RE: [asterisk-users] Asterisk 1.2.11 and # key
That why, when you dial one # then Asterisk wait to see if you dial two of them. You should consider changing the blindxfer function or play with the timer in the features.conf. In think its look like featuresdigittimeut. For the moment, if you dial #( wait 1 sec) then press it again, the second will work. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Michael Strelnikov Envoyé: 4 septembre 2006 22:47 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [asterisk-users] Asterisk 1.2.11 and # key I have blindxfer = ## line in my features.conf On 9/5/06, David Gagnon [EMAIL PROTECTED] wrote: Are you sure this is not because of the dynamic features in features.conf ? By default, # is defined for the transfer feature. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] De la part de Michael Strelnikov Envoyé: 4 septembre 2006 09:53 À: asterisk-users@lists.digium.com Objet: [asterisk-users] Asterisk 1.2.11 and # key Hello, Does anybody have problems with recognition of the hash (#) key with * 1.2.11? It seams that after pressing # the call is in a progress but no data is sent. Thanks in advance, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Where did you find 8.0.3 SIP image? Cisco website... I didn't noticed 8.0.3 SIP firmware there... Just tried now with the 8.0.2.SR1 image... Keeps on saying registering Have you tried the one on the end - Another SEPmac.xml.cnf example -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Blind transfer 3/4 digits
On Monday, September 04, 2006 3:22 PM Ronald Wiplinger wrote: What's happen to you guys? Nothing. Why? I am not yelling, just asking. Maybe in a bit stressed out kind of way. It is sure not a dialplan question! Without having all necessary information that is hard to say. Maybe one phone comes in a different context than the other etc. Lot's of things that could go wrong in the dialplan. If it would be a dialplan question, than it would be for each dialing, but it isn't. If we are talking about the same context and same way of dialing: True. You mentioned SIP message and that makes me wonder! Are we not using here dtmf ?? I somehow had the impression that you are using the transfer button on the SNOM which would tell the SNOM to transfer the call. You are obviously talking about attended/unattended transfer via Asterisk only, correct? Then ignore my suggestion. If it is a sequence of tones, Well... If you are using inband DTMF: correct. Otherwise DTMF may correspond to SIP messages as well but let's not get into that. I suppose you are using inband DTMF and G.711? than why is it different if it is in a string (like snom) or another phone, with single tones? If the dialplan is not responsible obviously the phones are behaving differently. Maybe the DTMF sequence is not transmitted correctly but on the other hand I am using SNOMs with inband DTMF without any problems. Maybe the phone (as others suggested) is doing some number/pattern matching magic which you have to fiddle with. If we understand this part, than is the question, where can I turn on the system to take a longer break between tones still as a string? The default setup should not be a problem with SNOMs (at least I never read anything about it) but have a look at the features.conf options. That should proof my thoughts (and that without yelling, ... hehehehe) But a lot of exclamationsmarks. :-) Just kidding. As others pointed out: We (at least I) would need the entire picture (the relevant parts of your dialplan etc.) to really help you here. Regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Blind transfer 3/4 digits
On Tuesday, September 05, 2006 2:06 AM Ronald Wiplinger wrote: In my opinion Asterisk remembers all numbers and therefore it does not wait for the 4, since it found a match. This is in VoIP (in my If both phones enter the dialplan the same way and one phone does work then it should not be a problem with the dialplan or with the way Asterisk is doing the match. You pointed that out yourself. AFAIK there is no overlapping in the dialplan. Either the phone (when dialing, doing a SIP transfer etc.) or Asterisk (when doing an attended/unattended transfer) is waiting the specified time for more digits. If no other number is received it then feeds the received number in the dialplan. So either your phone is just transmitting 601, Asterisk only understands 601 or you do have a problem with your dialplan. The only other option would be a general dialplan bug which is not too likely since most of us would have run into the exact same problem. What do debugs on the Asterisk show you? Do a SIP debug etc. Are you using inband or outband DTMF? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reading the raw E1 channels ?
Hi there, With the help of digium E1 card, is there any possibility / solution to tap into an E1 circuit (while sitting in a telco house) ?? Plz provide some guidance / external links. Regards Azher -- This message has been scanned for viruses and dangerous content by NIIT MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one.
Digium really did make an effort in this case and it's not worth I appreciate Digium and Mark. They have released the G729 and G731 source code and you need to pay the license directly to voiceage, not directly to digium. Also with the code of asterisk, there are lot of authors put in their code into asterisk and endedup in nothing. (See Asterisk Disclaimer Policy) in another area instead. History suggests this will come back to you in some History suggests, the authors of the source need to be given with proper rights but not renamed with Mark Spencer. G729 is developed by Voiceage or related group. The fees can directly go to them not swallowed by middle commission agents. There is no harm in paying to voiceage and everyone must do the same in protecting their intelligence in that. The step which Digum recently took is really admiring in releasing the source code of G729 and pay the royalty fees to voiceage. Don't waste anyone's time in discussing further on this issue, since Digum is moving to a different strategy with regards to the G729 licenses. Regards, Kannaiyan On 9/5/06, Justin Newman [EMAIL PROTECTED] wrote: Illegal or not, the license charge is so small it isn't worth the risk. Digium really did make an effort in this case and it's not worth reproducing. If you can, buy their licenses and spend your time contributing in another area instead. History suggests this will come back to you in some way. Justin - Original Message - From: Andrew Joakimsen [EMAIL PROTECTED] To: Justin Newman [EMAIL PROTECTED]; Commercial and Business-Oriented Asterisk Discussion asterisk-biz@lists.digium.com Sent: Monday, September 04, 2006 5:37 PM Subject: Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one. So even if we license from Intel the code, it is illegal to use it with Asterisk because Asterisk is GPL? I still don't get that part On 9/4/06, Justin Newman [EMAIL PROTECTED] wrote: Kannaiyan, It may be helpful to read... http://lists.digium.com/pipermail/asterisk-users/2004-September/057110.html http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Justin -- Date: Sat, 2 Sep 2006 16:21:40 +0800 From: Kannaiyan Natesan [EMAIL PROTECTED] Hi, I heard of a news, that there is a replacement codec available for g729 and accept the g729 codec data for decoding. Anyone familier with this? Also the good news is that it is noted that it works fine with asterisk and the g729 encoded data. Anyone has the link for the free asterisk distribution which can take unilimited channels of g729 codec, data. If there is any royalty need to pay, is that cheaper than the existing g729 cost?. I read somewhere in the net and forgot to save the url and google did not brought me that again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] includes in realtime ??
Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.11 and # key
But the behaviour should like: if pressed once the # should be transmitted. If pressed ## (fast) the it should be blind transfer. Isn't it?On 9/5/06, David Gagnon [EMAIL PROTECTED] wrote: That why, when you dial one # then Asterisk wait to see if you dial two of them. You should consider changing the blindxfer function or play with the timer in the features.conf. In think its look like featuresdigittimeut. For the moment, if you dial #( wait 1 sec) then press it again, the second will work. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] De la part de Michael Strelnikov Envoyé: 4 septembre 2006 22:47 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [asterisk-users] Asterisk 1.2.11 and # key I have blindxfer = ## line in my features.conf On 9/5/06, David Gagnon [EMAIL PROTECTED] wrote: Are you sure this is not because of the dynamic features in features.conf ? By default, # is defined for the transfer feature. David De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] De la part de Michael Strelnikov Envoyé: 4 septembre 2006 09:53 À: asterisk-users@lists.digium.com Objet: [asterisk-users] Asterisk 1.2.11 and # key Hello, Does anybody have problems with recognition of the hash (#) key with * 1.2.11? It seams that after pressing # the call is in a progress but no data is sent. Thanks in advance, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] End of call
Is there any way to know that call is finished? I know there are special tones sent by phone companies but how can I detect them and then configure Asterisk to use it?Thanks,Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] includes in realtime ??
Exactly my point! In my earlier mail, I had a typo in my command. I meant n again tried the command realtime load sipusers name 4000 and also realtime load sipusers username 4000 Its not working yet! Also, if Realtime, I shudn't even be having the need to use the realtime load commands!! I shud change the values in sql, and wham!! it shud be reflected in the call. cheerz, Ben. RR wrote: Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
On 05/09/06, Jay Moore [EMAIL PROTECTED] wrote: Perhaps if answering the simple things politely is too difficult for you, you'd be better off not answering at all. Someday, I hope, you'll find that 'simple' is a relative term. Perhaps if receiving accurate answers without biting off the hand of the person helping you is too difficult for you, you'd be better off paying for a support contract with some reputable organisation? That way you can do no work whatsoever yourself and enjoy never-ending handholding at $150 per incident. That may suit you better. Around peer-support lists, you tend to find an aversion to telling people things they could easily look up or find out for themselves in a few keystrokes. You'll also notice that I took the trouble not only to answer your question, but to come back and re-phrase my answer when I saw you hadn't understood my explanation. You got all that for free. Enjoy! Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware
Tomislav Parčina schrieb: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Where did you find 8.0.3 SIP image? Cisco website... I didn't noticed 8.0.3 SIP firmware there... Also on their ftp: -rwxrwxr-x1 518 201 8136838 Mar 6 2006 cmterm-7970_7971-sip.8-0-2-0.cop -rwxrwxr-x1 518 201 8136765 Mar 28 22:20 cmterm-7970_7971-sip.8-0-2SR1.cop -rwxrwxr-x1 518 201 4106360 May 17 20:19 cmterm-7970_7971-sip.8-0-3.cop -rw-r--r--1 518 201 4114898 Aug 29 20:20 cmterm-7970_7971-sip.8-0-4SR1.cop Just tried now with the 8.0.2.SR1 image... Keeps on saying registering Have you tried the one on the end - Another SEPmac.xml.cnf example Actual problem was with the Phonelabel string being too long (o; Found out with in the logs... Then I tested all images again and only 8.0.2 works with asterisk... all other say on the display that they are registered which asterisk acknowledges...but in the same moment it marks them as UNREACHABLE and only outbound calls are possible with images other than 8.0.2. So I'm staying with SIP 8.0.2 as it also supports XML push whereas the SCCP images don't support it at all... thanx for helping rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] connect with two servers multiple time
Hai,
I have two server. One for the inbound calls and one for
the outbound calls.
When a call somes on the inbound server, I pas this call with a switch
statement through to the outbound server.
On this outbound server I make a outbound call with the Dial statement. (this
works perfect.
If the outbount call is not answersed, I hangup the outbound call, and
signilling the innbound server that the outbound call is not answered.
The inbound call then tries after a couple of seconds (say 5 seconds) to
connect again to the outbound server with the switch statement.
The outbound server than makes a outbound call. (This works also perfect)
But if the outbound call is not answered for the second time, the outbound
server signalling the inbound server that the call is not answered.
Now the problem is that the inbound server doesn't pick up this second signal.
Does anybody got an idee?
These are my setting on the inbound server
iax.conf
outbound_server
type=peer
username=outboundserver
host=server_outbound
extensions.conf
outbound_server
switch = IAX2/outbound_server
These are my setting on the outbound server
iax.conf
outboundserver
type=user
username=outboundserver
context=outbound_dial_conf
extensions.conf
outbound_dial_conf
exten = _X.,1,Dial(Zap/g1/${tel_outdial},30,${Dial_variables})
Arjan
Kroon
Mobillion B.V.
Copernicuslaan 30
Postbus 554 / PO
Box 554
6710 BN Ede
tel: +31 (0)318-648920
fax: +31 (0)318-648839
mobile: +31 (0)6-55871460
email: [EMAIL PROTECTED]
internet: www.mobillion.nl
___
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Re: [asterisk-users] Re: FAX handling
Thanks all for the answers, Calls are being forced into the Fax, so there is no fax detection. I will send some CLI traces as soon as I have one stable platform on which taking them. Right now we are reinstalling things and also investigating which is the context where the fax calls are sent. We are not using NVfax detect as fax enters through a zap channel, not a SIP one. Anyway I will also give it a try. Will keep you posted. JoseOn 05/09/06, Justin Newman [EMAIL PROTECTED] wrote: Let me know if you guys need help with this...Justin--Message: 15Date: Mon, 4 Sep 2006 17:16:00 -0400From: Technical Support [EMAIL PROTECTED]Subject: RE: [asterisk-users] FAX handlingTo: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=us-asciiLook into NVDETECT, and fax2mail script on www.generationd.comFax detection is automaticMD___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to use Grandstream GX-2000 phones for paging
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... http://www.grandstream.com/FAQ/Asterisk.htm There's a PDF there that tells you (a) what settings to put on the phone, and (b) how to configure Asterisk to sent the SIP header that tells the phone to auto-answer. Is paging/intercom possible with Cisco 7905, 7912, 7940, 7960 and 7970 phones? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-dev] Re: [asterisk-users] Digum g729 and g723
Also any experts confirm that the code does not contain any hacking on the computer. I will also wait for the confirmation from Digium to use this code and a clearly defined procedure to pay the license or royalty fee. Kannaiyan On 9/5/06, Raphael Jacquot [EMAIL PROTECTED] wrote: Kannaiyan Natesan wrote: On 9/5/06, Joe shmoe [EMAIL PROTECTED] wrote: Would you like to have the codecs written by Mark Spencer for Asterisk? The same binary codecs available when you purchase a licence? You're in luck! The following link will allow you to have Digiums codecs. http://www.savefile.com/files/20972 Come one come all! would someone from Digium clearly state if this is legit or somehow someone somewhere stole your livelyhood ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] includes in realtime ??
