Re: [Asterisk-Users] ASTCC - INUSE Flag
Thanks. I have a question for the mailing list in general. Where should the card get marked as in use? Should it be as soon as you enter the number or should it be when it dials? I don't know for sure. Darren Wiebe [EMAIL PROTECTED] Michael K. Rodriguez wrote: This is my debug with the same issue The agi terminates during the sub tell_time() and exits without calling sub setinuse() or completing the reset of the script. AGI Tx agi_request: astcc.agi AGI Tx agi_channel: Zap/49-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1128401550.162 AGI Tx agi_callerid: xx AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 3 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 33 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: xx AGI Tx agi_rdnis: unknown AGI Tx agi_context: default AGI Tx agi_extension: xx AGI Tx agi_priority: 103 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: xxx AGI Tx 0-r1*CLI AGI Rx ANSWERLI AGI Tx 200 result=0 AGI Rx GET DATA astcc-enter-card-num 6000 -- Playing 'astcc-enter-card-num' (language 'en') AGI Tx 200 result=3546 AGI Rx STREAM FILE astcc-youhave 0123456789 AGI Tx 200 result=0 endpos=4480 AGI Rx SAY NUMBER 11 0123456789 -- Playing 'digits/11' (language 'en') AGI Tx 200 result=0 AGI Rx STREAM FILE astcc-dollars 0123456789 AGI Tx 200 result=0 endpos=6720 AGI Rx STREAM FILE astcc-and 0123456789 AGI Tx 200 result=0 endpos=3680 AGI Rx SAY NUMBER 88 0123456789 -- Playing 'digits/80' (language 'en') -- Channel 0/1, span 3 got hangup request AGI Tx 200 result=-1 == Spawn extension (default, x, 103) exited non-zero on 'Zap/49-1' -- Hungup 'Zap/49-1' -Michael On 10/3/05 10:52 PM, Darren Wiebe [EMAIL PROTECTED] wrote: Can you please post the output with debug agi on ? Darren Wiebe [EMAIL PROTECTED] Scott Wolfe wrote: I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in my astcc-exten.conf [incoming] exten = s,1,Answer ;exten = s,2,DeadAGI(astcc.agi) exten = s,2,AGI(astcc.agi) exten = s,3,Hangup I did some Google search on this issue and saw someone else had a problem but no response. -Scott ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Easy SIP.conf questien. Incomming call context?
Does the incomming call context in extensions.conf always have to be [default]? Can't i define different context for incomming like i can for Outgoing in the sip.conf? My default conf is getting very large. Regards, Arne morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intel Pentium Celeron
Hi all, im going to install asterisk with a 4 BRI (HFC chipset) on a Celeron at 2.6 GHz I dont known Celeron performance, but i listen that is not very good. Could I have some performance isuue with this kind of processor ? Thanks for all Giordano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Easy SIP.conf questien. Incomming call context?
Arne Morten Johansen wrote: Does the incomming call context in extensions.conf always have to be [default]? Can't i define different context for incomming like i can for Outgoing in the sip.conf? My default conf is getting very large. In sip.conf, you can set context in a few places: In the [general] section you set the context for unknown users, un-authenticated calls. If you do not set it at all, it will default to [default]. In a type=user, you can set the context for incoming calls that this device places. In a type=peer, you can set the context for incoming calls for this peer. In the cvs head and 1.2 beta you can also set a subscribecontext to limit what extensions a peer can subscribe to the status of. Normally, you want a limited set of extensions in the context you point to in the [general] section and more services for users/peers. Users and peers normally get outbound calling, which should not be allowed from outside users. Also do remember that you can include contexts within context with the include= statement. Please read the sample configuration files that we provide with Asterisk to learn more. There are many examples on the web and on the wiki as well, so you have some reading to do before you continue exploring Asterisk dialplans! Good luck! /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to enter digits using sjphone
Hi all, A small question relating sjphone Here it is I am connecting from pc to Asterisk using Sjphone. I can make outgoing calls according to dial plan setup, but I am not able to enter options asked during the call like enetering passwords for voicemail. SJ phone initiates just another call for it like [EMAIL PROTECTED] I tried with many other options but definetley lacking the key :( Also if any body can tell me about other good softphones for linux I know abt kphone, Iaxcomm, twinkle, linphone, sjphone regards /Gurmi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote call pick-up
...or test the PickUpChan command coming with the bristuff-patch from zapata Damian Funnell wrote: Hi, Does anyone have remote call pick-up working on * (either via SIP or otherwise)? If so then can you post your features.conf, sip.conf and/or zapata.conf? We can't seem to get this (seemingly simple) function to work. Check callgroups and pickupgroups in the channel configuration files. There are sample configurations in the sample configs. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatic callback feature *66
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+reverse+hold ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how can i let the user in 1th Asterisk can call the user in 2nd Asterisk?
Hi list, I set up two asterisk servers , 1001 is the first asterisk servers sip user, and 2001 is the second asterisk servers sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone give me some ideas about how to write this configuration in asterisk config files and which conf file should i use? Thanks, Erdem HAKI ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-users] Configuration QuadBRI Junghanns
What I can configuration my card Junghanns QuadBri? Where I can download drivers? Thanks? ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call transfer problem - something strange
Hi, I try to set up planet VIP-050 with asterisk. Everything works fine instead of the call transfer. When I press # console says something like this: Oct 5 11:11:20 DEBUG[25104]: chan_sip.c: sip_rtp_read: Oooh, format changed to 1024 Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 DEBUG[25104]: rtp.c:1193 ast_rtp_write: Ooh, format changed from ulaw to ilbc Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multipleof 50 bytes long from RTP (4)? Oct 5 11:11:20 WARNING[25104]:
Re: [Asterisk-Users] how can i let the user in 1th Asterisk can call the user in 2nd Asterisk?
hi,
I advice u to use IAX to connect 2 asterisk more better and less bandwitdh. with following config u can do. just re do for sip.
(serverA)iax.conf[general]register = username:password@serverB hostname or IP[serverB]type=frienduser=usernamesecret=password
host=serverB hostname or IPextensions.confexten = _7XXX,1,Dial(IAX2/serverB/${EXTEN:1},30,r)exten = _7XXX,2,Congestion(serverB)iax.conf[serverA]
type=frienduser=usernamesecret=passwordhost=dynamic | serverA hostname or IPextensions.conf exten = _8XXX,1,Dial(IAX2/serverA/${EXTEN:1},30,r)exten = _8XXX,2,Congestion
On 10/5/05, Erdem HAKİ [EMAIL PROTECTED] wrote:
Hi list,
I set up two asterisk servers , 1001 is the first asterisk server's sip user, and 2001 is the second asterisk server's sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk.
First asterisk server ip :
192.168.3.101
Second asterisk server ip :
192.168.3.102 can someone give me some ideas about how to write this configuration in asterisk config files and which conf file should i use?
Thanks,
Erdem HAKI
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[Asterisk-Users] can't run app_txfax
Anyone run app_txfax and app_rxfax successfully on asterisk 1.2 beta1? I get an error when trying to run asterisk: [app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_info Oct 5 12:05:24 WARNING[14665]: loader.c:543 load_modules: Loading module app_txfax.so failed! Ouch ... error while writing audio data: : Broken pipe What could be the problem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-users] Configuration QuadBRI Junghanns
Hi, Junghanns drivers can be found on Junghanns site, they are named something like BRIStuff. Giorgio Fabio Montemaggiore wrote: What I can configuration my card Junghanns QuadBri? Where I can download drivers? Thanks? ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM versions question
this was from Ian at digium: I have talked to some hardware guys around here, and they said that the rev H cards might still show up as rev E/F. The bottom line is that you are having problems with it though, so we need to take a look at it to see if there are any software-related problems first, and if it just seems to be bad hardware, we'll RMA it for you asap. Best Regards Greg Cirino Spam and Virus Free Email included with every email account Cirelle Enterprises Inc. 25 Indian Rock Rd #421 Windham NH, 03087 603-425-2221 canuck15 wrote: So what did you find out? I also have a TDM400P board labelled REV H that reports REV E/F. -Original Message- From: Cirelle Enterprises [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 04, 2005 7:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM versions question Kevin P. Fleming wrote: Cirelle Enterprises wrote: I have just realized while trying to research asterisk not acking incoming calls that the tdm400b card is stamped rev H, but when I issue the zap show status command in the manager interface, it indicates Wildcard TDM400P REV E/F Board 1 Please contact Digium Technical Support. I don't believe you should be seeing that combination. ___ Just did, thanks Best Regards Greg Cirino Spam and Virus Free Email included with every email account Cirelle Enterprises Inc. 25 Indian Rock Rd #421 Windham NH, 03087 603-425-2221 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
trixter http://www.0xdecafbad.com wrote: Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. Unless of course they don't live in the United Sue'ers of America. :D -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel TDM questions
Yes, we have an applications that needs to detect the actual answer of the call not when it is ringing. CCF -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Angus ComberSent: Friday, September 30, 2005 19:18To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Zaptel TDM questions I think the Asterisk must answer the call to be able to then dial out on the second port. This is what happens on any other PBX I have worked with in this sort of scenario. Is this a problem for you? Angus - Original Message - From: Chee Foong To: asterisk-users@lists.digium.com Sent: Friday, September 30, 2005 10:20 AM Subject: [Asterisk-Users] Zaptel TDM questions Hello, I have a TDM04B. I make a call into the first port of the card. Once asterisk receive the call, it will make another call out using the second port. From what i have observerd as soon as the called party on the second port starts ringing asterisk show the following : -- Zap/2-1 answered Zap/1-1 Any idea why asterisk thinks the call has been answered while actually the phone is still ringing? Anybody know how to avoid asterisk to answer the call while ringing? Also, I have no Answer or any Playbackcommand in the dial plan before making a call out of second port. I have also try setting callprogress to yes/no but the results are the same. Thanks CCF ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel tone description
Hi, Were trying to use TDM04B with a few analog switches and weve noticed that it works with the tones from USA only. As its documented saying that those tones are hard-coded in the source for analog cards. Wed like to know if theres anyone who could tell us under which file these settings would be hard-coded as we could like to do some experiments which will benefit to all Zaptel / Asterisk users in Asian part of the world. Would appreciate an early reply. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco AS5300 -- [SIP] -- Asterisk - NO AUDIO
OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xx Password: 1000xx Server: br.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk server and my asterisk server sends back the dial tone, no problem. The problem is when I pick up the phone, no audio. If I change the dial plan to do a Playback instead of Dial an extension I can see in the console it answers the call and starts to play the Playback but no audio. I can connect direclty to the Cisco AS5300 with sjphone or a budgetone 102 with no problem and get dial tone and full audio both ways but when I use the asterisk no audio. I have tried every codec possible. I have installed g729, g723 with no luck. I have tested both these codecs by forcing my budgetone to use with no problem so I know the codecs work. So the problem is when I ask asterisk to register to the Cisco AS5300 as a SIP Client it does everything right except pass the audio. There is no firewall configured. I know the Cisco SIP Server works because it works with the softphone SJPHONE and directly with the Budgetone 102. I have reinstalled Asterisk so many times. I have reinstalled g729 g723 so many times. The SIP debug output is pasted below. I have been struggling with this for weeks with no luck. Any help would be appreciated. Steven Ducat. * -- SIP read from 203.88.192.42:5160: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0 Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786 From: sip:[EMAIL PROTECTED];tag=1CA65AC-9C8 To: sip:[EMAIL PROTECTED] Date: Thu, 29 Sep 2005 20:14:40 GMT Call-ID: [EMAIL PROTECTED] Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2153363387-811340250-2169109749-53752559 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 5 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1128024880 Contact: sip:[EMAIL PROTECTED]:57786 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 432 P-hint: Proxied P-hint: usrloc applied v=0 o=CiscoSystemsSIP-GW-UserAgent 5786 3481 IN IP4 211.147.240.237 s=SIP Call c=IN IP4 211.147.240.237 t=0 0 m=audio 37708 RTP/AVP 18 4 3 8 0 110 c=IN IP4 203.88.192.42 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:110 X-NSE/8000 a=fmtp:110 192-194 a=direction:passive a=direction:active a=nortpproxy:yes --- (24 headers 19 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 203.88.192.42 : 5160 (non-NAT) Found no matching peer or user for '203.88.192.42:5160' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 110 Peer audio RTP is at port 211.147.240.237:37708 Found description format G729 Found description format G723 Found description format GSM Found description format PCMA Found description format PCMU Found description format X-NSE Capabilities: us - 0x100 (g729), peer - audio=0x30f (g723|gsm|ulaw|alaw|g729|speex)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 84104214 in default (domain 70.84.200.204) list_route: hop: sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on Transmitting (no NAT) to 203.88.192.42:5160: SIP/2.0 100 Trying Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0;received=203.88.192.42 Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786 From: sip:[EMAIL PROTECTED];tag=1CA65AC-9C8 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Dial(SIP/211.147.240.237-b7116c10, Local/2001/n) in new stack -- Executing Macro(Local/[EMAIL PROTECTED],2, oneline|SIP/stevenducat) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/stevenducat|20) in new stack -- Called 2001/n We're at 70.84.200.204 port 14922 Answering/Requesting with root capability 0x100 (g729) 12 headers, 8 lines Reliably Transmitting (NAT) to 83.146.11.93:60073: INVITE sip:[EMAIL PROTECTED]:18234 SIP/2.0 Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK502b5287;rport From: 0017911 sip:[EMAIL PROTECTED];tag=as2c8caf36 To: sip:[EMAIL PROTECTED]:18234 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL
[Asterisk-Users] agi-test.agi question - wierd results
Hello I am starting to learn AGI. I have setup an extension to play the agi-test.agi perl script and the output I get is this on console: On Polycom 300: -- Executing Answer(SIP/200-72d2, ) in new stack -- Executing AGI(SIP/200-72d2, agi-test.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi -- Playing 'digits/1' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/90' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/million' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/30' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/thousand' (language 'en') -- Playing 'digits/4' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/60' (language 'en') -- Playing 'digits/5' (language 'en') On other handsets: -- Executing Answer(SIP/201-4415, ) in new stack -- Executing AGI(SIP/201-4415, agi-test.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi -- Playing 'digits/1' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/90' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/million' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/30' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/thousand' (language 'en') -- Playing 'digits/4' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/60' (language 'en') -- Playing 'digits/5' (language 'en') -- AGI Script agi-test.agi completed, returning 0 -- Executing Hangup(SIP/201-4415, ) in new stack == Spawn extension (default, 290, 3) exited non-zero on 'SIP/201-4415' I don't get the other stuff - eg the send file, send text, etc. I have an Asterisk console open (used asterisk -r) on a putty session on a PC connected over network. There is no other asterisk console open. Also when I dial on a a Snom 190 or a Sipura-841 I hear all the digits as above correctly. But on a Polycom 300 I get to the digit 30 and it then seems to stop playing the digits. But they of course appear on the console. Why am I not getting the send file stuff etc on the console? The Polycom bit I expect is some setting on the phone I need to troubleshoot. But not getting all the expected output from the agi script seems strange. Is there possibly some problem with my environment? My handset? I am running on Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?
