrote: I know this might be an easy one but either way I am stuck...I need to execute asterisk cli commands using php agi and get the output via the same script. How to I execute let's say show hints and get the output back to the script? I have tried Use asterisk manager within the script. $asm-command(show hints); S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.722 problems with IAX
Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for ISDN also works good. But when I make a connection through IAX to another asterisk (having allow=g722 to really use G.722 in IAX) the voice is 'broken'. I also work on G.722 for twinklephone and encountered a special thing about G.722: It has a sample rate of 16000, but it announced as 8000 in SDP. Since I have similar problem with my G.722-twinkle implementation, it looks like the RTP and/or jitterbuffer code has a problem with that. Did I miss something here or is this really a bug? Thanks, Armin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Thank you for your reply . Do you mean my Asterisk extensions.conf must contain a line like the followings ? include = parkedcalls If so , can you please let me know where I have to put this line in my extensions.conf ? Thank you in advance Regards H.Motamedi On Thu, Sep 3, 2009 at 5:26 AM, Stephen Davies stephen.l.dav...@gmail.comwrote: In any event, the real problem is probably that you forgot to 'include = parkedcalls' in your dialplan. Steve On 9/2/09, Lyle Giese l...@lcrcomputer.net wrote: And now that the whole world of Asterisk has your sip user ids and passwords, you should change all of the passwords that are in that file and yes, change the passwords in all your phones. Lyle Giese LCR Computer Services, Inc. hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play prompt after hanup
Thanks. Is it possible to do the same after Queue command? After Queue command, hangup will hangup the call and won't go to the next priority. On Mon, Aug 17, 2009 at 7:22 PM, Trevor Hammondstre...@concipient.net wrote: On Monday, August 17, 2009, Rilawich Ango wrote: Thanks. DIALSTATUS works except ANSWER. When the phone hangup, the dialplan doesn't go to s-ANSWER. -- Executing [3...@default:12] Dial(SIP/10.31.0.32-09872150, SIP/3001|50|Tt) in new stack -- Called 3001 -- SIP/3001-0986d1d8 is ringing -- SIP/3001-0986d1d8 answered SIP/10.31.0.32-09872150 == Spawn extension (default, 3001, 12) exited non-zero on 'SIP/10.31.0.32-09872150' You need to ensure you specify the g option when you dial the destination (e.g. Dial(SIP/3001,50,Ttg)). Otherwise the call will jump to the h exten when either party hangs up. Sincerely, Trevor Hammonds ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sql error on trunk qualify....??
Hi, Whenever one of my trunks becomes unreachable or reachable again.. On logs i got the msg as follows: Jul 31 15:15:51] NOTICE[15112] chan_sip.c: Peer 'voiptrunk' is now Reachable. (12ms / 2000ms) [Jul 31 15:15:51] WARNING[15112] res_config_mysql.c: MySQL RealTime: Failed to query database. Check debug for more info. I dont wanna turn on the debug function because theres a lot of traffic goin on on this server and i dont wanna increase i/o load on it anymore.. And i cant just turn on the debug and block the traffic thru that trunk. SO.. turn on the debug and see what happens is not an option for me right now. So i wonder if it tries to write that errors to a field on db and i forgot to create it?? But if so, why in other errors it doesnt give such error also?? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voipbuster not ringing, other SIP peers are ringing...
Francesco Peeters wrote: Francesco Peeters wrote: Does anybody else see the same behavior for VoipBuster connections? When I trace one of the other SIP peers, I see it sends this message: -- --- SIP read from 82.101.62.99:5060 --- SIP/2.0 180 Ringing Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl Contact: sip:82.101.62.99:5060 Content-Type: application/sdp CSeq: 103 INVITE From: "**" sip:***...@sip.xs4all.nl;tag=as70e84199 Record-Route: sip:82.101.62.115;lr;r2=on;ftag=as70e84199,sip:82.101.63.5;lr;r2=on;ftag=as70e84199 Server: Cirpack/v4.41b (gw_sip) To: sip:0031*...@sip.xs4all.nl;tag=00-08168-044b6f36-245cd72c7 Via: SIP/2.0/UDP ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92 Content-Length: 182 v=0 o=cp10 125193221174 125193221174 IN IP4 82.101.62.66 s=SIP Call c=IN IP4 194.109.8.2 t=0 0 m=audio 36984 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:20 a=sendrecv - --- (12 headers 10 lines) --- Found RTP audio format 8 Peer audio RTP is at port 194.109.8.2:36984 Found audio description format PCMA for ID 8 Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.109.8.2:36984 -- SIP/*-089ca9b8 is ringing -- SIP/*-089ca9b8 is making progress passing it to IAX2/2104-2287 Scheduling destruction of SIP dialog '740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 82.101.62.99:5060: CANCEL sip:0031**...@sip.xs4all.nl SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport From: "**" sip:**...@sip.xs4all.nl;tag=as70e84199 To: sip:0031**...@sip.xs4all.nl Call-ID: 740540ee64fa957513ce89f03ef5e...@sip.xs4all.nl CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- However when I dial exactly the same from VoipBuster, I see this instead: -- --- SIP read from 77.72.169.129:5060 --- SIP/2.0 183 Session progress Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport From: "*" sip:**...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com;tag=120113ac4a54a269af9e2c Contact: sip:0031**...@77.72.169.129:5060 Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 162 v=0 o=* 1251932194 1251932194 IN IP4 194.221.62.33 s=SIP Call c=IN IP4 194.221.62.33 t=0 0 m=audio 8958 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 - --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 194.221.62.33:8958 Found audio description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 194.221.62.33:8958 -- SIP/-089dc538 is making progress passing it to IAX2/2104-8077 == Connect attempt from '127.0.0.1' unable to authenticate Scheduling destruction of SIP dialog '1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to 77.72.169.129:5060: CANCEL sip:0031**...@sip.voipbuster.com SIP/2.0 Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport From: "**" sip:***...@sip.voipbuster.com;tag=as1374705a To: sip:0031**...@sip.voipbuster.com Call-ID: 1949e0303d52a19b1b4f91f16ff94...@sip.voipbuster.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- As you can see, there are different packets being sent, and in the 2nd case, there is no "is ringing" message, which is rather irritating... Any suggestions would be appreciated... TIA BTW: I am talking about the ringtone the caller should hear... The other side is ringing, and calls are established just fine, but it is very irritating to hear nothing until the call either fails or gets picked up... NM! Found out this only happens on a single extension, and that one was using IAX... Changed it to SIP as well and got ringing there too! -- FP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing
Re: [asterisk-users] passing commands asterisk cli and getting output using PHP AGI
I have included that but my scripts goes silent at
AGI Rx EXEC Flite Hello 1215, you have dialed 1220.