Ben, that's exactly how it is, the load command is only for you to see what's being pulled from the database and to test if realtime has been configured properly. If you see nothing, then I suspect realtime for you isn't really working and the calls that are working are being looked up in the local conf file. You might have to start doing some toubleshooting. What does your extconfig.conf look like? You might wanna post it here. Also, remove or comment out any extensions related info from sip*.conf files. What's the output if you type: asterisk -rx sip show settings | grep -i realtime on the linux command line? Lastly, ensure there's no errors logged with regards to connectivity to the database. Many pieces need to be in sync for it to work properly. I use it with UnixODBC - FreeTDS - MS SQL Server and it works beautifully :) If you're using a local MySQL database, it should be a piece of cake. Check you're loading the res_mysql module, check for config issues in res_mysql.conf and ensure yur user has permissions to access your asterisk database. Hard to suggest how to do all that without knowing ur exact setup. Sorry, the best I can do for now :) Goodluck \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A couple more interviews with Digium staff
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 We've just completed a couple more interviews with Digium staff: File (Josh Colp): http://www.sineapps.com/news.php?rssid=1475 Mog (Matthew O'Gorman): http://www.sineapps.com/news.php?rssid=1465 We've got a couple more in the pipeline and I'll post the links once they're done. Enjoy :) - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE/TS6S6d5vy0jeVcRAosSAJsFb2Pew+RAD9me9y/KYdzGty2isgCfVOiR a/b8RklSQrU5vguVahq/sZk= =MOLM -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] includes in realtime ??
:) , done all!! neway, lemme know if am overlooking something. extconfig.conf == sipusers = mysql,astDb,sip_conf sippeers = mysql,astDb,sip_conf voicemail = mysql,astDb,voicemail_conf extensions = mysql,astDb,extensions_conf sip.conf has got all entries commented, except for [general] context=default rtcachefriends=yes (hmmm.. is the rtcache the culprit??thats my next investigation! but disabling has got issues with VoiceMail Waiting indication etc) Neway, carrying on... sip show settings === Global Settings: SIP Port: 5060 Bindaddress:0.0.0.0 Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Promsic. redir: No SIP domain support:No Call to non-local dom.:Yes URI user is phone no: No Our auth realmasterisk Realm. auth:No Always auth rejects:No User Agent: Asterisk PBX MWI checking interval: 10 secs Reg. context: (not set) Caller ID: asterisk From: Domain: Record SIP history: Off Call Events:Off IP ToS: 0x0 OSP Support:No SIP realtime: Enabled Global Signalling Settings: --- Codecs: none Relax DTMF: No Compact SIP headers:No RTP Timeout:0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: No Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Default Settings: - Context:default Nat:RFC3581 DTMF: rfc2833 Qualify:0 Use ClientCode: No Progress inband:Never Language: (Defaults to English) Musicclass: default Voice Mail Extension: asterisk Realtime SIP Settings: -- Realtime Peers: Yes Realtime Users: Yes Cache Friends: Yes Update: Yes Ignore Reg. Expire: No Auto Clear: 120 Modules loaded = *CLI show modules like res Module Description Use Count res_musiconhold.so Music On Hold Resource 1 res_indications.so Indications Configuration0 res_crypto.so Cryptographic Digital Signatures 1 res_adsi.soADSI Resource1 res_odbc.soODBC Resource0 res_config_odbc.so ODBC Configuration 1 res_agi.so Asterisk Gateway Interface (AGI) 0 res_monitor.so Call Monitoring Resource 1 res_features.soCall Features Resource 1 res_config_mysql.soMySQL RealTime Configuration Driver 0 chan_features.so Feature Proxy Channel0 11 modules loaded Anything else... ??? Theres no issue with mysql connection, cuz changes to extensions is reflected back immediately. cheerz Ben. RR wrote: Ben, that's exactly how it is, the load command is only for you to see what's being pulled from the database and to test if realtime has been configured properly. If you see nothing, then I suspect realtime for you isn't really working and the calls that are working are being looked up in the local conf file. You might have to start doing some toubleshooting. What does your extconfig.conf look like? You might wanna post it here. Also, remove or comment out any extensions related info from sip*.conf files. What's the output if you type: asterisk -rx sip show settings | grep -i realtime on the linux command line? Lastly, ensure there's no errors logged with regards to connectivity to the database. Many pieces need to be in sync for it to work properly. I use it with UnixODBC - FreeTDS - MS SQL Server and it works beautifully :) If you're using a local MySQL database, it should be a piece of cake. Check you're loading the res_mysql module, check for config issues in res_mysql.conf and ensure yur user has permissions to access your asterisk database. Hard to suggest how to do all that without knowing ur exact setup. Sorry, the best I can do for now :) Goodluck \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
Re: [asterisk-users] Codec Thread
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Joe shmoe wrote: Well you can call me a newb all you want.. The software was released to me by a birdie from digium. This is just the source code. Nothing more. You still need the license for the g729 or g723 but this code from digium will allow you to test. You still need to purchase your license remember that. /spy This is a lie. 1) Digium would never make a copy of the g729 code without licence (even for themselves) as they could just issue themselves as many licences as they needed. 2) The Digium G729 code does not do Annex B (as VAD is not supported in Asterisk), which this does. 3) The G723 codec also does VAD (which Asterisk doesn't support). I would not be surprised if these were exactly the same as the ReadyTechnology files from the Intel demo. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE/Tp9S6d5vy0jeVcRAnYxAJ9P7wkc9YSSR9Ykvh596XAbGNiT4wCfb72h Tzia4WItKfWIXpTMz2xi+Ks= =EpKW -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-dev] Re: [asterisk-users] Digum g729 and g723
Also any experts confirm that the code does not contain any hacking on the computer. I will also wait for the confirmation from Digium to use this code and a clearly defined procedure to pay the license or royalty fee. Kannaiyan On 9/5/06, Raphael Jacquot [EMAIL PROTECTED] wrote: Kannaiyan Natesan wrote: On 9/5/06, Joe shmoe [EMAIL PROTECTED] wrote: Would you like to have the codecs written by Mark Spencer for Asterisk? The same binary codecs available when you purchase a licence? You're in luck! The following link will allow you to have Digiums codecs. http://www.savefile.com/files/20972 Come one come all! would someone from Digium clearly state if this is legit or somehow someone somewhere stole your livelyhood ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Kannaiyan Natesan wrote: Digium really did make an effort in this case and it's not worth I appreciate Digium and Mark. They have released the G729 and G731 source code and you need to pay the license directly to voiceage, not directly to digium. Also with the code of asterisk, there are lot of authors put in their code into asterisk and endedup in nothing. (See Asterisk Disclaimer Policy) in another area instead. History suggests this will come back to you in some History suggests, the authors of the source need to be given with proper rights but not renamed with Mark Spencer. G729 is developed by Voiceage or related group. The fees can directly go to them not swallowed by middle commission agents. There is no harm in paying to voiceage and everyone must do the same in protecting their intelligence in that. The step which Digum recently took is really admiring in releasing the source code of G729 and pay the royalty fees to voiceage. Don't waste anyone's time in discussing further on this issue, since Digum is moving to a different strategy with regards to the G729 licenses. I would not be surprised if this is the same person who released the readytechnology code as Digium code. Claiming the code came from Digium may have been the nail in the coffin legally. I (and others) have already started tracking the source and will post all relevant details to the Digium and VoiceAge legal team. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE/T05S6d5vy0jeVcRAlvpAKCLQ/e2sD0lAQKScyxobyn/K7SEuACggL7y BtJZN8mWnM5+BFP3VPwy7zM= =uNqP -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...