I have the same problem, after about a month the card doesn't report anyincoming calls anymore to the console. Don't know the rev of my card yet, unloading asterisk and unloading the modules and then restartingeverything does seem to help though, no need to reboot.___ Same exact problem here. Problem starts after about 3 weeks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel TDM questions
Could you not just ignore the first answer and watch out for the answer when the remote end picks up? Angus - Original Message - From: Chee Foong To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, October 05, 2005 11:35 AM Subject: RE: [Asterisk-Users] Zaptel TDM questions Yes, we have an applications that needs to detect the actual answer of the call not when it is ringing. CCF -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Angus ComberSent: Friday, September 30, 2005 19:18To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Zaptel TDM questions I think the Asterisk must answer the call to be able to then dial out on the second port. This is what happens on any other PBX I have worked with in this sort of scenario. Is this a problem for you? Angus - Original Message - From: Chee Foong To: asterisk-users@lists.digium.com Sent: Friday, September 30, 2005 10:20 AM Subject: [Asterisk-Users] Zaptel TDM questions Hello, I have a TDM04B. I make a call into the first port of the card. Once asterisk receive the call, it will make another call out using the second port. From what i have observerd as soon as the called party on the second port starts ringing asterisk show the following : -- Zap/2-1 answered Zap/1-1 Any idea why asterisk thinks the call has been answered while actually the phone is still ringing? Anybody know how to avoid asterisk to answer the call while ringing? Also, I have no Answer or any Playbackcommand in the dial plan before making a call out of second port. I have also try setting callprogress to yes/no but the results are the same. Thanks CCF ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configure Mitel SX-2000 Lite to provide disconnect supervision for Asterisk
Where I am situated, our telephone lines are actually extensions (analogue, 2-wire) from a group of Mitel SX-2000 LITE PBXs I have 2 extensions, 1 of which I have connected to a TDM11B for incoming and outgoing calls using Asterisk. Is it possible to configure an SX-2000 LITE to provide disconnect supervision? I would be really very grateful if someone could let me know if it is possible, AND, if it is, how to do it! TIA Leigh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unwieldy outbound macro
I have the following pair of macros defined to handle outbound calls from *.
Rather than specifying full dialstrings in the main body of extensions.conf,
outbound dial commands are made using a macro call as follows:
Macro
(outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate
way4)
The final gateway defined is nearly always a fallback to PSTN if none of the
IAX or SIP gateways are working.
The problem is, its an incredibly unwieldy macro that's horrible to edit if
I want to add a new gateway to the thing. Looking through the documentation
for macro programming, I couldn't find the equivalent of a 'shift' statement
you'd find in other programming languages.
Can a kind soul suggest how I might tidy it up?
Thanks in advance.
[macro-outbound]
; ${ARG1} Number
; ${ARG2} Caller ID
; ${ARG3,4,5,6} Outgoing gateways in order of use
exten = s,1,GotoIf($[${ARG2} = ]?4)
exten = s,2,SetCIDNum(${ARG2})
exten = s,3,Goto(5)
exten = s,4,SetCIDNum(${DEFAULTCID})
exten = s,5,SetVar(GATEWAY=${ARG3})
exten = s,6,SetVar(ARG3=${ARG4})
exten = s,7,SetVar(ARG4=${ARG5})
exten = s,8,SetVar(ARG5=${ARG6})
exten = s,9,GotoIf($[${GATEWAY} = voip1]?14)
exten = s,10,GotoIf($[${GATEWAY} = voip2]?18)
exten = s,11,GotoIf($[${GATEWAY} = voip3]?16)
exten = s,12,Macro(dialout,SIP/[EMAIL PROTECTED])
exten = s,13,GotoIf($[${ARG3} = ]?20:5)
exten = s,14,Macro(dialout,IAX2/voip1/${ARG1})
exten = s,15,GotoIf($[${ARG3} = ]?20:5)
exten = s,16,Macro(dialout,IAX2/voip2/${ARG1})
exten = s,17,GotoIf($[${ARG3} = ]?20:5)
exten = s,18,Macro(dialout,SIP/[EMAIL PROTECTED])
exten = s,19,GotoIf($[${ARG3} = ]?20:5)
exten = s,20,Playtones(congestion)
exten = s,21,Congestion()
[macro-dialout]
; ${ARG1} Dialstring
exten = s,1,Dial(${ARG1},,W)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,Playtones(busy)
exten = s-BUSY,2,Busy()
exten = _s-.,1,NoOp(${DIALSTATUS})
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
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Re: [Asterisk-Users] Polycom config and DTMF problems
On Tuesday 04 October 2005 18:04, Anthony Rodgers wrote: I found the best reference to be the SoundPoint IP / SoundStation IP Admin Guide - SIP 1.5 from the Polycom web site - http://www.polycom.com/common/pw_item_show_doc/1,1276,4349,00.pdf. You're right - that admin guide is much more useful that I had initially thought - thanks! Not sure about the DTMF issue - I used the config files at http://www.krisk.org/asterisk/pcom/, if that helps Yeah, I have no idea either. I'm going to try to capture the RTP stream and see if it's being sent inband, but I clearly have my sip.cfg file set to rfc2833: DTMF tone.dtmf.level=-15 tone.dtmf.onTime=50 tone.dtmf.offTime=50 tone.dtmf.chassis.masking=0 tone.dtmf.stim.pac.offHookOnly=0 tone.dtmf.viaRtp=1 tone.dtmf.rfc2833Control=1 tone.dtmf.rfc2833Payload=101 / And I've already tried dtmfmode=inband in my asterisk sip.conf, so I'm not sure what's going on. -Doug -- Douglas E. Warner[EMAIL PROTECTED] Network Engineer CTI Networks, Inc. http://www.ctinetworks.com+1 717 975 9000 pgpSma0yCYIDV.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] From Database, PHP-Webinterface - TO flatfileconfiguration
Hi. I've started working on a PHP-project that generates the configuration files i need based on what's in my MYSQL database. I can add, delete and edit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up a java servlet so that i can get asterisk to reload by pushing a button from the web-interface. The php-scripts communicates with ip-sockets. So what's my question? I'm just wondering if this is a good idea. Any comments? I've looked into the mysql support in the addons but I find it hard to do and complicated. For me it's easier to write the config-files from a php-script. But what about performance? Any big difference here? What do you think is the pros and cons of a setup like this? Regards, Arne Morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE411P and TE406P stability
I am getting ready to purchase my first Digium card to start experimenting with Asterisk. Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ). I will be using Asterisk @ Home, so will be Asterisk v1.0.9. I took a quick poke at the lists, and it appears several people have been having issues. Am I better served with a TE410P until 1.2 becomes stable and released? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADSI -- is it dead? Worth bothering with?
Stephen Bosch wrote: Since Colin Anderson -- in a previous thread -- asked the question about whether ADSI was dead, I thought it was worth discussing. Does anybody else have anything to add? -Stephen- I hope ADSI is not dead! We have 100 Aastra 390 ADSI phones with 20 of them in service. We have locations where the infrastructure will not support packet based phones, even if we wanted to use them, yet we would like to give people some menus to lesson the learning curve. Having a menu that changes based on context is great. I really like the voice mail menu where you do not need to remember any of the keys to press but just read the item and press the button by your selection. The only problem is the length of time it takes to load the menu. I believe there is really no need to load the stuff every time voicemail is dialed. The local phone company offers ADSI service. They have a number to dial to program the ADSI menus. This number only needs to be dialed once. After that things are pretty snappy. I will soon start work on converting the rest of our company to primarily ADSI phones. Don Pobanz ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call-in/Call-out
snip
written by Crystal Stream, Incorporated
Here is my extensions.conf file. Things have been left
out or changed to protect the innocent.
Why isn't it working when I call from the outside that
when I press 124 it repeats the menu and doesn't
initiate DISA correctly to dial out?