AGI Tx 200 result=0
Below is my script
#!/usr/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$asm = $agi-new_AsteriskManager();
$agi-answer();
$callext = $agi-get_variable(DNID);
$callext=$callext['data'];
$callid = $agi-get_variable(CALLERID(num));
$callid=$callid['data'];
$agi-exec(Flite,\Hello $callid, you have dialed $callext.\);
$asm-command(show hints);
$agi-exec(flite,\Goodbye\);
$agi-hangup();
?
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Re: [asterisk-users] Versions of Asterisk 1.6
Hi David, Is T.38 Fax supported on both? I can tell you that I've been having problems with various version of Cisco IOS and T.38 on asterisk. I had a stable configuration fax-wise, but I had to upgrade the IOS because of a Cisco bug, and my T.38 has never been the same since. It's hard to blame asterisk for that problem. In fact, if you read through the T.38 bugs in Cisco IOS release notes it makes asterisk T.38 look solid by comparison. If downgrading didn't make my router freeze I'd downgrade the IOS. We are also having problems of interoperability between asterisk and CISCO. What version of the IOS was working for you? Thanks, Santi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GTalk functionality Asterisk
Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them . and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are available. Problem :- In order to Gtalk work with Asterisk ... We need to have make some changes in menu config.. ie in channel drivers we need to select chan_gtalk. but when i execute make manuconfig . it appears Applications [*] chan_agent Call Detail Recording [*] chan_alsa Channel Drivers XXX chan_console Codec Translators [*] chan_dahdi Format Interpreters XXX chan_gtalk Dialplan FunctionsXXX chan_h323 PBX Modules [*] chan_iax2 Resource Modules XXX chan_jingle Test Modules [*] chan_local the XXX makes me worry. how to remove this ?? include chan_gtalk also -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommendations about infrastructure to use with Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm investigating the possibility of using Asterisk as much for internal communication in an office as between offices and I would like to know what considerations you could comment to me being based on the experience that you have had. A priori two things come to my mind: * As to network topology, is advisable to have switches and dedicated networks for to use with the extensions? * Is advisable to have a dedicated Internet connection for intercommunication between the different offices? I imagine that yes, since of another way the VoIP traffic would have to compete with the rest and in that case we would require to apply some additional technique of QoS. In this point also I would include the optimal bandwidth that would have to have the dedicated link, for the case of using something of this type. Perhaps there is some other interesting questions that also it is necessary to consider. In order to give more additional information, the Internet connection between the different offices is made at the moment through two links of 2 Mbps, with load balance (one of fiber and another one of microwaves). The amount of extensions in one of the offices would be approximately of 50, whereas in the other there would be approximately about 80 extensions. Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkqfjPAACgkQZpa/GxTmHTfm8ACfXUHf8helAFxo5Tqmjk6TCiq2 5CwAnAyfGsCVEL+6g7O2juTPnLh9gHIj =v8+9 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe unactive pin access
Hello, I have conferences in my database. I need at some moments, to access the database without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GTalk functionality Asterisk
On 14:24, Thu 03 Sep 09, ABBAS SHAKEEL wrote: Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them . and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are available. Problem :- In order to Gtalk work with Asterisk ... We need to have make some changes in menu config.. ie in channel drivers we need to select chan_gtalk. but when i execute make manuconfig . it appears Applications [*] chan_agent Call Detail Recording [*] chan_alsa Channel Drivers XXX chan_console Codec Translators [*] chan_dahdi Format Interpreters XXX chan_gtalk Dialplan FunctionsXXX chan_h323 PBX Modules [*] chan_iax2 Resource Modules XXX chan_jingle Test Modules [*] chan_local the XXX makes me worry. how to remove this ?? include chan_gtalk also Use your arrow key to select chan_gtalk and chan_jingle. It will show on the bottom of your screen what you need. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe unactive pin access
Sorry, there are some errors, here the right question: Hello, I have conferences in my database. I need at some moments, to access the CONFEERENCE without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 http://www.acropolistelecom.net/ http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ François Envoyé : jeudi 3 septembre 2009 11:37 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] MeetMe unactive pin access Hello, I have conferences in my database. I need at some moments, to access the database without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI)
Karl Fife wrote: Any theories as to why one routine would behave differently than the other with Echo Cancellation enabled? In my mind, anything that alters the audio path may cause issues with DTMF detection. As to why, I'm not qualified to say; I'm not a programmer. You may want to check out http://issues.asterisk.org and see if there are any bugs open on the subject. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe unactive pin access