Hi all,I think i'm missing something very very basic! I want my calls with DID 48XX (From pstn E1 TE110P) to be answered then playback a file and hangup.Part of my extensions.conf where from-pstn is the context for all calls from pstn line is:
[from-pstn]include = from-pstn-custom ; create this context in extensions_custom.conf to include customizationsinclude = ext-did-custominclude = from-pstn-timecheck ; this has to be included otherwise it overrides ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)[ext-did-custom]exten = _48XX,1,Answerexten = _48XX,n,SetVar(FROM_DID=${EXTEN})exten = _48XX,n,Playback(vm-goodbye)exten = _48XX,n,Hangup
[from-pstn-timecheck]exten = _.,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a DID route hasn't matched them)exten = s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:)exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)exten = s,4,Goto(from-pstn-afthours,s,1)
Problem, look my Asterisk CLI : -- Accepting call from '2132' to '4888' on channel 0/1, span 1 -- Executing Answer(Zap/1-1, ) in new stack -- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack
-- Executing Playback(Zap/1-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'pt') -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1'
-- Executing Goto(Zap/1-1, s|1) in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack
-- Executing PlayTones(Zap/1-1, ring) in new stack -- Executing NVFaxDetect(Zap/1-1, 8) in new stack -- Channel 0/1, span 1 got hangup requestAfter the hangup the call seems to keep executing Dialplan why?? Does this is related with autofallback option in globals???
Why Hangup didn't exit dialplan?Hope some one can help me.-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] includes in realtime ??
Assuming you have the tables as named int he extconfig.conf as well as the database astDB, how about enabling the module app_realtime.so? Also, if you're using mysql, I don't think you need res_odbc, res_config_odbc. Instead try turning on app_realtime.so and pbx_realtime.so and see how you go :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX
Llorenç Suau [EMAIL PROTECTED] writes: Any suggestions, to how I can make that the PBX receives correctly the call, PREFIX+number, to make the external call. Does this link have the right to make calls to the outside world on the PBX? Normally this feature is turned off on typical PBX. cu, Wolfgang ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] includes in realtime ??
If it shows in the show modules command, it means, the module is loaded, right? If yes, ^CLIshow modules like app_re Module Description Use Count app_realtime.soRealtime Data Lookup/Rewrite 0 app_readfile.soStores output of file into a variable0 app_record.so Trivial Record Application 0 app_read.soRead Variable Application0 4 modules loaded *CLI show modules like pbx_realtime.so Module Description Use Count pbx_realtime.soRealtime Switch 1 1 modules loaded :| RR wrote: Assuming you have the tables as named int he extconfig.conf as well as the database astDB, how about enabling the module app_realtime.so? Also, if you're using mysql, I don't think you need res_odbc, res_config_odbc. Instead try turning on app_realtime.so and pbx_realtime.so and see how you go :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...
I've solved the problem, but still not understanding very well why do i need it:I've inserted inside [ext-did-custom]exten=h,1,hangupWhy do i need this? this is not usually used to run something after an hangupcall?
thks!On 9/5/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi all,I think i'm missing something very very basic! I want my calls with DID 48XX (From pstn E1 TE110P) to be answered then playback a file and hangup.Part of my extensions.conf where from-pstn is the context for all calls from pstn line is:
[from-pstn]include = from-pstn-custom ; create this context in extensions_custom.conf to include customizationsinclude = ext-did-custominclude = from-pstn-timecheck ; this has to be included otherwise it overrides ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)[ext-did-custom]exten = _48XX,1,Answerexten = _48XX,n,SetVar(FROM_DID=${EXTEN})exten = _48XX,n,Playback(vm-goodbye)exten = _48XX,n,Hangup
[from-pstn-timecheck]exten = _.,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a DID route hasn't matched them)exten = s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:)exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)exten = s,4,Goto(from-pstn-afthours,s,1)
Problem, look my Asterisk CLI : -- Accepting call from '2132' to '4888' on channel 0/1, span 1 -- Executing Answer(Zap/1-1, ) in new stack -- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack
-- Executing Playback(Zap/1-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'pt') -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1'
-- Executing Goto(Zap/1-1, s|1) in new stack -- Goto (from-pstn,s,1) -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack
-- Executing PlayTones(Zap/1-1, ring) in new stack -- Executing NVFaxDetect(Zap/1-1, 8) in new stack -- Channel 0/1, span 1 got hangup request
After the hangup the call seems to keep executing Dialplan why?? Does this is related with autofallback option in globals???
Why Hangup didn't exit dialplan?Hope some one can help me.-- Com os melhores cumprimentos,Marco Mouta
-- Com os melhores cumprimentos,Marco Mouta
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Re: [asterisk-users] Can not hear the telco System Announcement
On 9/1/06, Xue Liangliang [EMAIL PROTECTED] wrote: Hi, all. I am from Singapore, we deployed a few PABX based on Asterisk. Here in Singapore there are two Teleco providing E1 pri service, we encountered a strange problem : when calling a number that is unavailible or line suspended, one of the E1 provider keep the call ongoing, because there are system announcement like The line currently I have something similar on a european E1. I do think this has something to do with the PBX.. (asterisk in this case) I have the same 'issue' on a BRI (ISDN) interface. The 'old' PBX (a classic PBX) did sent out the telco announcement. I have tried changing priindication, but this didn't help. I can see the hangup_cause and can play prompts according to the hangup_cause, but I would prefer using the telco announcement. cheers.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...
In article [EMAIL PROTECTED],
Marco Mouta [EMAIL PROTECTED] wrote:
I've solved the problem, but still not understanding very well why do i need
it:
I've inserted inside [ext-did-custom]
exten=h,1,hangup
Why do i need this? this is not usually used to run something after an
hangupcall?
thks!
Your problem is this line:
exten = _.,1,Goto(s,1) ; catch-all matching for calls that have DID
info (if a DID route hasn't matched them)
The pattern _. will match absolutely anything, and so when the line hangs up,
and Asterisk looks for the 'h' extension, it finds _. which matches, and does
the goto back to 's'!!!
You should never use _. as a pattern. If you want to match any NUMBER, you can
do _X. to match two or more digits, and if you also want to match a single
digit you add a second line with _X as the extension.
Using X ensures that the pattern won't match any of the special non-numeric
extensions like h, i, t and so on.
Hope this helps.