[general]
static=yes
writeprotect=yes
[globals]
[voicepulse-in]
exten = ${OURVOIP1},1,Noop(${DATETIME} ${CALLERID})
exten = ${OURVOIP1},2,Answer
exten = ${OURVOIP1},3,Goto(main-menu,s,2)
exten = ${OURVOIP1},4,Hangup
[nufone-in]
exten = ${OURVOIP3},1,Noop(${DATETIME} ${CALLERID})
exten = ${OURVOIP3},2,Answer
exten = ${OURVOIP3},3,Goto(main-menu,s,2)
exten = ${OURVOIP3},4,Hangup
[incoming-sip]
include = voicepulse-in
include = nufone-in
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,Answer
exten = s,3,Goto(main-menu,s,6)
exten = s,4,Hangup
MAIN MENU ;;
[main-menu]
include = operator
include = queues
; if pressed 4-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for main menu selections:
exten = 1,1,Goto(office-day,1,1)
exten = 2,1,Goto(office-day,2,1)
exten = 3,1,Goto(office-day,3,1)
exten = 4,1,Goto(office-day,4,1)
exten = 5,1,Goto(office-day,5,1)
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,Wait(${WAIT_AFTER_ANSWER})
exten = s,3,SetCallerID(${CALLERID})
exten = s,4,DigitTimeout,2
exten = s,5,ResponseTimeout,7
exten =
s,6,Background(/usr/local/etc/asterisk/ivr/UPGRADEPHONES)
exten =
s,7,Background(/usr/local/etc/asterisk/ivr/GREETING)
exten = s,8,WaitExten(1.2)
exten = s,9,SetGlobalVar(prompt_loops=0)
exten =
s,10,GotoIfTime(07:00-18:00|mon-thu|*|*?office-day,s,2)
exten =
s,11,GotoIfTime(10:00-16:30|fri|*|*?office-day,s,2)
exten = s,12,Goto(office-night,s,1)
exten = t,1,Goto(main-menu,#,1) ; If they
take too long, go to hangup
; invalid
exten = i,1,Wait(1)
exten = i,2,Playback(invalid) ; That's not valid,
try again
exten = i,3,Wait(1)
exten = i,4,Goto(s,6)
; #=hangup
exten = #,1,Wait(1)
exten = #,2,Playback(vm-goodbye)
exten = #,3,Wait(2)
exten = #,4,Hangup
[office-day]
include = operator
include = queues
; if pressed 3-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for accessing voicemail:
include = voicemail
exten = s,1,SetGlobalVar(prompt_loops=1)
exten = s,2,WaitExten(${BETWEEN_PROMPTS})
exten =
s,3,Background(/usr/local/etc/asterisk/ivr/MAINMENU)
exten = s,4,WaitExten(4)
exten =
s,5,SetGlobalVar(prompt_loops=$[${prompt_loops} + 1])
exten = s,6,GotoIf($[${prompt_loops}
${MAX_MENU_LOOPS}] ? 2:23)
exten = s,7,Goto(operator,0,1)
; invalid
exten = i,1,Playback(invalid) ; That's not valid,
try again
exten = i,2,Wait(1)
exten = i,3,Goto(s,7)
; timeout
exten = t,1,Goto(operator,0,1)
[office-night]
include = operator
include = queues
; if pressed 3-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for accessing voicemail:
include = voicemail
exten = s,1,SetGlobalVar(prompt_loops=1)
exten = s,2,WaitExten(${BETWEEN_PROMPTS})
exten =
s,3,Background(/usr/local/etc/asterisk/ivr/MAINMENU)
exten = s,4,WaitExten(4)
exten =
s,5,SetGlobalVar(prompt_loops=$[${prompt_loops} + 1])
exten = s,6,GotoIf($[${prompt_loops}
${MAX_MENU_LOOPS}] ? 2:23)
exten = s,7,Goto(operator,0,1)
; invalid
exten = i,1,Wait(1)
exten = i,2,Playback(invalid) ; That's not valid,
try again
exten = i,3,Wait(1)
exten = i,4,Goto(s,4)
; timeout
exten = t,1,Goto(main-menu,#,1) ; If they
take too long, go to hangup
[local]
; Directory:
exten = 411,1,Directory(crystal-sip|local)
exten = 411,2,Hangup
; DISA
exten = 124,1,Answer
exten = 124,2,DigitTimeout(5)
exten = 124,3,ResponseTimeout(10)
exten = 124,4,Authenticate(16435679)
exten = 124,DISA(4376194673164379|crystal-sip)
and snip
---
Why did you not set a priority here or is it a typo?
Erik
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RE: [Asterisk-Users] Intel Pentium Celeron
Try it out and let us know! J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: Wednesday, October 05, 2005 3:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Intel Pentium Celeron Hi all, im going to install asterisk with a 4 BRI (HFC chipset) on a Celeron at 2.6 GHz I dont known Celeron performance, but i listen that is not very good. Could I have some performance isuue with this kind of processor ? Thanks for all Giordano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] From Database, PHP-Webinterface - TO flatfileconfiguration
Hey Arne! My project 'PhoneCALL' http://www.vecsector.com/phonecall does pretty much the same thing as you are describing - stores the configs in mysql then submits the changes to flat files reloads asterisk on completion. For me my clients - there hasn't been any noticeable difference, as we used mysql/php for CDR anyway. Also - the flat-files are loaded in memory once Asterisk starts, so it's not like it's constantly hammering the mysql database for info. Go for it! :-) --Dustin Arne Morten Johansen wrote: Hi. I've started working on a PHP-project that generates the configuration files i need based on what's in my MYSQL database. I can add, delete and edit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up a java servlet so that i can get asterisk to reload by pushing a button from the web-interface. The php-scripts communicates with ip-sockets. So what's my question? I'm just wondering if this is a good idea. Any comments? I've looked into the mysql support in the addons but I find it hard to do and complicated. For me it's easier to write the config-files from a php-script. But what about performance? Any big difference here? What do you think is the pros and cons of a setup like this? Regards, Arne Morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inter Asterisk trunking IAX /IAX2
Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks, Geo ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?
Brent, what version of asterisk are you using? Best Regards Greg Cirino Spam and Virus Free Email included with every email account Cirelle Enterprises Inc. 25 Indian Rock Rd #421 Windham NH, 03087 603-425-2221 Brent Franks wrote: I have the same problem, after about a month the card doesn't report any incoming calls anymore to the console. Don't know the rev of my card yet, unloading asterisk and unloading the modules and then restarting everything does seem to help though, no need to reboot. ___ Same exact problem here. Problem starts after about 3 weeks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer directly to voicemail (blind transfer)?
Doug wrote:
Hi,
Have looked around for info about this:
http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail
If we are using 5 digit extensions (10102: 10 for the company,
102 for the extension), where can we put something
so that 102* goes straight to voicemail without
waiting while the extension rings?
Here is what we have in extensions_additional.conf:
exten = 100,1,Goto(ext-local,10100,1)
exten = 101,1,Goto(ext-local,10101,1)
exten = 102,1,Goto(ext-local,10102,1)
exten = 103,1,Goto(ext-local,10103,1)
Would something like this in extensions.conf work?
exten = _XXX*,1,Voicemail(u${EXTEN:1})
I don't believe this is the correct syntax. It should be:
exten = _XXX*,1,Voicemail(u${EXTEN:0:3})
http://www.voip-info.org/wiki/view/Asterisk+variables
Another method would be to prefix with a digit instead of suffix with an
*. For us, all of our extensions are three digits and begin with a 5
or a 6 (5xx or 6xx). To transfer to voice mail we stick an eight in
front of the extension (85xx or 86xx). It works well for us.
Don Pobanz
___
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RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote: Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge. Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. Marvellous. Another company with a monopoly over aspects of VoIP technology. I don't have the millions required to mount a defence in a North American court, so I should just consider myself lucky that I live in a free country. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Asterisk/OH323
I have configured my asterisk to connect to an H323 gateway in order to place calls to the PSTN. The calls go through with no problem, but what I experience is a loss of received sound after about 5 mins in the call (the sound comes in very intermittent), while the other party continues to receive the call with no problem (it's a one-way loss). When this happens, I can see that the pc (using top) CPU utilization goes upto 75% and the computer becomes sluggish. I have tried this on another server using a P4 3GHz with 2GB of RAM, but the problem exists even on that server. Has anyone experienced this, or knows where the problem is? SIP to SIP calls or SIP to IAX does not give such a problem ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DPH-140S SIP Phone - SOLVED!
-Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Enviado el: Martes, 04 de Octubre de 2005 01:31 p.m. Para: Asterisk-Users@lists.digium.com Asunto: [Asterisk-Users] DPH-140S SIP Phone oddities Hi, list! I'm playing on an [EMAIL PROTECTED] installation, since a month or two. I've had no trouble setting it up 'n running. I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk. From this phones, I can make receive calls with no trouble, but, when I try to use some interactive function (eg Directory or Voicemail), the phone seems unable to transmit the digits to Asterisk. With the same config, but with a softphone (X-Lite), the digits are transmitted with no trouble at all. Please, do anyone have any clue? Thanks in advance. Juan. Just in case some other guy get crazy like me! ;) I solved it forcing phones' codec to ulaw, and setting DTMF to inband in sip_additional.conf. Regards, and thanks to all. Juan -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.10/119 - Release Date: 04/10/2005 attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel Pentium Celeron
Giordano Grandis [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) i'm going to install asterisk with a 4 BRI (HFC chipset) on a Celeron at 2.6 GHz I dont known Celeron performance, but i listen that is not very good. Could I have some performance isuue with this kind of processor ? You could have performance issues with any processor; It all depends upon what you want to do with it. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
Geo [EMAIL PROTECTED] wrote: Anyone using inter Asterisk trunking IAX /IAX2 ? No - you're the first to think of that. Congratulations. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel TDM questions
I'll jump in here with a couple of comments... What you're trying to deal with is detecting answer supervision for the outbound call, and not all telco's provide that. Since there is a wide variation (in and out of the US), asterisk does not try to detect when a call has been answered. It considers that outbound call answered as sson as it is done dialing. What does your telco provide to you on analog pstn lines for answer supervision? There is a high probability your telco isn't providing anything. In some cases, the telco can turn options for this but you still need to know what they are going to send to you. Could be a tone or simple reversal of tip/ring voltage. Yes, we have an applications that needs to detect the actual answer of the call not when it is ringing. CCF I think the Asterisk must answer the call to be able to then dial out on the second port. This is what happens on any other PBX I have worked with in this sort of scenario. Is this a problem for you? Angus Hello, I have a TDM04B. I make a call into the first port of the card. Once asterisk receive the call, it will make another call out using the second port. From what i have observerd as soon as the called party on the second port starts ringing asterisk show the following : -- Zap/2-1 answered Zap/1-1 Any idea why asterisk thinks the call has been answered while actually the phone is still ringing? Anybody know how to avoid asterisk to answer the call while ringing? Also, I have no Answer or any Playback command in the dial plan before making a call out of second port. I have also try setting callprogress to yes/no but the results are the same. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
Depending upon the patents in question, a few companies (Cisco comes to mind) may have prior art here. I know that a company cisco bought was doing VoIP in 1998, but no indications of which patents this is, or when they were filed. Paul trixter http://www.0xdecafbad.com wrote: Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge. Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. http://kansascity.bizjournals.com/kansascity/stories/2005/10/03/daily23.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't run app_txfax
On Wednesday 05 October 2005 12:31, Roman wrote: Anyone run app_txfax and app_rxfax successfully on asterisk 1.2 beta1? I get an error when trying to run asterisk: [app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_info Oct 5 12:05:24 WARNING[14665]: loader.c:543 load_modules: Loading module app_txfax.so failed! Ouch ... error while writing audio data: : Broken pipe What could be the problem? the same problem with current cvs HEAD I'm using spandsp 0.0.2pre20 Anyone? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco AS5300 -- [SIP] -- Asterisk - NO AUDIO
OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xx Password: 1000xx Server: br.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk server and my asterisk server sends back the dial tone, no problem. The problem is when I pick up the phone, no audio. Try inserting canreinvite=no in the sip.conf definition for the phone and restart asterisk. The trace suggests that your provider and the phone were told to establish a sessions between themselves, and that is not happening correctly. There is nothing in that trace that would suggest a codec problem, so I'm not sure how you jumped to that conclusion. In fact, the trace tells you there are several compatible codecs available between asterisk and your provider, and it chose g729 successfully. If that doesn't help, then copy/paste the important sections of sip.conf and extensions.conf that would reflect the handling of a call, and post that to this list. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compiling astrisk