I found ! If I need to enter in a conference (without pinacces) which is in the database (and have a pin access), Just add ,thepinacces at the end of meetme! Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ François Envoyé : jeudi 3 septembre 2009 11:42 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] MeetMe unactive pin access Sorry, there are some errors, here the right question: Hello, I have conferences in my database. I need at some moments, to access the CONFEERENCE without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 http://www.acropolistelecom.net/ http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ François Envoyé : jeudi 3 septembre 2009 11:37 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] MeetMe unactive pin access Hello, I have conferences in my database. I need at some moments, to access the database without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - log rotation
Hi, It seems Asterisk needs to be notified that log rotation happened tough applications like astmanproxy or FOP doesn't need to be restarted (nor notified of any rotation). Is this personal observation true ? How could this be explained ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - log rotation
On 3 Sep 2009, at 11:39, Olivier wrote: It seems Asterisk needs to be notified that log rotation happened tough applications like astmanproxy or FOP doesn't need to be restarted (nor notified of any rotation). Is this personal observation true ? How could this be explained ? They don't use the log? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Payload size of 30ms
On Sep 3, 2009, at 2:34 AM, Olle E. Johansson wrote: 2 sep 2009 kl. 22.40 skrev Fred Posner: Here's the story... Nortel system set to use g711 @ 30ms payload ... Asterisk box would need to communicate to that box @ 30 ms and another end point at 20 ms. I've seen discussions of setting this to a different size, but seems to be limited to the entire codec and not on a per peer basis. Anyone have luck with this? The Asterisk can be 1.4 or 1.6.x... I've a preference for 1.6.0.x but it's not set in stone :) You can set it per peer in sip.conf when you add the allow= option. There's a readme that documents the payload sizes acceptable for different codecs. From sip.conf.sample: ;allow=ilbc ; see doc/rtp-packetization for framing options /O --- * Olle E Johansson - o...@edvina.net * Open Unified communication - Asterisk, Kamailio, Sip-router projects Fantastic, thank you! Fred Posner ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to Disable CDR for callfile?
I want to do a callback scenario. Each time asterisk receive a call, it creates a callfile, sends back the hangup signal and dial back the extension. Here the default CDR logging is enabled. If a dial attempt is failed then a CDR is generated. How I do a trick to stop CDR logging for all callfiles, without changing the default behaviour of CDR logging. I know its NoCDR() function that will disable CDR() logging, But how it will be done in callfiles ? Thanks, M. Faheem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does L(x:y:z) Dial option work on Asterisk version 1.4 ?
Sorry guys.
My bad!
As you can see, the command on prior message is incorret.
I've changed to:
Dial(SIP/${EXTEN}|20|RtTL(30:6:2))
and it's working now.
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br mailto:@tqi.com.br
: www.tqi.com.br http://www.tqi.com.br
( + 55 (34)3291-1700
( + 55 (34)9971-2572
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Re: [asterisk-users] How to Disable CDR for callfile?
Have your callfile work through a context instead of dialing. The context can disable CDR. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faheem Sent: Thursday, September 03, 2009 7:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Disable CDR for callfile? I want to do a callback scenario. Each time asterisk receive a call, it creates a callfile, sends back the hangup signal and dial back the extension. Here the default CDR logging is enabled. If a dial attempt is failed then a CDR is generated. How I do a trick to stop CDR logging for all callfiles, without changing the default behaviour of CDR logging. I know its NoCDR() function that will disable CDR() logging, But how it will be done in callfiles ? Thanks, M. Faheem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722 problems with IAX
On Thursday 03 September 2009 02:47:05 Armin Schindler wrote: Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for ISDN also works good. But when I make a connection through IAX to another asterisk (having allow=g722 to really use G.722 in IAX) the voice is 'broken'. I also work on G.722 for twinklephone and encountered a special thing about G.722: It has a sample rate of 16000, but it announced as 8000 in SDP. Since I have similar problem with my G.722-twinkle implementation, it looks like the RTP and/or jitterbuffer code has a problem with that. Did I miss something here or is this really a bug? You missed that the IETF has a typo in the specification, stating that G.722 is to be stated as 8000, even though it's 16000. This will remain, due to backwards compatibility concerns. Please see RFC 3551, section 4.5.2. http://www.apps.ietf.org/rfc/rfc3551.html#sec-4.5.2 -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Originate calls with AMI.
Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: Action: originate Channel: SIP/zoiper Exten: yziquel Priority: 1 Timeout: 30 Context: internal Response: Error Message: Originate failed Event: Newchannel Privilege: call,all Channel: SIP/zoiper-019a3000 State: Down CallerIDNum: unknown CallerIDName: unknown Uniqueid: asterisk-1251987055.7 Event: Newcallerid Privilege: call,all Channel: SIP/zoiper-019a3000 CallerID: Unknown CallerIDName: Unknown Uniqueid: asterisk-1251987055.7 CID-CallingPres: 0 (Presentation Allowed, Not Screened) Event: Hangup Privilege: call,all Channel: SIP/zoiper-019a3000 Uniqueid: asterisk-1251987055.7 Cause: 0 Cause-txt: Unknown And then the 'zoiper' softphone starts ringing continuously. It says Incoming Call from asterisk and not from 'yziquel'. Moreover when I pick up the phone it says You are now talking to asterisk, and then Zoiper closes the call immediately. There's surely something I do not get right here, and I'd appreciate some help. All the best, Guillaume YZiquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] probleme with web-meetme.3.1.0
Hi everybody I have a problem and want to know if anyone has already seen it before : I try to use web-meetme.3.1.0 and follow these instructions http://sourceforge.net/docman/display_doc.php?docid=48924group_id=164788 1) when i do make command in cbmysql folder, errors happened * cc -pipe -I/usr/include/mysql -L/usr/lib/mysql -fPIC -I/usr/src/asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_cbmysql.o app_cbmysql.c app_cbmysql.c:584: attention : initialization from incompatible pointer type app_cbmysql.c:585: attention : initialization from incompatible pointer type app_cbmysql.c:585: attention : initialization makes integer from pointer without a cast app_cbmysql.c:594:38: erreur: macro « ast_config_load » requiert 2 arguments, mais seulement 1 ont été passés app_cbmysql.c: In function ‘load_config’: app_cbmysql.c:594: erreur: ‘ast_config_load’ undeclared (first use in this function) app_cbmysql.c:594: erreur: (Each undeclared identifier is reported only once app_cbmysql.c:594: erreur: for each function it appears in.) make: *** [app_cbmysql.o] Erreur 1 * when I try to go on the web page I have these messages * Notice: Undefined variable: s in /var/www/html/web-meetme-v3.1.0/meetme_control.php on line 9 Notice: Undefined variable: logoff_section in /var/www/html/web-meetme-v3.1.0/meetme_control.php on line 12 Notice: Undefined variable: logoff_section in /var/www/html/web-meetme-v3.1.0/meetme_control.php on line 19 Notice: Undefined index: auth in /var/www/html/web-meetme-v3.1.0/meetme_control.php on line 29 Notice: Undefined variable: AUTH_USER in /var/www/html/web-meetme-v3.1.0/meetme_control.php on line 39 Notice: Undefined index: auth in /var/www/html/web-meetme-v3.1.0/meetme_control.php on line 45 Notice: Undefined index: privilege in /var/www/html/web-meetme-v3.1.0/lib/header.inc on line 8 Notice: Undefined index: auth in /var/www/html/web-meetme-v3.1.0/lib/header.inc on line 28 Notice: Undefined variable: logoff_sel in /var/www/html/web-meetme-v3.1.0/lib/header.inc on line 35 * note that I want to use sqldb.conf for users authentication and not ldap. Regards Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi configuraion / error
Tzafrir Cohen escreveu: On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard this is an R23 connection, so I dont think genconfig will help. Also, I already had this working but not its not working... I dont know why (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? Yes, it has a cable connected (in port 1) to a Telco, so Im configuerd as slave. Here is the dahdi_cfg -vv ( sorry for the long post ) [r...@catumbela modules]# dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: D-channel (Default) (Echo Canceler: none) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 24) Channel 25: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 25) Channel 26: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 26) Channel 27: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 27) Channel 28: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 28) Channel 29: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 29) Channel 30: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 30) Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31) Channel 32: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 32) Channel 33: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 33) Channel 34: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 34) Channel 35: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 35) Channel 36: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 36) Channel 37: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 37) Channel 38: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 38) Channel 39: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 39) Channel 40: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 40) Channel 41: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 41) Channel 42: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 42) Channel 43: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 43) Channel 44: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 44) Channel 45: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 45) Channel 46: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 46) Channel 47: D-channel (Default) (Echo Canceler: none) (Slaves: 47) Channel 48: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 48) Channel 49: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 49) Channel 50: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 50) Channel 51: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 51) Channel 52: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 52) Channel 53: Clear channel
Re: [asterisk-users] Dahdi configuraion / error