Cheers
Tony
On 9/5/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi all,
I think i'm missing something very very basic! I want my calls with DID
48XX (From pstn E1 TE110P) to be answered then playback a file and hangup.
Part of my extensions.conf where from-pstn is the context for all calls
from pstn line is:
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did-custom
include = from-pstn-timecheck ; this has to be included
otherwise it overrides ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
[ext-did-custom]
exten = _48XX,1,Answer
exten = _48XX,n,SetVar(FROM_DID=${EXTEN})
exten = _48XX,n,Playback(vm-goodbye)
exten = _48XX,n,Hangup
[from-pstn-timecheck]
exten = _.,1,Goto(s,1) ; catch-all matching for calls that have
DID info (if a DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours,s,1:)
exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
exten = s,4,Goto(from-pstn-afthours,s,1)
Problem, look my Asterisk CLI :
-- Accepting call from '2132' to '4888' on channel 0/1, span 1
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack
-- Executing Playback(Zap/1-1, vm-goodbye) in new stack
-- Playing 'vm-goodbye' (language 'pt')
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1'
-- Executing Goto(Zap/1-1, s|1) in new stack
-- Goto (from-pstn,s,1)
-- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new
stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in
new stack
-- Goto (from-pstn-reghours,s,2)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing PlayTones(Zap/1-1, ring) in new stack
-- Executing NVFaxDetect(Zap/1-1, 8) in new stack
-- Channel 0/1, span 1 got hangup request
After the hangup the call seems to keep executing Dialplan why?? Does
this is related with autofallback option in globals???
Why Hangup didn't exit dialplan?
Hope some one can help me.
--
Com os melhores cumprimentos,
Marco Mouta
--
Com os melhores cumprimentos,
Marco Mouta
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-=-=-=-=-=-
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Matra 6501
EHLO (o; Anyone succeeded with hooking up a Matra 6501 PBX to * ? cheers rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: why executed Hangup doesn't exit DialPlan?look my dialplan...
Thank you Very MUCH I really appreciate your explanation, i wasn't getting it!On 9/5/06, Tony Mountifield
[EMAIL PROTECTED] wrote:In article
[EMAIL PROTECTED],Marco Mouta [EMAIL PROTECTED] wrote: I've solved the problem, but still not understanding very well why do i need
it: I've inserted inside [ext-did-custom] exten=h,1,hangup Why do i need this? this is not usually used to run something after an hangupcall? thks!
Your problem is this line:exten = _.,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a DID route hasn't matched them)The pattern _. will match absolutely anything, and so when the line hangs up,
and Asterisk looks for the 'h' extension, it finds _. which matches, and doesthe goto back to 's'!!!You should never use _. as a pattern. If you want to match any NUMBER, you cando _X. to match two or more digits, and if you also want to match a single
digit you add a second line with _X as the extension.Using X ensures that the pattern won't match any of the special non-numericextensions like h, i, t and so on.Hope this helps.CheersTony
On 9/5/06, Marco Mouta [EMAIL PROTECTED] wrote: Hi all, I think i'm missing something very very basic! I want my calls with DID
48XX (From pstn E1 TE110P) to be answered then playback a file and hangup. Part of my extensions.conf where from-pstn is the context for all calls from pstn line is:
[from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did-custom
include = from-pstn-timecheck; this has to be included otherwise it overrides ext-did exten = fax,1,Goto(ext-fax,in_fax,1)[ext-did-custom]
exten = _48XX,1,Answer exten = _48XX,n,SetVar(FROM_DID=${EXTEN}) exten = _48XX,n,Playback(vm-goodbye) exten = _48XX,n,Hangup [from-pstn-timecheck]
exten = _.,1,Goto(s,1) ; catch-all matching for calls that have DID info (if a DID route hasn't matched them) exten = s,1,GotoIf($[${IN_OVERRIDE} = forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} = forceafthours]?from-pstn-afthours,s,1:) exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:) exten = s,4,Goto(from-pstn-afthours,s,1)
Problem, look my Asterisk CLI : -- Accepting call from '2132' to '4888' on channel 0/1, span 1 -- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack -- Executing Playback(Zap/1-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'pt')
-- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1' -- Executing Goto(Zap/1-1, s|1) in new stack
-- Goto (from-pstn,s,1) -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in
new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack -- Executing PlayTones(Zap/1-1, ring) in new stack
-- Executing NVFaxDetect(Zap/1-1, 8) in new stack -- Channel 0/1, span 1 got hangup request After the hangup the call seems to keep executing Dialplan why?? Does
this is related with autofallback option in globals??? Why Hangup didn't exit dialplan? Hope some one can help me. -- Com os melhores cumprimentos,
Marco Mouta -- Com os melhores cumprimentos, Marco Mouta -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=-
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--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED]
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-- Com os melhores cumprimentos,Marco Mouta
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[asterisk-users] How to manipulate a plus in a phone number
I'm hoping someone has solved this problem before, because I'm stuck!
I get phone numbers from a database into my dial plan via AGI.
Some of the numbers use the + sign to denote an 'international'
number.
I need to re-write these numbers into a format my IAX provider
can deal with (ie a us style international number starting with 011).
So +3115162728 - 00113115162728
Unfortunately all the ways I've tried to manipulate the number
in the dialplan fail because the + is an operator and I can't get
the parser to treat it as a 'normal' string.
Here are some things I have tried to use to detect the +
exten = 1,n,set(INNAT=${ATELNO:\+})
and
exten = 1,n,set(INNAT=${REGEX(^\+,${ATELNO})})
and
exten = 1,n,set(INNAT=${FIELDQTY(+,${ATELNO})})
None of them work.
I know I could get the database to do the rewriting, but then
I've got database code that is IAX provider dependent, which I'd like
to avoid.
Anyone got any neat tricks??
Tim Panton
www.mexuar.com
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Re: [asterisk-users] Any Hardphone with VPNClient embedded?
Hi! Does any of you knows an Hardphone with VPN client embedded? Take a look at Zultys SIP phones. VPN enabled. www.zultys.com As I too am interested in IPsec capable hardphones (or ATA's), do you have a suggestion what to look at instead? I mean: It's nice to say the company may not be around for long, but if there's no alternative, what choice does one have? You might take a look here (and enhance it where you see fit): http://www.voip-info.org/wiki/view/Asterisk+encryption Apart from that: * Innovaphone has a H.323 ISDN phone with VPN client; not sure if they now also have a SIP phone with VPN client, though * You could employ an AVM Fritz!Box with modded firmware; OpenVPN is a relatively common solution here, not sure about IPSec Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to manipulate a plus in a phone number
Tim Panton wrote:
exten = 1,n,set(INNAT=${REGEX(^\+,${ATELNO})})
Just a grab in the dark, have you tried using single quotes instead of
the double?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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[asterisk-users] latest CentOS-asterisk-freepbx installation procedure
I've tried all those at voip.info.org but I just couldn't get it right. and I don't have the luxury of time to try figure out how to make it work by myself. any other very useful new guides you guys have? tnx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to manipulate a plus in a phone number
On 5 Sep 2006, at 12:04, Doug Lytle wrote:
Tim Panton wrote:
exten = 1,n,set(INNAT=${REGEX(^\+,${ATELNO})})
Just a grab in the dark, have you tried using single quotes instead
of the double?