I am trying to compile the astrisk-1.0.9 tarball on a RedHat 9 linux box with dev environment. I get a lot of the following as a result of a make /usr/bin/ld /usr/lib/crtn.o: invalid string offset 10 for section `.shstrtab' and final show stopper ./gentone busy 480 620 make[1]:***[busy.h] segmentation fault What do I need to fix in order to get a clean make? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P and TE406P stability
Do you specifically need hardware echo cancellation? If not you may want to try the TE405P/TE410P first and if you find that you need the hardware echo-canceller you can upgrade the card for $900(for some strange reason actually saves $100 over the list price of the TE411P) TE405P = $1495.00 quoted upgrade price for TE405P to TE406P = $900 upgraded TE405P to TE406P cost = $2395 TE406P = $2495.00 So it may save you money anyway if you go the upgrade path. As for a version, 1.2beta 1 is very stable, we've been using it on two high-volume production servers for over a month now with only one crash in that time. I would recommend Asterisk 1.2beta1 Hope this helps, MATT--- On 10/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I am getting ready to purchase my first Digium card to start experimenting with Asterisk.Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ).I will be using Asterisk @ Home, so will be Asterisk v1.0.9.I took a quick poke at the lists, and it appears several people have been having issues.Am I better served with a TE410P until 1.2 becomes stable and released?___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail not updating password SQL
I solved this problem by externpass to run a perl script that updates my database. Although I do believe that this should work through the built in ODBC setup. Ryan Hulsker On Tue, 2005-10-04 at 17:16, Ryan Hulsker wrote: I am using asterisk-1.2.0-beta1 with ODBC connecting to a MySQL database. All voicemail functions work correctly, except for updating passwords. When I try to change my password the system tells me it has changed, but it has not. My mysql.log shows no attempts to update the voicemail_users table in my database. Does anyone have any clues as to what is going on here? I have searched the wiki several times and can't find any similar issues. Thanks. Ryan Hulsker ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] From Database, PHP-Webinterface - TO flatfileconfiguration
Dear Arne Morten For me the best solution is to use MySQL. That is the reason we have developed http://astbill.com. There is no performance issues. AstBill is Open Source and for each SIP and IAX account you can choose if you want to use Static confguration from config-files or Asterisk REALTIME from the MySQL database. AstBill is a Web Based Billing, Routing and Management Software for Asterisk and MySQL based on Drupal. AstBill Provides pre and post Paid Billing Services. AstBill completely automates Asterisk billing and configuration from start to finish. Key benefits is the Central Web-based installation, Credit Control on outgoing calls and the User Management and call routing module. The system is fully themes based. Are Casilla Viking Diversified Ltd, London, United Kingdom http://astartelecom.com - Independent VOIP Telecoms Broker. Consulting in Asterisk and Drupal http://astbill.com - Billing, Routing and Management software for Asterisk and MySQL. Running on Drupal AstBill DEMO: http://demo.astbill.com On 10/5/05, Arne Morten Johansen [EMAIL PROTECTED] wrote: Hi.I've started working on a PHP-project that generates the configurationfiles i need based on what's in my MYSQL database. I can add, delete andedit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up ajava servlet so that i can get asterisk to reload by pushing a buttonfrom the web-interface. The php-scripts communicates with ip-sockets. So what's my question? I'm just wondering if this is a good idea. Anycomments? I've looked into the mysql support in the addons but I find ithard to do and complicated. For me it's easier to write the config-files from a php-script. But what about performance? Any big difference here?What do you think is the pros and cons of a setup like this?Regards,Arne Morten___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel TDM questions
Rich Adamson wrote: I'll jump in here with a couple of comments... What you're trying to deal with is detecting answer supervision for the outbound call, and not all telco's provide that. Since there is a wide variation (in and out of the US), asterisk does not try to detect when a call has been answered. It considers that outbound call answered as sson as it is done dialing. Perhaps though, a configurable option to set that to xx seconds would be useful, much as many LD resellers used to do for billing. If the call was up for longer than, lets say, 30 seconds, it was assumed to have been answered and billing would start. John Novack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P and TE406P stability
Would you consider a crash a month "very stable"? As for a version, 1.2beta 1 is very stable, we've been using it on two high-volume production servers for over a month now with only one crash in that time. I would recommend Asterisk 1.2beta1Hope this helps,MATT--- On 10/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I am getting ready to purchase my first Digium card to start experimenting with Asterisk.Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ).I will be using Asterisk @ Home, so will be Asterisk v1.0.9.I took a quick poke at the lists, and it appears several people have been having issues.Am I better served with a TE410P until 1.2 becomes stable and released?___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.9/118 - Release Date: 10/3/05 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADSI -- is it dead? Worth bothering with?
On 10/5/05, Don Pobanz [EMAIL PROTECTED] wrote: Stephen Bosch wrote: Since Colin Anderson -- in a previous thread -- asked the question about whether ADSI was dead, I thought it was worth discussing. Does anybody else have anything to add? -Stephen- I hope ADSI is not dead! We have 100 Aastra 390 ADSI phones with 20 of them in service. We have locations where the infrastructure will not support packet based phones, even if we wanted to use them, yet we would like to give people some menus to lesson the learning curve. Having a menu that changes based on context is great. I really like the voice mail menu where you do not need to remember any of the keys to press but just read the item and press the button by your selection. The only problem is the length of time it takes to load the menu. I believe there is really no need to load the stuff every time voicemail is dialed. The local phone company offers ADSI service. They have a number to dial to program the ADSI menus. This number only needs to be dialed once. After that things are pretty snappy. I will soon start work on converting the rest of our company to primarily ADSI phones. Don Pobanz First my disclaimer, I have never done anything with ADSI beyond theory. I have always thought that ADSI is the best approach for RJ11 connections that require PBX type features on the phone (mostly support for keys such as call log, messages, options, applications, mute and so on). May be due to my own ignorance, I tried CISCO IP phones (sccp) and SIP IP phones and found some of the features were supported. Granted, not as rich as ADSI or my Merridian PBX desk phone but sufficient for 80% of the cases. I have also discovered to my satisfaction that through tftp I can manage a large number of CISCO sccp and SIP phones with a one time effort. I would be curious about what other members of this list think about the best practices for giving clients functionality on their desk phones. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P and TE406P stability
Hi Matt, Thanks for the response. I wasn't aware you could upgrade the card. In that case, I think I'll make my boss happy by saving money and going with the regular card first. I'm basically checking out how well Asterisk works before putting into production for our office. I figure I'll start with 1.0.9, and hopefully have time to check out 1.2 before we go production, and in that time, the stability will increase even more. Thanks, James Texter From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Wednesday, October 05, 2005 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE411P and TE406P stability Do you specifically need hardware echo cancellation? If not you may want to try the TE405P/TE410P first and if you find that you need the hardware echo-canceller you can upgrade the card for $900(for some strange reason actually saves $100 over the list price of the TE411P) TE405P = $1495.00 quoted upgrade price for TE405P to TE406P = $900 upgraded TE405P to TE406P cost = $2395 TE406P = $2495.00 So it may save you money anyway if you go the upgrade path. As for a version, 1.2beta 1 is very stable, we've been using it on two high-volume production servers for over a month now with only one crash in that time. I would recommend Asterisk 1.2beta1 Hope this helps, MATT--- On 10/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I am getting ready to purchase my first Digium card to start experimenting with Asterisk. Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ). I will be using Asterisk @ Home, so will be Asterisk v1.0.9. I took a quick poke at the lists, and it appears several people have been having issues. Am I better served with a TE410P until 1.2 becomes stable and released? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o memory leak when using call files ?
Hi all,
I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P
(EuroISDN cpe)
connected to another similar asterisk box B acting as EuroISDN master.
I'm performing some load tests by contiously feeding up to concurrent 30
call files to /var/spool/asterisk/outgoing/ on box A
which inititate via a dialplan context/extension a outbound call
(redirected via chan_local) to box b
playing some preexisting wavfiles followed by hangup.
I see a memory growth for the asterisk process from 39M VIRT / 11M RES to
179M VIRT / 22M RES after 1000 completed calls.
The box B acepting/recording the calls doesnt show such a memory growth
over time.
Are there known issues about call files / memory growth in asterisk 1.0.9?
The dial construct via chan_local is there to have some additional
information about the callfile
to be tunneld thru asterisk and to provide direct feedback about the call
success/failure to the callfile generator process.
TIA for any hints,
Bruno
callfile: TESTT05100514541601721
Channel: Local/TESTT05100514541601721:[EMAIL PROTECTED]
Context: test
Extension: s
Priority: 1
CallerId: 01722270201
MaxRetries: 0
WaitTime: 35
RetryTime: 10
Account: TEST
SetVar: SMID1=TESTT05100514541601
SetVar: SMID2=721
SetVar: SMID3=
SetVar: SMINFO=TESTINFO
SetVar: SMOADC=+491722270201
SetVar: SMADC=+491234567890
SetVar: SMRETRYCNT=3
extensions.conf:
[general]
TRUNKTEST=Zap/r1
[test-dial]
;exten = _TEST.,1,SetCallerPres(prohib)
exten = _TEST.,1,SetCallerPres(allowed)
exten = _TEST.,2,Cut(SMID=EXTEN,:,1)
exten = _TEST.,3,Cut(REALEXTEN=EXTEN,:,2)
exten = _TEST.,4,Dial(${TRUNKTEST}/${REALEXTEN},30,ng)
exten = _TEST.,5,DBPut(DIALTEST/${SMID}/HANGUPCAUSE=${HANGUPCAUSE})
exten = _TEST.,6,DBPut(DIALTESTVOICE/${SMID}/DIALSTATUS=${DIALSTATUS})
exten = _TEST.,105,goto(5)
exten = _TEST.,205,goto(5)
[test]
exten = s,1,SetCDRUserField(ID=${SMID1}${SMID2}${SMID3}\;)
exten = s,2,AppendCDRUserField(ADC=${SMADC}\;)
exten = s,3,AppendCDRUserField(OADC=${SMOADC}\;)
exten = s,4,AppendCDRUserField(INFO=${SMINFO}\;)
exten = s,5,Playback(/opt/gucky/test/intro)
exten =
s,6,Playback(/opt/gucky/test/pending/${SMID1}${SMID2}${SMID3}-filea)
exten = s,7,Playback(/opt/gucky/test/gap1)
exten =
s,8,Playback(/opt/gucky/test/pending/${SMID1}${SMID2}${SMID3}-fileb)
exten = s,9,Playback(/opt/gucky/test/gap2)
exten = s,10,Playback(/opt/gucky/test/${SMID1}${SMID2}${SMID3}-filec)
exten = s,11,Hangup
; Hangup during play
exten = h,1,GotoIf($[${CHANNEL} = OutgoingSpoolFailed]?4)
; notify callgenerator about successful accepted call
exten = h,2,TrySystem(rm -f
/opt/gucky/test/pending/${SMID1}${SMID2}${SMID3}*)
exten = h,3,TrySystem(/bin/echo
${CALLGEN_PATH}/done/${SMID1}${SMID2}${SMID3}.SMT)
; do nothing
exten = h,4,NoOp(CHANNEL=${CHANNEL})
; dial attempt failed, have attempt logged as CDR with SM ID, ADC, OADC
exten = failed,1,SetCDRUserField(ID=${SMID1}${SMID2}${SMID3}\;)
exten = failed,2,AppendCDRUserField(ADC=${SMADC}\;)
exten = failed,3,AppendCDRUserField(OADC=${SMOADC}\;)
exten = failed,4,AppendCDRUserField(INFO=${SMINFO}\;)
exten = failed,5,AppendCDRUserField(RETRYCNT=${SMRETRYCNT}\;)
exten = failed,6,AppendCDRUserField(DIALSTATUS=${DIALSTATUS}\;)
; retrieve HANGUPCAUSE,DIALSTATUS stored by -dial context
exten =
failed,7,DBGet(THANGUPCAUSE=DIALTEST/${SMID1}${SMID2}${SMID3}/HANGUPCAUSE)
exten = failed,8,AppendCDRUserField(HANGUPCAUSE=${THANGUPCAUSE}\;)
exten = failed,108,Goto(9)
exten =
failed,9,DBGet(TDIALSTATUS=DIALTEST/${SMID1}${SMID2}${SMID3}/DIALSTATUS)
exten = failed,10,AppendCDRUserField(DIALSTATUS=${TDIALSTATUS}\;)
exten = failed,110,Goto(11)
exten = failed,11,DBDeltree(DIALTEST/${SMID1}${SMID2}${SMID3})
; notify callgenerator about failed call
exten = failed,12,SetVar(UPDASTCALL=/bin/echo \\DIALSTATUS\\)
exten = failed,13,SetVar(UPDASTCALL=${UPDASTCALL}${TDIALSTATUS})
exten =
failed,14,SetVar(UPDASTCALL=${UPDASTCALL}\\/DIALSTATUS\\HANGUPCAUSE\\)
exten = failed,15,SetVar(UPDASTCALL=${UPDASTCALL}${THANGUPCAUSE})
exten = failed,16,SetVar(UPDASTCALL=${UPDASTCALL}\\/HANGUPCAUSE\\)
exten = failed,17,TrySystem(${UPDASTCALL}
${CALLGEN_PATH}/fail/${SMID1}${SMID2}${SMID3}.SMT.tmp /bin/mv
${CALLGEN_PATH}/fail/${SMID1}${SMID2}${SMID3}.SMT.tmp
${CALLGEN_PATH}/fail/${SMID1}${SMID2}${SMID3}.SMT)
;
___
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[Asterisk-Users] Agent/Queue Scalability (Formerly: UPDATE - 512 Calls...)