No such device is sometimes an indication that /etc/init.d/dahdi start did not load the driver. What does /etc/dahdi/modules look like? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, September 03, 2009 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error Tzafrir Cohen escreveu: On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard this is an R23 connection, so I dont think genconfig will help. Also, I already had this working but not its not working... I dont know why (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? Yes, it has a cable connected (in port 1) to a Telco, so Im configuerd as slave. Here is the dahdi_cfg -vv ( sorry for the long post ) [r...@catumbela modules]# dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: D-channel (Default) (Echo Canceler: none) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 24) Channel 25: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 25) Channel 26: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 26) Channel 27: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 27) Channel 28: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 28) Channel 29: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 29) Channel 30: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 30) Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31) Channel 32: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 32) Channel 33: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 33) Channel 34: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 34) Channel 35: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 35) Channel 36: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 36) Channel 37: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 37) Channel 38: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 38) Channel 39: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 39) Channel 40: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 40) Channel 41: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 41) Channel 42: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 42) Channel 43: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 43) Channel 44: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 44) Channel 45: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 45) Channel 46: Clear channel (Default) (Echo Canceler: mg2)
Re: [asterisk-users] Dahdi configuraion / error
it looks like this: tail /etc/dahdi/modules # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on Wed Jun 24 12:41:26 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. wct4xxp Danny Nicholas escreveu: No such device is sometimes an indication that /etc/init.d/dahdi start did not load the driver. What does /etc/dahdi/modules look like? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, September 03, 2009 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error Tzafrir Cohen escreveu: On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard this is an R23 connection, so I dont think genconfig will help. Also, I already had this working but not its not working... I dont know why (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? Yes, it has a cable connected (in port 1) to a Telco, so Im configuerd as slave. Here is the dahdi_cfg -vv ( sorry for the long post ) [r...@catumbela modules]# dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: D-channel (Default) (Echo Canceler: none) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 24) Channel 25: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 25) Channel 26: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 26) Channel 27: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 27) Channel 28: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 28) Channel 29: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 29) Channel 30: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 30) Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31) Channel 32: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 32) Channel 33: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 33) Channel 34: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 34) Channel 35: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 35) Channel 36: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 36) Channel 37: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 37) Channel 38: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 38) Channel 39: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 39) Channel 40: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 40) Channel 41: Clear channel (Default) (Echo
Re: [asterisk-users] Dahdi configuraion / error
here are my logs when I start the dahdi driver: /etc/rc.d/init.d/dahdi start Sep 3 18:02:39 catumbela kernel: Found TE4XXP at base address fdcff000, remapped to f88a8000 Sep 3 18:02:39 catumbela kernel: TE4XXP version c01a0164, burst OFF Sep 3 18:02:39 catumbela kernel: FALC version: 0005, Board ID: 00 Sep 3 18:02:39 catumbela kernel: Reg 0: 0x2afea400 Sep 3 18:02:39 catumbela kernel: Reg 1: 0x2afea000 Sep 3 18:02:39 catumbela kernel: Reg 2: 0x Sep 3 18:02:39 catumbela kernel: Reg 3: 0x Sep 3 18:02:39 catumbela kernel: Reg 4: 0x Sep 3 18:02:39 catumbela kernel: Reg 5: 0x Sep 3 18:02:39 catumbela kernel: Reg 6: 0xc01a0164 Sep 3 18:02:39 catumbela kernel: Reg 7: 0x1f00 Sep 3 18:02:39 catumbela kernel: Reg 8: 0x010200ff Sep 3 18:02:39 catumbela kernel: Reg 9: 0x00fd Sep 3 18:02:39 catumbela kernel: Reg 10: 0x004a Sep 3 18:02:39 catumbela kernel: Found a Wildcard: Wildcard TE405P (2nd Gen) Sep 3 18:02:39 catumbela kernel: TE4XXP: Launching card: 0 Sep 3 18:02:39 catumbela kernel: TE4XXP: Setting up global serial parameters Sep 3 18:02:39 catumbela dahdi: wct4xxp: succeeded Sep 3 18:02:39 catumbela kernel: About to enter spanconfig! Sep 3 18:02:39 catumbela kernel: Done with spanconfig! Sep 3 18:02:39 catumbela kernel: About to enter startup! Sep 3 18:02:39 catumbela kernel: TE4XXP: Span 1 configured for CAS/HDB3 Sep 3 18:02:39 catumbela kernel: timing source auto card 0! Sep 3 18:02:39 catumbela kernel: wct4xxp: Setting yellow alarm on span 1 Sep 3 18:02:39 catumbela kernel: timing source auto card 0! Sep 3 18:02:39 catumbela kernel: SPAN 1: Primary Sync Source Sep 3 18:02:39 catumbela kernel: VPM400: Not Present Sep 3 18:02:39 catumbela kernel: VPM450: Not Present Sep 3 18:02:39 catumbela kernel: Completed startup! Sep 3 18:02:39 catumbela dahdi: Running dahdi_cfg: succeeded Joao Gomes Pereira escreveu: it looks like this: tail /etc/dahdi/modules # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on Wed Jun 24 12:41:26 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. wct4xxp Danny Nicholas escreveu: No such device is sometimes an indication that /etc/init.d/dahdi start did not load the driver. What does /etc/dahdi/modules look like? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, September 03, 2009 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error Tzafrir Cohen escreveu: On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard this is an R23 connection, so I dont think genconfig will help. Also, I already had this working but not its not working... I dont know why (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? Yes, it has a cable connected (in port 1) to a Telco, so Im configuerd as slave. Here is the dahdi_cfg -vv ( sorry for the long post ) [r...@catumbela modules]# dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo
Re: [asterisk-users] Dahdi configuraion / error