Sadly not, with :
exten = 1,n,set(INNAT=${REGEX('^\+','${ATELNO}')})
I get :
Sep 5 12:24:01 WARNING[23074]: func_strings.c:105
builtin_function_regex: Malformed input REGEX(^+|+441234567890):
Invalid preceding regular expression
Doug
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www.mexuar.com
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Re: [asterisk-users] latest CentOS-asterisk-freepbx installation procedure
www.nerdvittles.comOn 9/5/06, Roland [EMAIL PROTECTED] wrote: I've tried all those at voip.info.org but I just couldn't get it right. and I don't have the luxury of time to try figure out how tomake it work by myself.any other very useful new guides you guys have? tnx___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Experience Patton BRI gateways and Asterisk?
Hi, can anybody comment on patton inalp voice gateways and Asterisk? How good is there echo cancellation? How good is the interop with Asterisk? I am especially looking for reports on 4630 and 45xx series with BRI. Thanks a lot in advance! Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] latest CentOS-asterisk-freepbx installation procedure
Roland wrote: I've tried all those at voip.info.org but I just couldn't get it right. and I don't have the luxury of time to try figure out how to make it work by myself. The official FreePBX install docs (which have Asterisk instructions as well) for CentOS are here: http://aussievoip.com/wiki/index.php?page=freePBX-Centos cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to manipulate a plus in a phone number
On Tue, Sep 05, 2006 at 11:21:43AM +0100, Tim Panton wrote:
exten = 1,n,set(INNAT=${REGEX(^\+,${ATELNO})})
exten = 1,n,Set(PLUS=\\+)
exten = 1,n,set(INNAT=${REGEX(^${PLUS} ${ATELNO})})
If it is an extension, this should work too
exten = _+.,1,Goto(011${EXTEN:1},1)
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Re: [asterisk-users] Re: How to manipulate a plus in a phone number
On 5 Sep 2006, at 13:21, Stefan Tichy wrote:
On Tue, Sep 05, 2006 at 11:21:43AM +0100, Tim Panton wrote:
exten = 1,n,set(INNAT=${REGEX(^\+,${ATELNO})})
exten = 1,n,Set(PLUS=\\+)
exten = 1,n,set(INNAT=${REGEX(^${PLUS} ${ATELNO})})
That worked, Thanks!
Tim.
Tim Panton
www.mexuar.com
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Re: [asterisk-users] Codec Thread
3) The G723 codec also does VAD (which Asterisk doesn't support). Shame it doesn't... if you could do IAX2 trunking with g723 5.1kbps + VAD, that'd be awesome for narrow links (which is very common in developing countries). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] telco error message on PRI and BRI
hello, Since a few days I'm looking for the 'best' way to get the telco error messages when dialing wrong/busy/non-existing numbers. I can't get it to work on E1 or ISDN BRI. An alternative option is to detect the hangup_cause (no problem here) and play our own voice prompts. I would like to avoid this to make sure the users experience the same behaviour as before. (with the 'traditional PBX') I have tried changing the priindication setting (tried inband,outofband and passthrough) but this didn't change anything. Does anyone have any idea as to how I can debug this? cheers, stoffell ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LinkSys PAP2 ATA Siemens Cordless 3010
Hi everyone, I'm having a problem using this cordless with this ATA. When I try to call that phone, the line is busy. When this phone tries to call someone, no line up. Ata is working with another phone that's not a cordless so it's configured correctly. Any clue about this problem? Best regards Oscar Bossi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN config EWSD
How to configure asterisk and zaptel for ISDN EWSD? Its possible? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can not hear the telco System Announcement
stoffell a écrit : On 9/1/06, Xue Liangliang [EMAIL PROTECTED] wrote: Hi, all. I am from Singapore, we deployed a few PABX based on Asterisk. Here in Singapore there are two Teleco providing E1 pri service, we encountered a strange problem : when calling a number that is unavailible or line suspended, one of the E1 provider keep the call ongoing, because there are system announcement like The line currently I have something similar on a european E1. I do think this has something to do with the PBX.. (asterisk in this case) I have the same 'issue' on a BRI (ISDN) interface. The 'old' PBX (a classic PBX) did sent out the telco announcement. I have tried changing priindication, but this didn't help. I can see the hangup_cause and can play prompts according to the hangup_cause, but I would prefer using the telco announcement. Have you tried progressinband=yes? As far as understand it, it forwards early RTP (that is, stuff that is received prior to the ANSWER), which might just do the trick. I had this working when interconnecting with Chile mobile. When somebody is on the line, they have some music and a message in spanish! Needless to say, with g729 the music part sounds pretty awful (in fact it already sounds awful with g711 anyway...) NB: As far as I can tell, progressinband=yes isn't supported in chan_h323, which is a shame :( Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LinkSys PAP2 ATA Siemens Cordless 3010
yeah...I Got a Siemens Phone and i can't hear the ringing. Try to change these settings in the pap2 device (Admin - Advanced Mode-Regional settings) Voltage = 90V frequency = 20 Hz impedance = 900 ohms waveform = trapezoidal not sure about the question but this is a must i think..On 9/5/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:Hi everyone, I'm having a problem using this cordless with this ATA.When I try to call that phone, the line is busy. When this phone tries to callsomeone, no line up.Ata is working with another phone that's not a cordless so it's configured correctly.Any clue about this problem?Best regardsOscar Bossi___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zero length queue
Hi, I have the following problem. I need queue because of dynamic agents but I only want service as many callers as available members are and want zero length waiting queue. For example. I have two queues (q1,q2) and I use AddQueueMembers and RemoveQueueMembers for maintain queue members. exten = login,1,AddQueueMembers(q1) exten = login,n,AddQueueMembers(q2) exten = logout,1,RemoveQueueMembers(q1) exten = logout,n,RemoveQueueMembers(q2) I look for a solution that make possible if my three agents are logged in and a fouth caller come we give a message that 'all agents are busy. call later' and hangup. No waiting callers in queue just as many calls as many agents. So if Asterisk see this: x*CLI show queue q1 Members: Zap/15 (dynamic) (Busy) has taken 2 calls (last was 126 secs ago) Zap/14 (dynamic) (Busy) has taken 2 calls (last was 739 secs ago) Zap/17 (dynamic) (Busy) has taken 3 calls (last was 341 secs ago) Zap/13 (dynamic) (Busy) has taken 7 calls (last was 156 secs ago) then don't put the next caller in queue but skip Queue(q1) over. Same as with maxlen 0 but with maxlen = 0. The maxlen parameter for queues is exactly what I want but 0 is meaning unlimited in that context so no luck. I have tried with joinempty = no and joinempty = strict but doesn't work. Any idea? bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: Re: [asterisk-users] LinkSys PAP2 ATA Siemens Cordless 3010