List users, Please review this exchange between myself and Matt Florell. I am looking for ANY data concerning Asterisk servers running standard Agents and Queues. Hardware configurations, software configurations, Asterisk configurations (especially the number of agents and queues), and descriptions of identified bottlenecks would be ideal. For perspective, we have scaled a singler Asterisk server (4 x 3.16 GHz Xeon processors, 4 GB RAM) to 256 simultaneous calls with digital recording at 70% idle and to 512 simultaneous calls with digital recording at 20% idle. I would like to know what to expect when we add agent channels to handle these calls directed to them via queues in a standard inbound call center environment. I would GREATLY appreciate any user experiences or knowledge you can share. Matt, Thank you so much for all of your help. I hope I can reciprocate in the future. On 10/4/05, *Matt Roth* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We do not use Asterisk Queues or Agents because we found them limiting in terms of functionality and scalability as well as not being as open to call manipulation as the system we built is. Because of this we haven't really used the Queue stat analysis tools out there any more than to look at the kind of stats they generate. I figured as much, but with your experience I didn't think it would hurt to ask. Your statement concerning scalability being a problem with Asterisk Queues and Agents concerns me, because we are planning on using both in our large scale (250-500 concurrent calls) call center setup. Could you expound on it? Are they resource intensive at large numbers of calls, does the code have hard limits, etc? We've dealt with the scalability issue of the Monitor application and thought that would be the last such hurdle. The Asterisk source code looks to be very well written and I was hoping that the other applications that aren't bound to disk I/O do not introduce other bottlenecks. I don't know of any installation that has over 100 agents on a single Asterisk server. You may want to contact the GnuDialer guys, because they thought that they would be able to take a quad Xeon server and put 150 agents using agent/queue on it but they never were able to reliably get over 50 agents on that server. the agent/queue functions on Asterisk really are much more CPU-intensive than just a standard SIP stream. In our initial test 2 years ago the agent/queue load was actually not much less than the meetme architecture that we ended up using and with meetme you have a lot more control and functionality. Your best bet may be to have several agent servers that get their calls sent to them from your main server through IAX2 channels or something like that. It may be wishful thinking, but 50 seemed to be the magic number associated with digital recording scalability. We have contacted the GNU Dialer team but if they do not respond, is anyone aware if they were trying to record the agent channels? If this is strictly an Agents/Queues issue, does anyone know where the CPU spikes occur? For example, will we see significant resource consumption by simply logging in a large number of agents to the Asterisk server or does the load occur as calls are routed through the queues? We'd like to avoid a server farm if possible. Are there other Asterisk applications that present serious scaling issues? We were hoping that our hardware (four 3.16 GHz Intel Xeon processors and 4 GB of RAM) would help us overcome most of them. agent/queue has a lot of functionality (you can see that in the code) and it is not exactly designed for a low memory/CPU footprint. Other than that, if you want to be as optimized as possible, don't use AGI scripts at all and deactivate every Asterisk module you can. Done and done. Are there any metrics concerning Agents/Queues and the required memory/CPU for varying numbers of each? Do they scale linearly? It seems that a large problem with scaling Asterisk is that numbers such as these are hard to come by. Could you tell me a little more about what it is exactly that you are trying to build? Sure. We are developing a three server system to handle inbound calling in a call center environment: 1) Asterisk Server 2) Digital Recording Server 3) Reporting Server The Asterisk Server itself performs no codec translations and no DSP. It will be connected to a Cisco AS5400HPX Universal Gateway that terminates the Ts and handles all TDM to VoIP (SIP) processing. As you know it's a pretty beefy box and we've tried to reserve as many resources on it as possible for running Asterisk and its applications. We
Re: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
On Wed, Oct 05, 2005 at 03:51:38PM -0800, Geo wrote: Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks, Not sure about IAX (1), but IAX2 is widely used. Before asking trivial questions you probably should take the time reading about it in http://voip-info/wiki-Asterisk and similar places, though. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about 3Com(r) 3101 Basic Phone
The 3101 is in the same boat as the rest of the 31xx and 2102B/PE series of 3Com phones. They are all SIP Capable but currently only when used in conjunction with a 3Com VCX system. Every time the phone boots up, it must download a runtime image from either a 3Com NBX or VCX system. The VCX download is SIP, while the NBX is something else (the phones can go either way). Until we can get this sip downloader from 3Com, we will be unable to register any of the referenced phones with Asterisk. Jared Valentine [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, October 04, 2005 12:21 AM To: Jorge Cisneros; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Question about 3Com(r) 3101 Basic Phone I have no idea about the 3XXX series of phones. The 2XXX used to have SIP firmware but I could never get my hands on it. I used to see the SIP 2XXX phones selling on Ebay from time to time. I imagine that even if you can locate the SIP firmware for the old phones, you would have to upload it to the NBX since the phones download their firmware from the NBX at bootup. The best way to connect a 3com system to an * box is via T1/E1 (I have had great sucess). You could also connect them via H323 but 3com's H323 solution involves a Windows server. I would like to see someone reverse engineer thierpcXset or wave software and make an*addon that emulates it. Thanks, Steve Hi, i have one question, the 3Com 3101 Basic Phone work with asterisk, if so i any a especial firmware o another thing. And wath other 3com ip phone product work with asterisk. I think is a good idea to create a list with the all voip device and the status with asterisk. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P and TE406P stability
One piece of advice though. Eliminate any echo as best you can before introducing your PBX into Production. Once users have experienced echo issues they will often continue to complain even after the issue is corrected. It really can taint the overall preception of asterisk's performance. Also get some decent phones. Thanks, Steve Hi Matt, Thanks for the response. I wasn't aware you could upgrade the card. In that case, I think I'll make my boss happy by saving money and going with the regular card first. I'm basically checking out how well Asterisk works before putting into production for our office. I figure I'll start with 1.0.9, and hopefully have time to check out 1.2 before we go production, and in that time, the stability will increase even more. Thanks, James Texter From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Wednesday, October 05, 2005 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE411P and TE406P stability Do you specifically need hardware echo cancellation? If not you may want to try the TE405P/TE410P first and if you find that you need the hardware echo-canceller you can upgrade the card for $900(for some strange reason actually saves $100 over the list price of the TE411P) TE405P = $1495.00 quoted upgrade price for TE405P to TE406P = $900 upgraded TE405P to TE406P cost = $2395 TE406P = $2495.00 So it may save you money anyway if you go the upgrade path. As for a version, 1.2beta 1 is very stable, we've been using it on two high-volume production servers for over a month now with only one crash in that time. I would recommend Asterisk 1.2beta1 Hope this helps, MATT--- On 10/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I am getting ready to purchase my first Digium card to start experimenting with Asterisk. Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ). I will be using Asterisk @ Home, so will be Asterisk v1.0.9. I took a quick poke at the lists, and it appears several people have been having issues. Am I better served with a TE410P until 1.2 becomes stable and released? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.9/116 - Release Date: 9/30/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Canceling
You know what.. I have sporadic echo issues too and I just checked my dmesg and also see that! What's this all about? *STOP* You will receive these messages if you send or receive faxes. I asked for this particular procedure to be executed because I was curious to see if zaptel was seeing an echo cancel disable tone when calling the numbers with extreme echo. Again, this is NORMAL to see and EXPECTED if you are sending or receiving faxes. He's not calling a fax machine (I suspect he's not anyway) so I wanted to make sure that the zaptel echocan was NOT hearing the disable tone. Identifying why a echo cancel tone is occurring on a normal voice call is reasonable, but why would a _local_ echo canceller be needed on a four-wire full-duplex digital link? If the end-to-end call is digital all the way, there really isn't a need for it. So, isn't the issue one of who has responsibility for inserting the echo canceller when a 4-wire to 2-wire hybrid is involved? (Obviously, its not the originating site since one would have no idea what the destination site is doing.) Or, is there an assumption going on that says an echo canceller is always needed on all pstn calls regardless of whether is doing anything or not? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about 3Com(r) 3101 Basic Phone
Maybe you can buy this phone and extract the SIP firmware. These phones pre-date the VCX system. http://cgi.ebay.com/3COM-SIP-Phone-IP-VBX-Ethernet-Ports-Like-2102-1102_W0QQitemZ5815517891QQcategoryZ11909QQssPageNameZWDVWQQrdZ1QQcmdZViewItem - Original Message - From: Jared Valentine To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; 'Jorge Cisneros' Sent: Wednesday, October 05, 2005 12:20 PM Subject: RE: [Asterisk-Users] Question about 3Com(r) 3101 Basic Phone The 3101 is in the same boat as the rest of the 31xx and 2102B/PE series of 3Com phones. They are all SIP Capable but currently only when used in conjunction with a 3Com VCX system. Every time the phone boots up, it must download a runtime image from either a 3Com NBX or VCX system. The VCX download is SIP, while the NBX is something else (the phones can go either way). Until we can get this sip downloader from 3Com, we will be unable to register any of the referenced phones with Asterisk. Jared Valentine [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve TotaroSent: Tuesday, October 04, 2005 12:21 AMTo: Jorge Cisneros; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Question about 3Com(r) 3101 Basic Phone I have no idea about the 3XXX series of phones. The 2XXX used to have SIP firmware but I could never get my hands on it. I used to see the SIP 2XXX phones selling on Ebay from time to time. I imagine that even if you can locate the SIP firmware for the old phones, you would have to upload it to the NBX since the phones download their firmware from the NBX at bootup. The best way to connect a 3com system to an * box is via T1/E1 (I have had great sucess). You could also connect them via H323 but 3com's H323 solution involves a Windows server. I would like to see someone reverse engineer thierpcXset or wave software and make an*addon that emulates it. Thanks, Steve Hi, i have one question, the 3Com® 3101 Basic Phone work with asterisk, if so i any a especial firmware o another thing. And wath other 3com ip phone product work with asterisk. I think is a good idea to create a list with the all voip device and the status with asterisk. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.11.9/116 - Release Date: 9/30/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuration settings required for Vonage
Hi, Does anybody know what configuration settings are required to setup Asterisk for vonage? Zeeshan A Zakaria ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuration settings required for Vonage
Do you have a SIP phone account with them?On 10/5/05, Zeeshan [EMAIL PROTECTED] wrote: Hi,Does anybody know what configuration settings are required to setupAsterisk for vonage?Zeeshan A Zakaria___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel tone description
Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo Poppi. Date: Wed, 5 Oct 2005 19:02:23 +0800 From: Lilantha Karunaratne [EMAIL PROTECTED] Subject: [Asterisk-Users] Zaptel tone description To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: !~!UENERkVCMDkAAQACABgA2xL2SF4ybEylW69jV2juZcKQJ0Zvf9/YaUil/[EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi, We're trying to use TDM04B with a few analog switches and we've noticed that it works with the tones from USA only. As it's documented saying that those tones are hard-coded in the source for analog cards. We'd like to know if there's anyone who could tell us under which file these settings would be hard-coded as we could like to do some experiments which will benefit to all Zaptel / Asterisk users in Asian part of the world. Would appreciate an early reply. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call-in/Call-out
Ah It was a typo. It should work now! L:)
--- Erik Slooff [EMAIL PROTECTED] wrote:
snip
written by Crystal Stream, Incorporated
Here is my extensions.conf file. Things have been
left
out or changed to protect the innocent.