Here it is: [r...@catumbela ~]# lsmod|grep wct4xxp wct4xxp 242176 0 dahdi 197640 5 wct4xxp [r...@catumbela ~]# dmesg is in attach :) Danny Nicholas escreveu: Okay. What is the output of these commands? dmesg lsmod|grep wct4xxp -Original Message- From: Joao Gomes Pereira [mailto:gomespere...@startel.pt] Sent: Thursday, September 03, 2009 11:56 AM To: Danny Nicholas; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error it looks like this: tail /etc/dahdi/modules # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) on Wed Jun 24 12:41:26 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. wct4xxp Danny Nicholas escreveu: No such device is sometimes an indication that /etc/init.d/dahdi start did not load the driver. What does /etc/dahdi/modules look like? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Thursday, September 03, 2009 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi configuraion / error Tzafrir Cohen escreveu: On Tue, Sep 01, 2009 at 08:53:04AM -0500, Danny Nicholas wrote: This may be dumb and/or obvious, but did you do these steps? 1. dahdi_genconf dahdi modules user to make sure all of the configuration files are up to standard this is an R23 connection, so I dont think genconfig will help. Also, I already had this working but not its not working... I dont know why (Which will default to generate a ccs configuration for it, rather than cas) Current configuration appears to be OK at first glance. 2. dahdi_cfg -vv to see if any obvious messages came up (causing the red condition(s) ) The information you have provided is useful, but it boils down to this (IMO) - RED is dead! Is there actually a cable plugged? Connecting it to a live system? Yes, it has a cable connected (in port 1) to a Telco, so Im configuerd as slave. Here is the dahdi_cfg -vv ( sorry for the long post ) [r...@catumbela modules]# dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: D-channel (Default) (Echo Canceler: none) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 24) Channel 25: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 25) Channel 26: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 26) Channel 27: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 27) Channel 28: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 28) Channel 29: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 29) Channel 30: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 30) Channel 31: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 31) Channel 32: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 32) Channel 33: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 33) Channel 34: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 34) Channel 35:
Re: [asterisk-users] Recommendations about infrastructure to use with Asterisk
On Thu, 2009-09-03 at 06:30 -0300, Daniel Bareiro wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! I'm investigating the possibility of using Asterisk as much for internal communication in an office as between offices and I would like to know what considerations you could comment to me being based on the experience that you have had. A priori two things come to my mind: * As to network topology, is advisable to have switches and dedicated networks for to use with the extensions? * Is advisable to have a dedicated Internet connection for intercommunication between the different offices? I imagine that yes, since of another way the VoIP traffic would have to compete with the rest and in that case we would require to apply some additional technique of QoS. In this point also I would include the optimal bandwidth that would have to have the dedicated link, for the case of using something of this type. Perhaps there is some other interesting questions that also it is necessary to consider. In order to give more additional information, the Internet connection between the different offices is made at the moment through two links of 2 Mbps, with load balance (one of fiber and another one of microwaves). The amount of extensions in one of the offices would be approximately of 50, whereas in the other there would be approximately about 80 extensions. snip I'll begin by saying there are others on this list with much more experience than I. Given that reservation, it is the unending balance between cost and performance. It would be ideal to have two separate networks if it is affordable. Integration between the PBX and the data network, e.g., integrating voice mail and email, can be done via a separate interface on the PBX. There are challenges in using the Internet for inter-office connectivity. Better to use a private WAN or Internet connections with a single carrier who will honor Class of Service settings. Once one dumps one's traffic onto the Internet, there are no guarantees that real time traffic will be prioritized over bulk traffic. Since the greatest point of congestion is probably the last mile, you may be able to make a creative deal with your carrier to implement CoS at your upstream router. That will prioritize ingress traffic turning off the super-highway and onto your back road. Likewise, you can implement CoS on your Internet gateway to prioritize egress traffic. These prioritization methods can help not only your inter-office traffic flow but can also be used if you cannot afford separate internal networks. We implemented CoS in our switches and chose settings on the end points to take advantage of the defaults settings for our Linux gateways and devices. Specifically, our end points set the DSCP bits to 101100 rather than 101110 (expedited forwarding) - b0 instead of b8 in Asterisk, 176 instead of 184 in Snom, 44 in iptables. This is because the Linux default pfifo-fast packet queueing looks at a different set of bits in the same TCP field and consequently places 101110 in a lower priority queue (band 1 I believe - the default band) than 101100 (I believe band 0). We set our switches to map packets with DSCP 101100 into the highest priority queue. We elected not to change our MTU to 576 (typical default is 1500) but this improves quality on highly congested lines. The voice packets may be prioritized but, if a big packet sneaks in while the voice queue is momentarily empty, it may take a while to transmit depending on the bandwidth of the link. You should not implement jumbo packets on a shared network for this reason. You may also have some security considerations in a shared network. We have found that connection tracking / stateful inspection support is spotty for SIP traffic. This is especially true if you set canreinvite=nonat or yes to try to shift the media stream away from Asterisk and to the end devices. We have found that most of the conntrack mechanisms do a good job tracking the shift from the SIP port to the RTP port when the call is passing through Asterisk. Some struggle when Asterisk reinvites the call. We haven't tracked it down but it appears that iptables takes about 30 seconds to kick in - we suspect it does not make the proper association until there is a SIP exchange between the new end points but we have not confirmed that. Even if we get around that problem, we have a bigger mess if one end point speaking directly with another end point transfers the call. The option is to allow access no only to the SIP port but to some or all high UDP ports. We were very unhappy with that arrangement but, after struggling unsuccessfully to have iptables pick up these transfer scenarios, we opted for a compromise where we open up the high UDP ports to hard phones but do not for soft phones since we do not want to expose any user applications which also happen to be listening on high UDP ports to