thank you for your help. I'll try as soo as I can. Oscar Bossi br br Messaggio originalebr Dal: [EMAIL PROTECTED]br Data: 05/09/2006 15.55br A: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.combr Ogg: Re: [asterisk-users] LinkSys PAP2 ATA amp; Siemens Cordless 3010br br yeah...brbrpstrongI Got a Siemens Phone and i can't hear the ringing. /strongbr /p Try to change these settings in the pap2 device (Admin - Advanced Mode- Regional settings)br Voltage = 90V br frequency = 20 Hz br impedance = 900 ohmsbr waveform = trapezoidal brbrnot sure about the question but this is a must i think..brbrbrbrdivspan class=gmail_quoteOn 9/5/06, b class=gmail_sendernamea href=mailto:[EMAIL PROTECTED][EMAIL PROTECTED] /a/b a href=mailto:[EMAIL PROTECTED][EMAIL PROTECTED]/a wrote: /spanblockquote class=gmail_quote style=border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;Hi everyone, brbrI'm having a problem using this cordless with this ATA.brbrWhen I try to call that phone, the line is busy. When this phone tries to callbrsomeone, no line up.brbrAta is working with another phone that's not a cordless so it's configured brcorrectly.brbrAny clue about this problem?brbrBest regardsbrOscar Bossibr___br-- Bandwidth and Colocation provided by a href=http://Easynews.com;Easynews.com /a --brbrasterisk-users mailing listbrTo UNSUBSCRIBE or update options visit:brnbsp;nbsp; a href=http://lists.digium. com/mailman/listinfo/asterisk-usershttp://lists.digium. com/mailman/listinfo/asterisk-users/abr /blockquote/divbrbr clear=allbr-- brMikebrSales Managerbra href=http://www.theclubvoip.com;http://www.theclubvoip.com/abrMaking it happenbr1.888.470.7253 br br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN config EWSD
Virmones Pereira Tavares de Miranda schrieb: How to configure asterisk and zaptel for ISDN ? EWSD? Hi, below is the ISDN part of my zaptel.conf. Imho crc4 is software selectable in EWSD, thus ask your provider! The D-channel could be found at another location, thus ask your provider! For T1 (or J1) links, the numbers of channels are different! (24 channels per span instead of 32.) Roger. ... # - PRI span 4 for E410P span=4,4,0,ccs,hdb3,crc4 bchan=94-108 dchan=109 bchan=110-124 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can not hear the telco System Announcement
On 9/5/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Have you tried progressinband=yes? As far as understand it, it forwards early RTP (that is, stuff that is received prior to the ANSWER), which might just do the trick. Hm, I have just added this in zapata.conf and sip.conf, and also tried the other values (no, never) but neither of one worked. I can find out the hangup-cause but the telco's message is not played back to me. (line gets dropped and hangupcaused is available, but that's it) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Keys pressed not registering ...
Ok .. So I have moved asterisk to an unrestrictive line and still once the IVR gets going; any keys pressed don't trigger any of my menu options. I have tried all sorts of settings in the sip.conf. John, you mention to switch modes in the trunks/sip.conf, but how can I tell the provider Stanaphone to do this also? I doubt that anyone has to go through such changes being that Stanaphone offers a large variety of their services to the public to all do the same thing. So I'm guessing that there I something improperly configured. Regards, LB -Original Message- From: John covici [mailto:[EMAIL PROTECTED] Sent: Saturday, September 02, 2006 1:13 PM To: Lenny Subject: RE: [asterisk-users] Keys pressed not registering ... The dtmf is in the peer details of the trunk which turns into the sip.conf, however remember that if you change this, your provider has to change it also. on Saturday 09/02/2006 Lenny([EMAIL PROTECTED]) wrote ** See my last email.. Ronald suggested in the sip.conf You suggest the peer details .. That would be for the outgoing settings; isn't this a incoming handler? Anywho .. none of the suggestions worked.. Check my last email as a potential culprit might be the connection im using.. What are your thoughts? Thanks.. LB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Saturday, September 02, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Keys pressed not registering ... In freepbx, its in the peer details of the trunk. on Sunday 09/03/2006 Ronald Wiplinger([EMAIL PROTECTED]) wrote Lenny wrote: Hello Ronald .. This is what I'm trying to learn of now .. Where in freepbx do I place these settings? sip.conf ;-) that was easy, ... do you have another question? bye Ronald Trunk settings? If I could just get that bit of info.. Thanks LB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Saturday, September 02, 2006 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Keys pressed not registering ... Lenny wrote: Hello all, For some reason when dialing in I get the IVR or if I forward to my conference line... any keys pressed seem like they aren?t received .. Like I?m pressing them, but they aren?t being registered with the server .. Any ideas? I?m using the vmware nerdvittles build, the latest trixbox v1.1 .. FreePBX 2.1.1. Everything else works just fine. I?m using VoIPDiscount for outgoing and Stana-in/Stanaphone to receive calls. Any help is appreciated.. Have a look at the dtmfmode settings, inband, rfc2833, ... and try different settings. bye Ronald Regards, LB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Keys pressed not registering ...
Sorry folks.. all is well and the options are now being triggered.. The problem was that while I was configuring the settings I didn't fill in the mode from working on this last week :) Silly mistake; but its all up and running .. I'm sure I'll be back soon, but until thank take care and thanks! Regards, LB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lenny Sent: Tuesday, September 05, 2006 10:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Keys pressed not registering ... Ok .. So I have moved asterisk to an unrestrictive line and still once the IVR gets going; any keys pressed don't trigger any of my menu options. I have tried all sorts of settings in the sip.conf. John, you mention to switch modes in the trunks/sip.conf, but how can I tell the provider Stanaphone to do this also? I doubt that anyone has to go through such changes being that Stanaphone offers a large variety of their services to the public to all do the same thing. So I'm guessing that there I something improperly configured. Regards, LB -Original Message- From: John covici [mailto:[EMAIL PROTECTED] Sent: Saturday, September 02, 2006 1:13 PM To: Lenny Subject: RE: [asterisk-users] Keys pressed not registering ... The dtmf is in the peer details of the trunk which turns into the sip.conf, however remember that if you change this, your provider has to change it also. on Saturday 09/02/2006 Lenny([EMAIL PROTECTED]) wrote ** See my last email.. Ronald suggested in the sip.conf You suggest the peer details .. That would be for the outgoing settings; isn't this a incoming handler? Anywho .. none of the suggestions worked.. Check my last email as a potential culprit might be the connection im using.. What are your thoughts? Thanks.. LB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Saturday, September 02, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Keys pressed not registering ... In freepbx, its in the peer details of the trunk. on Sunday 09/03/2006 Ronald Wiplinger([EMAIL PROTECTED]) wrote Lenny wrote: Hello Ronald .. This is what I'm trying to learn of now .. Where in freepbx do I place these settings? sip.conf ;-) that was easy, ... do you have another question? bye Ronald Trunk settings? If I could just get that bit of info.. Thanks LB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Saturday, September 02, 2006 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Keys pressed not registering ... Lenny wrote: Hello all, For some reason when dialing in I get the IVR or if I forward to my conference line... any keys pressed seem like they aren?t received .. Like I?m pressing them, but they aren?t being registered with the server .. Any ideas? I?m using the vmware nerdvittles build, the latest trixbox v1.1 .. FreePBX 2.1.1. Everything else works just fine. I?m using VoIPDiscount for outgoing and Stana-in/Stanaphone to receive calls. Any help is appreciated.. Have a look at the dtmfmode settings, inband, rfc2833, ... and try different settings. bye Ronald Regards, LB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] includes in realtime ??