Why isn't it working when I call from the outside
that
when I press 124 it repeats the menu and doesn't
initiate DISA correctly to dial out?
[general]
static=yes
writeprotect=yes
[globals]
[voicepulse-in]
exten = ${OURVOIP1},1,Noop(${DATETIME}
${CALLERID})
exten = ${OURVOIP1},2,Answer
exten = ${OURVOIP1},3,Goto(main-menu,s,2)
exten = ${OURVOIP1},4,Hangup
[nufone-in]
exten = ${OURVOIP3},1,Noop(${DATETIME}
${CALLERID})
exten = ${OURVOIP3},2,Answer
exten = ${OURVOIP3},3,Goto(main-menu,s,2)
exten = ${OURVOIP3},4,Hangup
[incoming-sip]
include = voicepulse-in
include = nufone-in
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,Answer
exten = s,3,Goto(main-menu,s,6)
exten = s,4,Hangup
MAIN MENU ;;
[main-menu]
include = operator
include = queues
; if pressed 4-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for main menu selections:
exten = 1,1,Goto(office-day,1,1)
exten = 2,1,Goto(office-day,2,1)
exten = 3,1,Goto(office-day,3,1)
exten = 4,1,Goto(office-day,4,1)
exten = 5,1,Goto(office-day,5,1)
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,Wait(${WAIT_AFTER_ANSWER})
exten = s,3,SetCallerID(${CALLERID})
exten = s,4,DigitTimeout,2
exten = s,5,ResponseTimeout,7
exten =
s,6,Background(/usr/local/etc/asterisk/ivr/UPGRADEPHONES)
exten =
s,7,Background(/usr/local/etc/asterisk/ivr/GREETING)
exten = s,8,WaitExten(1.2)
exten = s,9,SetGlobalVar(prompt_loops=0)
exten =
s,10,GotoIfTime(07:00-18:00|mon-thu|*|*?office-day,s,2)
exten =
s,11,GotoIfTime(10:00-16:30|fri|*|*?office-day,s,2)
exten = s,12,Goto(office-night,s,1)
exten = t,1,Goto(main-menu,#,1) ; If
they
take too long, go to hangup
; invalid
exten = i,1,Wait(1)
exten = i,2,Playback(invalid) ; That's not
valid,
try again
exten = i,3,Wait(1)
exten = i,4,Goto(s,6)
; #=hangup
exten = #,1,Wait(1)
exten = #,2,Playback(vm-goodbye)
exten = #,3,Wait(2)
exten = #,4,Hangup
[office-day]
include = operator
include = queues
; if pressed 3-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for accessing voicemail:
include = voicemail
exten = s,1,SetGlobalVar(prompt_loops=1)
exten = s,2,WaitExten(${BETWEEN_PROMPTS})
exten =
s,3,Background(/usr/local/etc/asterisk/ivr/MAINMENU)
exten = s,4,WaitExten(4)
exten =
s,5,SetGlobalVar(prompt_loops=$[${prompt_loops} +
1])
exten = s,6,GotoIf($[${prompt_loops}
${MAX_MENU_LOOPS}] ? 2:23)
exten = s,7,Goto(operator,0,1)
; invalid
exten = i,1,Playback(invalid) ; That's not
valid,
try again
exten = i,2,Wait(1)
exten = i,3,Goto(s,7)
; timeout
exten = t,1,Goto(operator,0,1)
[office-night]
include = operator
include = queues
; if pressed 3-digit extension:
include = local
; conferences from outside:
include = conferences-external
; for accessing voicemail:
include = voicemail
exten = s,1,SetGlobalVar(prompt_loops=1)
exten = s,2,WaitExten(${BETWEEN_PROMPTS})
exten =
s,3,Background(/usr/local/etc/asterisk/ivr/MAINMENU)
exten = s,4,WaitExten(4)
exten =
s,5,SetGlobalVar(prompt_loops=$[${prompt_loops} +
1])
exten = s,6,GotoIf($[${prompt_loops}
${MAX_MENU_LOOPS}] ? 2:23)
exten = s,7,Goto(operator,0,1)
; invalid
exten = i,1,Wait(1)
exten = i,2,Playback(invalid) ; That's not
valid,
try again
exten = i,3,Wait(1)
exten = i,4,Goto(s,4)
; timeout
exten = t,1,Goto(main-menu,#,1) ; If
they
take too long, go to hangup
[local]
; Directory:
exten = 411,1,Directory(crystal-sip|local)
exten = 411,2,Hangup
; DISA
exten = 124,1,Answer
exten = 124,2,DigitTimeout(5)
exten = 124,3,ResponseTimeout(10)
exten = 124,4,Authenticate(16435679)
exten = 124,DISA(4376194673164379|crystal-sip)
and snip
---
Why did you not set a priority here or is it a typo?
Erik
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Re: [Asterisk-Users] Echo Canceling
On Wednesday 05 October 2005 12:22, Rich Adamson wrote: Identifying why a echo cancel tone is occurring on a normal voice call is reasonable, but why would a _local_ echo canceller be needed on a four-wire full-duplex digital link? It's to cancel far-end echo which is occuring on the terminating side of the call (since my end is PRI in and 4-wire right up to the KSU). Admittedly it's not an ideal situation but the normal echo you'd receive is exacerbated because of the delays introduced by bringing the call in to a PC. Ideally the echo canceller should be wherever the call is converted from 4-wire to 2-wire (i.e. at the terminating telco's FXO rack, if my terminology is correct.) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't run app_txfax
I found the problem! Installing spandsp .3 created a symlink that was not removed. Installing spandsp .2 did not replace the link. That cause the wrong library linking in ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura Adapter SPA-2002
Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to make and receive calls properly). Here's Asterisk's log file as we try to access the voicemail with the SPA-2002: Oct 4 12:36:09 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:36:43 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:43:45 WARNING[7130] app_voicemail.c: Couldn't read username Oct 4 12:45:35 WARNING[7287] app_voicemail.c: Couldn't read username Oct 4 12:47:57 WARNING[7490] app_voicemail.c: Couldn't read username Oct 4 12:54:18 WARNING[7931] app_voicemail.c: Couldn't read username Oct 4 13:03:42 WARNING[8608] app_voicemail.c: Unable to read password Oct 4 13:10:27 WARNING[9113] app_voicemail.c: Couldn't read username Can anyone help us? Voicemail-less, Randy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.
On Tue, Oct 04, 2005 at 09:59:56PM +0200, Olle E. Johansson wrote: I think this is a bug. Please open a report in the bug tracker, attaching all the requested information. If a re-invite fails, we should not cancel the call. I am afraid that is exactly what is happening here and would like to investigate this issue further. It is indeed an interesting call flow that we have to prepared for as we are implementing T.38. Well, looks like the Asterisk team did not consider it a bug :-) I kinda think they are right and Asterisk is doing the right thing. It's our ISP's gateway that is not performing according to the RFC. Only thing I can think of to try is to shove an SDP payload in the 488 message and see if the other side honors it. Ray ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hardware vs. Network Inputs
Michael, Doing an All-Network setup is completely doable but there are many factors to consider. First of all, I didn't see any mention of how many connections it takes before Asterisk starts having difficulty with DTMF. You mentioned that the computer is directly connected to a T1, is it the only computer using the T1 or are there others? Also what kind of network is it? Do you have a good SLA? What kind of packet loss do you experience on average? What is your ping time to the Broadvoice proxy that you're using? Are you using any kind of QoS? Remember that Broadvoice only uses G.711u/a so with RTP + UDP + IP overhead you're looking at ~85kbit/s so at around 9-10 concurrent calls you're going to be pushing it a bit with 900Kbit available bandwidth. You might try turning the SIP RelaxDTMF setting on, that may help, also if you don't have and are not planning on getting any Zaptel hardware, consider using Ztdummy or ZapRTC as an RTP timing source. I know that on the wiki it says that they are really only useful for MoH or MeetME but I've found it to help greatly with audio quality and Asterisk's DTMF detection. YMMV. Good Luck! -Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura Adapter SPA-2002
Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to make and receive calls properly). Here's Asterisk's log file as we try to access the voicemail with the SPA-2002: Oct 4 12:36:09 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:36:43 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:43:45 WARNING[7130] app_voicemail.c: Couldn't read username Oct 4 12:45:35 WARNING[7287] app_voicemail.c: Couldn't read username Oct 4 12:47:57 WARNING[7490] app_voicemail.c: Couldn't read username Oct 4 12:54:18 WARNING[7931] app_voicemail.c: Couldn't read username Oct 4 13:03:42 WARNING[8608] app_voicemail.c: Unable to read password Oct 4 13:10:27 WARNING[9113] app_voicemail.c: Couldn't read username Can anyone help us? Voicemail-less, Randy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMOE Badness in kernel...
Has anyone succesfully used TDMoE on Fedora Core 2.6.12+ Kernel versions? I'm having the issue that is in the Mantis bug database with badness with the kernel. My Story: I can get the dynamic span to come up and show OK in the zttool on both machines. However i get errors every second (Warning: detected alarm on channel 1... then channel 2...) And then the next second, i get : alarm cleared on channel 1,... channel 2... etc... No call will go through across the link becuase of the alarms. It looks as if 2.4 Kernel works, but it would be a lot of work to go back in time. Can anyone give me some direction on this. I have setup IAX2 between the two machines, but I would like the ability to use Dial(Zap/group number/Exten) I havent found a solution in reading through the wiki's about doing something similar with IAX.. i.e (Dial/IAX2/group number/$exten)... There are some scripts and macros that require you to code variables and check status of each trunk etc but it would be nice to use a group with IAX, and in the IAX.conf place iax in groups... (unless i just havent' found it)... Help? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] success story: TE406P (quadspan with hardware echocan)
Hi Andrew, I'm using a TE406P too, and I have echocancel=yes in zapata.conf. Is this redundant? Should I take the line out? Please advice. Thanks. AK On 10/3/05, Rod Bacon [EMAIL PROTECTED] wrote: Which version of asterisk and zaptel are you using?Will they work with 1.0.9 ?== Rod BaconEmpowered CommunicationsGround Floor, 102 York St. South MelbourneVictoria, Australia. 3205Phone: +613 99401600Fax: +613 99401650FWD: 512237 ICQ: 5662270== Andrew Kohlsmith wrote: I just wanted to post here and let everyone know that the TE406P (quadspan T1/E1 with hardware echo can) kicks some serious ass. We've been running a PRI now for over a year with Asterisk (every single call in and out is through two Asterisk boxes, including faxes) and while the software based echo cancellation is more than adequate, we'd get the occassional edgy echo and very infrequently get full-out holy shit echo. So far the TE406 has eliminated that entirely. Anyway as I said I just wanted to post here and tell the world that at least as far as I have been able to determine, the extra cost of the hardware echo can is *well* worth the money. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura Adapter SPA-2002
Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to make and receive calls properly). Here's Asterisk's log file as we try to access the voicemail with the SPA-2002: Oct 4 12:36:09 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:36:43 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:43:45 WARNING[7130] app_voicemail.c: Couldn't read username Oct 4 12:45:35 WARNING[7287] app_voicemail.c: Couldn't read username Oct 4 12:47:57 WARNING[7490] app_voicemail.c: Couldn't read username Oct 4 12:54:18 WARNING[7931] app_voicemail.c: Couldn't read username Oct 4 13:03:42 WARNING[8608] app_voicemail.c: Unable to read password Oct 4 13:10:27 WARNING[9113] app_voicemail.c: Couldn't read username Can anyone help us? I don't use the 2002's, but do have a spa3k that is working fine. The above messages tend to suggest that either the definition in voicemail.conf for extn does not exist, or, you might have a 'context' problem, or, the syntax in voicemail.conf is messed up. Copy/paste the key sections of sip.conf (for this adapter), the extensions.conf section that is supposed to handle voicemail, and the definitions in voicemail.conf. Maybe we can spot the problem from that. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo Canceling
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Wednesday, October 05, 2005 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo Canceling
{clip}
Identifying why a echo cancel tone is occurring on a normal voice call
is reasonable, but why would a _local_ echo canceller be needed on a
four-wire full-duplex digital link?