[asterisk-users] transcoder card
hi folks. i have several remote sites with total of 200 sip phones connect to our Asterisk server. i want to minimize bandwidth usage and thinking about getting a Digium TC400B transcoder card. what are your experience with it? how's the quality? also if there are 120 active channels in used. will the 121 person able to make calls? will it support more channels if i put 2 cards in the system? thanks. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Noises on Batphones
Hello, The company I work for recently purchased 2 Rhino CB24s and a Rhino PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2 PRIs from our telco. The CB24s are for all internal analog phones. Most of the phones are setup in batphone mode, which is immediate=on in the DAHDI config. They are set up this way because we are an outgoing call center, and the context that the batphones go to a database table to pull the phone number they are calling. Along with this new hardware, we changed to a new server (just a Dell E520 workstation with 4 gigs of RAM and 2 250 gig SATA drives software RAIDed) with the following software: Ubuntu 9.04 Asterisk 1.6.1.4 Asterisk-addons 1.6.1.1 (for the cdr-mysql plugin) dahdi-linux-complete 2.2.0.2 + 2.2.0 libpri 1.4.10.1 rhino drivers 0.99.2 Since day one, all batphones have had a weird noise at the very beginning of the call. I contacted Rhino about it and the support tech told me that it's fsk tones that have caller ID and MWI information and advised me to turn off advanced features like mailboxes. The phones already didn't have mailboxes, but I put in mwisendtype=nofsk in chan_dahdi.conf anyway, and all features like faxdetect and transfer are turned off. Has anyone else experienced this issue and fixed it? Thanks. Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1 Rochester, NY 14624 Office: 888-865-0065 x202 Mobile: 585-705-1400 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - log rotation
On 3/09/09 10:39 PM, Olivier wrote: Hi, It seems Asterisk needs to be notified that log rotation happened tough applications like astmanproxy or FOP doesn't need to be restarted (nor notified of any rotation). In logrotate we just add a command to be run after rotation to do: asterisk -rx 'logger reload' Is this personal observation true ? Nope :) As posted neither use the logs - although it may be possible that Asterisk has a problem if the file handle it has open ends up pointing at nothing. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate calls with AMI.
On 4/09/09 2:41 AM, Guillaume Yziquel wrote: Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: Action: originate Channel: SIP/zoiper Exten: yziquel Priority: 1 Timeout: 30 Context: internal To start with I'd do (just rearranging but makes me feel better): Action: originate Channel: SIP/zoiper Context: internal Exten: yziquel Priority: 1 Timeout: 30 Callerid: yziquel But also, are you sure that the extension yziquel exists in the internal context? type the following: dialplan show internal -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] probleme with web-meetme.3.1.0
On 4/09/09 3:24 AM, harry R wrote: Hi everybody I have a problem and want to know if anyone has already seen it before : I try to use web-meetme.3.1.0 and follow these instructions First off, (even though I don't understand French) your error is that ast_config_load in the version of Asterisk you're using is expecting two arguments but is being provided with one. I.E. Asterisk code has changed since that version of web-meetme was written. What version of Asterisk are you using? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate calls with AMI.
Hello. Matt Riddell a écrit : To start with I'd do (just rearranging but makes me feel better): Action: originate Channel: SIP/zoiper Context: internal Exten: yziquel Priority: 1 Timeout: 30 Callerid: yziquel Thank you for your answer. But also, are you sure that the extension yziquel exists in the internal context? Yes, it does. I finally got it right (no rearrangement) with Action: originate Channel: SIP/zoiper WaitTime: 30 CallerId: yziquel Exten: yziquel Context: internal Priority: 1 Somehow surprised that the only needed change was to change Timeout to Waitime... type the following: dialplan show internal Here it is: seldon*CLI dialplan show internal [ Context 'internal' created by 'pbx_config' ] '500' = 1. Verbose(1|Echo test application) [pbx_config] 2. Echo() [pbx_config] 3. Hangup() [pbx_config] 'yziquel' = 1. Verbose(1|Extension yziquel) [pbx_config] 2. Dial(SIP/yziquel|30) [pbx_config] 3. Hangup() [pbx_config] 'zoiper' = 1. Verbose(1|Extension zoiper)[pbx_config] 2. Dial(SIP/zoiper|30)[pbx_config] 3. Hangup() [pbx_config] -= 3 extensions (9 priorities) in 1 context. =- Thanks a lot. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] setvar=CDR(accountcode)=${EXTEN} in sip.conf ???
Trying to do something like this in the sip.conf under my incoming provider
profiles:
setvar=CDR(accountcode)=${EXTEN}
It seems to show up in the CDR but it's showing up exactly like this
${EXTEN}.
Is there a way to stuff the DNIS (number dialed) into the accountcode for
CDR?
I have already accomplished setting on a number by number basis, I just want
to do it globally for all number when they come in.
Thanks in advance.
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Re: [asterisk-users] setvar=CDR(accountcode)=${EXTEN} in sip.conf ???
Todd Routhier wrote:
Trying to do something like this in the sip.conf under my incoming
provider profiles:
setvar=CDR(accountcode)=${EXTEN}
Set(CDR(accountcode)=${EXTEN})
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] setvar=CDR(accountcode)=${EXTEN} in sip.conf ???