If the end-to-end call is digital all the way, there really isn't a need
for it. So, isn't the issue one of who has responsibility for inserting
the echo canceller when a 4-wire to 2-wire hybrid is involved? (Obviously,
its not the originating site since one would have no idea what the
destination site is doing.) Or, is there an assumption going on that says
an echo canceller is always needed on all pstn calls
regardless of whether is doing anything or not?
Certainly this is correct if you're only considering the transport network,
however it's also possible there are acoustic echos occuring inside the remote
parties handset (ie. cheap handsets) or, if they've got you on a speakerphone,
acoustic echos from the room itself. We do PRI-IAX-PRI calls between Norstar
PBXs (digital path from handset to handset) and were still forced to install
hardware echo cancellers facing towards each PBX to supress the acoustic echos
introduced when a user went on to handsfree mode. If you consider the enormous
number of trashy sets out in the world it becomes more reasonable to simply
provision your own echo cancellation system facing the PSTN just incase.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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Re: [Asterisk-Users] Hardware vs. Network Inputs
I'm also using Broadvoice and was having a lot of problems with DTMF. I 'm in Ft. Lauderdale, FL and I was (inadvertently) using the dca proxy. When I changed it to use the Miami proxy, my DTMF tones started to work reliably I had done some digging and found various posts on the internet where others using Broadvoice had similar problems and changing proxies seemed to have resolved the issue. You may want to give that a shot and see if it helps. On 10/4/05, Michael Stearne [EMAIL PROTECTED] wrote: We are trying to determine how to build out an IVR system we are working on. The system needs to be able to handle probably at most 5-10 concurrent calls at peak times. Other times we are just looking for a reliable service. For incoming calls we've been using BroadVoice VOIP and before that VoicePulse VOIP. VoicePulse's IAX service started dropping DTMF inputs that we were processing soon after launch and after a few months of reliable service from BoradVoice SIP, we are experiencing problems catching the digits (simple 6 digit numbers) that people are inputting. The question: Is it unrealistic to think that an all network solution (meaning calls VOIP in to the machine and out from the machine) for this kind of load is doable? Or, would it be better to get POTS phone lines involved with a hardware solution like: http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400P (or a better product)? If the network approach is possible why are we having so much problems just capturing the digits people are pressing? The machine is connected directly to a T1 that has at least 900kb up and down available at any time and is 3GHz with decent specs. What is the problem here? (Asterisk 1.0.9) Thanks for any input, Michael ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Derek M. A. Lee-Wo Email: (Home) [EMAIL PROTECTED] (Work) [EMAIL PROTECTED] Fax: (US) 413-826-0641 (UK) 08701 338414 Family Portal: http://www.LeeWo.net Personal Blog: http://www.DereksPerspective.com Those who will not risk cannot win - John Paul Jones ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR MySQL
Hi Asteriskers! Ive enable CDR to store data on a remote machine using MySQL. But I have a problem. Analyzing the log, I see some ERROR messages as: -- SIP/21-3787 is ringing == Spawn extension (default, 21, 1) exited non-zero on 'SIP/21-ce14' Oct 5 13:22:54 ERROR[8576]: cdr_addon_mysql.c:161 mysql_log: cdr_mysql: Unknown connection error: (2013) Lost connection to MySQL server during query This occurs every time that extension hangs up the call. Anyone know why asterisk lost connection during query? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codec g723 on Via C3
On Mon, Oct 03, 2005 at 01:05:55PM +0200, Giordano Grandis wrote: Hi, just a question: anyone has never installed g729 codec on VIA motherboard with C3 processor ? I'm having problem with IPP libraries, and Intel said that it works only on Inter processor. Any suggestion? It works for me fine, but I compiled my on p4 proc with p3 optimization... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura Adapter SPA-2002
Here you are: - sip.conf [] type=friend username= secret=password host=dynamic context=sip-trusted [EMAIL PROTECTED] nat=yes qualify=yes canreinvite=no - voicemail.conf = ,Randy Vinas,[EMAIL PROTECTED] - extensions.conf exten = ,1,Macro(sip-stdext,,Bellcore-r2) Randy Rich Adamson wrote: Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to make and receive calls properly). Here's Asterisk's log file as we try to access the voicemail with the SPA-2002: Oct 4 12:36:09 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:36:43 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:43:45 WARNING[7130] app_voicemail.c: Couldn't read username Oct 4 12:45:35 WARNING[7287] app_voicemail.c: Couldn't read username Oct 4 12:47:57 WARNING[7490] app_voicemail.c: Couldn't read username Oct 4 12:54:18 WARNING[7931] app_voicemail.c: Couldn't read username Oct 4 13:03:42 WARNING[8608] app_voicemail.c: Unable to read password Oct 4 13:10:27 WARNING[9113] app_voicemail.c: Couldn't read username Can anyone help us? I don't use the 2002's, but do have a spa3k that is working fine. The above messages tend to suggest that either the definition in voicemail.conf for extn does not exist, or, you might have a 'context' problem, or, the syntax in voicemail.conf is messed up. Copy/paste the key sections of sip.conf (for this adapter), the extensions.conf section that is supposed to handle voicemail, and the definitions in voicemail.conf. Maybe we can spot the problem from that. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer directly to voicemail (blind transfer)?
At 09:11 10/5/2005, Don Pobanz, wrote:
Doug wrote:
Hi,
Have looked around for info about this:
http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail
If we are using 5 digit extensions (10102: 10 for the company,
102 for the extension), where can we put something
so that 102* goes straight to voicemail without
waiting while the extension rings?
Here is what we have in extensions_additional.conf:
exten = 100,1,Goto(ext-local,10100,1)
exten = 101,1,Goto(ext-local,10101,1)
exten = 102,1,Goto(ext-local,10102,1)
exten = 103,1,Goto(ext-local,10103,1)
Would something like this in extensions.conf work?
exten = _XXX*,1,Voicemail(u${EXTEN:1})
I don't believe this is the correct syntax. It should be:
exten = _XXX*,1,Voicemail(u${EXTEN:0:3})
http://www.voip-info.org/wiki/view/Asterisk+variables
Another method would be to prefix with a digit instead of suffix with an
*. For us, all of our extensions are three digits and begin with a 5
or a 6 (5xx or 6xx). To transfer to voice mail we stick an eight in
front of the extension (85xx or 86xx). It works well for us.
Don Pobanz
So, in your case the line would something like this?
exten = _8XXX,1,Voicemail(u${EXTEN:3})
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[Asterisk-Users] IPComms Setup
Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect attempt from 69.15.xxx.xx, request '[EMAIL PROTECTED]' does not exist __ Yahoo! for Good Donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ADSI -- is it dead? Worth bothering with?
I would be curious about what other members of this list think about the best practices for giving clients functionality on their desk phones. See, I have a dirty little secret. One of the primary justifications that is used for VoIP PBX is consolidated physical network - I mean, it's supposed to be easier, right? One network and all that. But you know, I've found myself muttering sometimes: Man, if this was a *regular* phone I wouldn't be having these problems and by that I mean things like bandwidth issues, latency, no spare Ethernet port, vlan'ing, router's messed up today, blabla, all of those considerations go *away* when you use a PSTN emulation or ADSI. Plus you use a phone that the *user* is familiar with. Who doesn't have a Vista phone? This, to me, is a best practice : make sure your user interface is consistent and instantly familiar. ADSI I find interesting because you can still do all of the VoIP goodness with a legacy phone. I was just wondering if there was a future for it, since the ILEC here in Edmonton I don't think even gives out a Vista for residential anymore, they try to sell you a cordless phone. I would seriously consider it if I could get really nice unlocked phones for under $100 and I could deal with the number of ports required. Hell, I still have kilometers of Cat 3 in place from the Meridian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What the heck? Sprint sues Vonage
http://news.com.com/Sprint+Nextel+sues+Vonage+over+VoIP+patents/2100-7352_3-5888789.html?tag=nefd.top Does anyone have any clue what the suit is over and if/how this affects Asterisk's implimentation of VoIP? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: What the heck? Sprint sues Vonage
SORRY! Duplicate... ignore this thread. On 10/5/05, Matt [EMAIL PROTECTED] wrote: http://news.com.com/Sprint+Nextel+sues+Vonage+over+VoIP+patents/2100-7352_3-5888789.html?tag=nefd.top Does anyone have any clue what the suit is over and if/how this affects Asterisk's implimentation of VoIP? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk not detecting PSTN hang-up
On Tue, 4 Oct 2005, Leigh Fereday wrote: I upgraded to CVS, but get the same message in the log. If you are on CVS-HEAD or 1.2, perhaps busypattern= will help you. Call into your Asterisk box on one of the incoming analogue lines and dial through to an extension. Whilst listening to the incoming call, hang up from the calling side. Listen to the called side, and listen to what tone your PBX makes to signal that the line is disconnected. Time the length of the beep and the silence. Say it comes out 1.5 second beep and 0.5 second silence. Put in your zapata.conf for the channel: busydetect=yes busypattern=1500,500 busycount=4 callprogress=no The 1500 is 1500msec or 1.5 secs. 500msec = .5secs. Asterisk will listen to the call and when it hears 4 repeats of that beep-silence pattern it will take the call as finished. Regards, Steve Davies ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer directly to voicemail (blind transfer)?
Doug wrote:
Another method would be to prefix with a digit instead of suffix with an
*. For us, all of our extensions are three digits and begin with a 5
or a 6 (5xx or 6xx). To transfer to voice mail we stick an eight in
front of the extension (85xx or 86xx). It works well for us.
Don Pobanz
So, in your case the line would something like this?
exten = _8XXX,1,Voicemail(u${EXTEN:3})
the 3 needs to be changed to a 1 (strip off the leading 1 character)
I have these two lines in my extensions.conf since all extensions begin
with a 5 or a 6 and I do some other things with numbers that begin with
82xx or 83xx.
exten = _85xx,1,Voicemail(u${EXTEN:1})
exten = _86xx,1,Voicemail(u${EXTEN:1})
Don Pobanz
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Re: [Asterisk-Users] Sipura Adapter SPA-2002
On Wednesday 05 October 2005 19:52, Rich Adamson wrote: Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to make and receive calls properly). Here's Asterisk's log file as we try to access the voicemail with the SPA-2002: Oct 4 12:36:09 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:36:43 WARNING[6636] app_voicemail.c: No entry in voicemail config file for '' Oct 4 12:43:45 WARNING[7130] app_voicemail.c: Couldn't read username Oct 4 12:45:35 WARNING[7287] app_voicemail.c: Couldn't read username Oct 4 12:47:57 WARNING[7490] app_voicemail.c: Couldn't read username Oct 4 12:54:18 WARNING[7931] app_voicemail.c: Couldn't read username Oct 4 13:03:42 WARNING[8608] app_voicemail.c: Unable to read password Oct 4 13:10:27 WARNING[9113] app_voicemail.c: Couldn't read username Can anyone help us? I had a similar problem with sipura and swissvoice phones whereby the phone does not accept DTMF after the initial dial to the voicemail extension. The above messages are generated when * does not recognise your input when asked for the mailbox/password. Look for a DTMF setting in the sipura webpage setup - it defaults to 'auto' but maybe u should try experimenting with 'inband' - this also goes hand-in-hand with the dtmfmode setting for each extension in sip.conf Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel tone description
On Wednesday 05 October 2005 17:46, Ricardo Poppi wrote: Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. Also look at /etc/asterisk/indications.conf Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as H323 gateway
Juanjo, can you provide some more detail about which version you are using both for asterisk and OpenH323, the hardware dimensioning and the amount of traffic you manage with this solution: how many lines, codecs you use? We should manage a full blown PRI (30 channels), the server is SuperMicro with P4DP8G2 motherboard, dual Xeon 2,8 GHZ, 2 GB RAM and dual SCSI 15K HD in RAID 0. We plan to use G729 codec and a Digium TE110P Card. Any detail will be very useful brgds Francesco Pellegrini ++ | Frame Srl | | Via Antonio Cantore 62/10 | | 16149 Genova | | Tel. +39 010 8680570| | Fax. +39 010 6591413 | | Cell. +348 2237798 | ++ On 10/4/05, Juan Jose Comellas juanjo at comellas.com.ar wrote: I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323 in Buenos Aires, Argentina. Currently I'm using direct connections to the telephone company's (iplan) H.323 gateway, but I'm working on using an intermediate H.323 gatekeeper to take advantage of the telephone company's redundant servers. I think the telco uses Cisco hardware, but I'm not completely sure. We've just started using this, but it seems stable so far. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPComms Setup
this isn't working
[IPComms-in]
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,SetCallerID(${CALLERID})
exten = s,3,Answer
exten = s,4,Goto(main-menu,s,2)
exten = s,5,Hangup
What I have is a block of 20 DIDs and I want to accept
calls from all of them.
It would be way to freaking complicated to do
exten = 2027575120,1,Noop(
.
exten = 2027575121,1,Noop(
et cetera
How do I get this done?
__
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Re: [Asterisk-Users] IPComms Setup
it is trying to match the did in your context which it can't do ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New astGUIclient/VICIDIAL version released 1.1.7
Hello, We've released another update to our Asterisk GUI Client suite: 1.1.7 http://astguiclient.sf.net/ The client suite runs on Windows, UNIX and Mac, includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app auto-dialer. This package is free as in GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this revision, we have finished our internationalization framework, introduced HotKeys key-binding into VICIDIAL and made the server-side apps more compliant with Asterisk 1.2. As of this release, all client web-apps and administration pages are available in English and Spanish, with rough translations of French, German, Italian, Portuguese and Greek for the client web-apps only. Check out the project blog for screenshots and more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPComms Setup
Crystal Stream, Incorporated [EMAIL PROTECTED] wrote: Hey Moo. I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect attempt from 69.15.xxx.xx, request '[EMAIL PROTECTED]' does not exist Do you have a context called [voicepulse-in] containing a definition of what to do when someone calls that DDI number? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura Adapter SPA-2002
Here you are: - sip.conf [] type=friend username= secret=password host=dynamic context=sip-trusted [EMAIL PROTECTED] nat=yes qualify=yes canreinvite=no - voicemail.conf = ,Randy Vinas,[EMAIL PROTECTED] - extensions.conf exten = ,1,Macro(sip-stdext,,Bellcore-r2) That all looks very reasonable. I assume the entry in voicemail.conf is in the [local] section/context. Might double-check that. Also, it seems to me that I seen some patches come through on the asterisk-cvs list involving vm and contexts. A quick google search only found a couple hundred entries with only a few from the last couple of months. What version of asterisk are you running? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
At 2:43 PM -0700 10/4/05, trixter http://www.0xdecafbad.com wrote: Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others. So if its not codecs I wonder if its something so generic that the patent would be tossed out upon challenge. Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. http://kansascity.bizjournals.com/kansascity/stories/2005/10/03/daily23.html -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 This perhaps is quite relevant to the Asterisk community. While I don't know the specifics about Vonage, I do know that they have been rumored to have (in the past, or present) used Asterisk in their core for some services. (Voicemail? Conference? Messages?) This, however, is not confirmed. http://www.ilocus.com/ui_dataFiles/news18aug05.htm http://www.google.com/search?num=50hl=enlr=newwindow=1safe=offc2coff=1q=%22vonage+uses+asterisk%22btnG=Search According to public information, Voiceglo uses IAX and Asterisk: http://lists.digium.com/pipermail/asterisk-users/2004-February/036311.html http://www.business2.com/b2/web/articles/0,17863,1059204,00.html FYI: Voiceglo and theglobe.com are the same company for all intents and purposes. Therefore, I am very interested to see if this is merely co-incidental or if there is a reason that Sprint picked out two providers that use Asterisk in their core. Despite hysteria or misinformation on this (and other) lists, there is no direct information that I've seen that this is Sprint making a blanket patent lawsuit against anyone using VoIP. Perhaps this is just some specific feature that they have a legitimate patent on which has been infringed. I doubt this is a codec patent issue, nor an equipment patent issue (as previously discussed on -biz list.) Is there anyone with better detail on the lawsuit specifics able to comment? JT ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please, help test asynchronous generation patch for inclusion in version 1.2
Hi! If you have problems with MusicOnHold or run Meetme, please, gives this patch a try as it might help you. If enough people test this, it will potentially included in the upcoming 1.2 release. Here's the info on Mantis: http://bugs.digium.com/view.php?id=5374 Please, provide feedback on Mantis, both problems and successes. Thanks! Carlos-- We hold [...] that all men are created equal; that they areendowed [...] with certain inalienable rights; that amongthese are life, liberty, and the pursuit of happiness -- Thomas Jefferson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answering Machine Detection
Anyone aware if Digium or Sangoma, or possibly a function of Asterisk, supports answering machine detection on an outbound call? -- Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMOE Badness in kernel...
I'm seeing the same behavior on a Debian system with 2.6.12. I have two systems with Digium Quad T1s in each and I trunk them with TDMoEThis always worked great on 2.4 and up to 2.6.8 but beyond that it either spits out copious amounts of kernel badness and paralyzes the system completely or gives the same results as you mentioned with constant Alarms/Alarm Clears. I've mentioned it on this forum before as well. On Wed, 2005-10-05 at 09:49 -0700, [EMAIL PROTECTED] wrote: Has anyone succesfully used TDMoE on Fedora Core 2.6.12+ Kernel versions? I'm having the issue that is in the Mantis bug database with badness with the kernel. My Story: I can get the dynamic span to come up and show OK in the zttool on both machines. However i get errors every second (Warning: detected alarm on channel 1... then channel 2...) And then the next second, i get : alarm cleared on channel 1,... channel 2... etc... No call will go through across the link becuase of the alarms. It looks as if 2.4 Kernel works, but it would be a lot of work to go back in time. Can anyone give me some direction on this. I have setup IAX2 between the two machines, but I would like the ability to use Dial(Zap/group number/Exten) I havent found a solution in reading through the wiki's about doing something similar with IAX.. i.e (Dial/IAX2/group number/$exten)... There are some scripts and macros that require you to code variables and check status of each trunk etc but it would be nice to use a group with IAX, and in the IAX.conf place iax in groups... (unless i just havent' found it)... Help? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR MySQL
Hi Asteriskers! I've enable CDR to store data on a remote machine using MySQL. But I have a problem. Analyzing the log, I see some ERROR messages as: -- SIP/21-3787 is ringing == Spawn extension (default, 21, 1) exited non-zero on 'SIP/21-ce14' Oct 5 13:22:54 ERROR[8576]: cdr_addon_mysql.c:161 mysql_log: cdr_mysql: Unknown connection error: (2013) Lost connection to MySQL server during query This occurs every time that extension hangs up the call. Anyone know why asterisk lost connection during query? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPComms Setup
Crystal Stream, Incorporated [EMAIL PROTECTED] wrote:
this isn't working
[IPComms-in]
exten = s,1,Noop(${DATETIME} ${CALLERID})
exten = s,2,SetCallerID(${CALLERID})
exten = s,3,Answer
exten = s,4,Goto(main-menu,s,2)
exten = s,5,Hangup
What I have is a block of 20 DIDs and I want to accept calls from all of
them.
It would be way to freaking complicated to do
exten = 2027575120,1,Noop(
.
exten = 2027575121,1,Noop(
et cetera
How do I get this done?
You could wildcard your DDIs, replacing 20, 21 etc. with [23][0-9], or
whatever. Alternatively, you could create a macro that would look
a lot like the body of your [IPComms-in] context, and then call that
from 20 separate DDI exten lines. I'd just go with the wildcard.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_/_/ _/_/
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Re: [Asterisk-Users] inter Asterisk trunking IAX /IAX2
I am using the inter asterisk trunking and the article in voip-info.org will not work. On 10/5/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Oct 05, 2005 at 03:51:38PM -0800, Geo wrote: Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks,Not sure about IAX (1), but IAX2 is widely used. Before asking trivial questions you probably should take the time reading about it inhttp://voip-info/wiki-Asterisk and similar places, though.--Tzafrir Cohen | [EMAIL PROTECTED] | VIM ishttp://tzafrir.org.il | | a Mutt's[EMAIL PROTECTED] | |bestICQ# 16849755 | | friend___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as H323 gateway
I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323 in --may i know which version of asterisk and oh323? On 10/5/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Juanjo, can you provide some more detail about which version you are using both for asterisk and OpenH323, the hardware dimensioning and the amount of traffic you manage with this solution: how many lines, codecs you use? We should manage a full blown PRI (30 channels), the server is SuperMicro with P4DP8G2 motherboard, dual Xeon 2,8 GHZ, 2 GB RAM and dual SCSI 15K HD in RAID 0. We plan to use G729 codec and a Digium TE110P Card. Any detail will be very useful brgds Francesco Pellegrini ++ | Frame Srl | | Via Antonio Cantore 62/10 | | 16149 Genova | | Tel. +39 010 8680570| | Fax. +39 010 6591413 | | Cell. +348 2237798 | ++ On 10/4/05, Juan Jose Comellas juanjo at comellas.com.ar wrote: I have setup Asterisk with chan_oh323 to connect to the PSTN over H.323 in Buenos Aires, Argentina. Currently I'm using direct connections to the telephone company's (iplan) H.323 gateway, but I'm working on using an intermediate H.323 gatekeeper to take advantage of the telephone company's redundant servers. I think the telco uses Cisco hardware, but I'm not completely sure. We've just started using this, but it seems stable so far. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codec g723 on Via C3
try compiling with 586 and change the makefile to disable mmx codes (if any). I remember tohave this working on a few different processors, but forgot how I did it. -apu On 10/3/05, Giordano Grandis [EMAIL PROTECTED] wrote: Hi, just a question: anyone has never installed g729 codec on VIA motherboard with C3 processor ? I'm having problem with IPP libraries, and Intel said that it works only on Inter processor. Any suggestion? Thanks Giordano___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemailmain automatic extension detection?
Is there a way I can have voice mail check calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for clues! -- Mason Loring Bliss [EMAIL PROTECTED] http://blisses.org/ Anything can be impossible, given sufficient bureaucracy. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