On 4/09/09 10:41 AM, Doug Lytle wrote:
Todd Routhier wrote:
Trying to do something like this in the sip.conf under my incoming
provider profiles:
setvar=CDR(accountcode)=${EXTEN}
Set(CDR(accountcode)=${EXTEN})
Nah he's trying to do it in sip.conf
Basically what you should do is add the line that Doug recommended to
the dialplan - sip.conf can't know what the extension is because it's
setting these variables once for a peer.
--
Cheers,
Matt Riddell
Director
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[asterisk-users] Asterisk 1.2.35, 1.4.26.2, 1.6.0.15, and 1.6.1.6 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.2.35, 1.4.26.2, 1.6.0.15, and 1.6.1.6. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to an IAX2 denial of service vulnerability. For more information about the details of this vulnerability, please read the security advisory AST-2009-006, which was released at the same time as this announcement. The announcement is available at http://downloads.asterisk.org/pub/security/AST-2009-006.pdf Also, please see the PDF in doc/IAX2-security.pdf in your Asterisk source. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.2.35 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.26.2 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.15 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.6 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AST-2009-006: IAX2 Call Number Resource Exhaustion
Asterisk Project Security Advisory - AST-2009-006 ++ | Product | Asterisk | |+---| | Summary | IAX2 Call Number Resource Exhaustion | |+---| | Nature of Advisory | Denial of Service | |+---| | Susceptibility | Remote unauthenticated sessions | |+---| | Severity | Major | |+---| | Exploits Known | Yes - Published by Blake Cornell blake AT | || remoteorigin DOT com on voip0day.com| |+---| |Reported On | June 22, 2008 | |+---| |Reported By | Noam Rathaus noamr AT beyondsecurity DOT com , | || with his SSD program, also by Blake Cornell | |+---| | Posted On | September 3, 2009 | |+---| | Last Updated On | September 3, 2009 | |+---| | Advisory Contact | Russell Bryant russell AT digium DOT com | |+---| | CVE Name | CVE-2009-2346 | ++ ++ | Description | The IAX2 protocol uses a call number to associate| | | messages with the call that they belong to. However, the | | | protocol defines the call number field in messages as a | | | fixed size 15 bit field. So, if all call numbers are in | | | use, no additional sessions can be handled. | | | | | | A call number gets created at the start of an IAX2 | | | message exchange. So, an attacker can send a large | | | number of messages and consume the call number space.| | | The attack is also possible using spoofed source IP | | | addresses as no handshake is required before a call | | | number is assigned. | ++ ++ | Resolution | Upgrade to a version of Asterisk listed in this document | || as containing the IAX2 protocol security enhancements. In | || addition to upgrading, administrators should consult the | || users guide section of the IAX2 Security document | || (IAX2-security.pdf), as well as the sample configuration | || file for chan_iax2 that have been distributed with those | || releases for assistance with new options that have been | || provided. | ++ ++ | Discussion | A lot of time was spent trying to come up with a way to | || resolve this issue in a way that was completely backwards | || compatible. However, the final resolution ended up| || requiring a modification to the IAX2 protocol. This | || modification is referred to as call token validation. | || Call token validation is used as a handshake before call | || numbers are assigned to IAX2 connections. | || | || Call token validation by itself does not resolve the | || issue. However, it does allow an IAX2 server to validate | || that the source of the messages has not been spoofed. In | ||
[asterisk-users] chan_mobile -- bluetooth
Hi, I'm having trouble with chan_mobile. mobile search is give me only headset device even if they are laptop or cellphone. I want to use my nokia 6630 with asterisk and I cannot understand what is missing. I already google that problem and read the forum.. but i'm not getting out of this problem. Please help me! magopieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF with duration = 0
Hi All, I'm receivig DTMF from my provider in RFC2833 but my provider send in Event Duration the value 0 and when asterisk forward this DTMF to PSTN asterisk play the DTMF very fast. Anybody now how fixes this problem ? Thank You, Bruno Rodrigues___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Disable CDR for callfile?
yes, callfile work through context. When control is in the dialplan context/extension/priority, I can enable/disable CDR's. Problem comes when asterisk dial a call and user is busy or did not answered the call. In this case a CDR is generated. No CDR should be generated on busy or failed call attemps? How I do it? CallFile: Channel: SIP/username CallerID: callback 100 MaxRetries: 3 RetryTime: 10 WaitTime: 40 Context: bridgecall Extension: 12129339037 Set:NoCDR Priority: 1 Account: 123; Thanks M. Faheem --- On Thu, 9/3/09, Danny Nicholas da...@debsinc.com wrote: From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] How to Disable CDR for callfile? To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Thursday, September 3, 2009, 6:01 PM Have your callfile work through a context instead of dialing. The context can disable CDR. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faheem Sent: Thursday, September 03, 2009 7:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Disable CDR for callfile? I want to do a callback scenario. Each time asterisk receive a call, it creates a callfile, sends back the hangup signal and dial back the extension. Here the default CDR logging is enabled. If a dial attempt is failed then a CDR is generated. How I do a trick to stop CDR logging for all callfiles, without changing the default behaviour of CDR logging. I know its NoCDR() function that will disable CDR() logging, But how it will be done in callfiles ? Thanks, M. Faheem -Inline Attachment Follows- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