If you want to use MWI, and I imagine most people would, you have to cache your realtime data, which means that changes to the tables do not become effective immediately. They become effective after you prune the entry in memory. Doug. -Original Message- From: RR [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 05, 2006 12:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] includes in realtime ?? Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zero length queue
I wrote a little patch to app_queue.c so that the function QUEUEAGENTCOUNT
will only return members that are not busy.
My dialplan goes something like this (in AEL):
SET(QACFREE=${QUEUEAGENTCOUNT(abcstaff|free)});
if (${QACFREE} 0) Queue(abc|trnd|||20);
So if there are free agents, it will join the queue, otherwise, it can do
something else. Contact me off-list if you're interested.
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Artifex Maximus
Sent: Tuesday, September 05, 2006 9:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Zero length queue
Hi,
I have the following problem.
I need queue because of dynamic agents but I only want service as many
callers as available members are and want zero length waiting queue.
For example. I have two queues (q1,q2) and I use AddQueueMembers and
RemoveQueueMembers for maintain queue members.
exten = login,1,AddQueueMembers(q1)
exten = login,n,AddQueueMembers(q2)
exten = logout,1,RemoveQueueMembers(q1)
exten = logout,n,RemoveQueueMembers(q2)
I look for a solution that make possible if my three agents are logged
in and a fouth caller come we give a message that 'all agents are
busy. call later' and hangup. No waiting callers in queue just as many
calls as many agents. So if Asterisk see this:
x*CLI show queue q1
Members:
Zap/15 (dynamic) (Busy) has taken 2 calls (last was 126 secs ago)
Zap/14 (dynamic) (Busy) has taken 2 calls (last was 739 secs ago)
Zap/17 (dynamic) (Busy) has taken 3 calls (last was 341 secs ago)
Zap/13 (dynamic) (Busy) has taken 7 calls (last was 156 secs ago)
then don't put the next caller in queue but skip Queue(q1) over. Same
as with maxlen 0 but with maxlen = 0. The maxlen parameter for
queues is exactly what I want but 0 is meaning unlimited in that
context so no luck.
I have tried with joinempty = no and joinempty = strict but doesn't work.
Any idea?
bye,
Zsolt
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[asterisk-users] Unable to make calls from CallManager to Asterisk
HI,I have successfully integrated CallManager and Asterisk and was able to make call from one of Asterisk phone to CallManager Phone.But Could not able to make call from CallManager to asterisk.I have also tried the below link :- http://www.voip-info.org/wiki/index.php?page=Asterisk+Cisco+CallManager+Integration But still not able to place calls from CallManager to AsteriskCan anybody send me sample of Configuration that i have to make to make calls from CallManager to Asterisk. This is really Urgent for me!!.Thanks in Advance,Anantha Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Experience Patton BRI gateways and Asterisk?
-Ursprüngliche Nachricht- Von: Koopmann, Jan-Peter [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 5. September 2006 13:54 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Experience Patton BRI gateways and Asterisk? Hi, can anybody comment on patton inalp voice gateways and Asterisk? How good is there echo cancellation? How good is the interop with Asterisk? I am especially looking for reports on 4630 and 45xx series with BRI. Hi JP, we used the Smartnodes 1400 and the Smartnodes 2300 in the past. Echo canceler is great and they work really rock stable. Good Support from Patton/Inalp was included. You get many functions for your money, (integrated DSL-Router,QOS,SIP/H323 Support etc.). But eventuallly you pay for functions, you don't really need. I found the BRI Cards from Gerdes Primux2S0/Te/NT and Primux4S0/Te/NT work great and you have to configure only one device, your Asterisk. Another benefit of ISDN cards I see in handling the ISDN-Ports direct in Asterisk, in your dialplan. This gives you a more flexible way of call routing. BTW, multiple Primux Cards in one system are supported! On the other hand, you have some kind of backup by using the Smartnodes, if your asterisk dies. Hope, the informations are usefull Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] File structure question
Thanks Peter, I've also learned with your tips ;)On 9/5/06, Peter Bowyer [EMAIL PROTECTED] wrote: On 05/09/06, Jay Moore [EMAIL PROTECTED] wrote: Perhaps if answering the simple things politely is too difficult for you, you'd be better off not answering at all.Someday, I hope, you'll find that 'simple' is a relative term.Perhaps if receiving accurate answers without biting off the hand of the person helping you is too difficult for you, you'd be better offpaying for a support contract with some reputable organisation? Thatway you can do no work whatsoever yourself and enjoy never-endinghandholding at $150 per incident. That may suit you better. Around peer-support lists, you tend to find an aversion to tellingpeople things they could easily look up or find out for themselves ina few keystrokes.You'll also notice that I took the trouble not only to answer your question, but to come back and re-phrase my answer when I saw youhadn't understood my explanation. You got all that for free. Enjoy!Peter--Peter BowyerEmail: [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different MOH in waiting calls and parked calls
Can I configure different MOH for waiting calls than parked calls?Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wrong CallerID passed to SIP phone
Evenin' (o; Following strange problem: 7970G SIP phone - asterisk - SIP provider In sip.conf I register to my SIP provider to receive calls from them...but as soon the numer rings I see as CallerID the configured outbound number from my SIP account and not who is actually calling... So I gotta lots of missed calls from myself (o; I thought I saw somewhere an option to the Dial command somewhere to pass the CallerID...and oddly I don't see any calling phone number on the CLI with verbosity even set to 8 thanx in advance rick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages
Hi, I hava an Asterisk box with a monoBRI + install-misdn-mqueue 0.3.1-rc23 package installed. I can make outbound calls but cannot receive any. I get no Asterisk messages on the console except for these: P[ 1] GOT IGNORE SETUP P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE] P[ 1] release_chan: Ch not found! Is there anybody who can help me, please? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server
Hi all,Do you think it could be an affordable solution using a two fxs ATA device to connect an old legacy pbx (with few users) with a main asterisk server.phonesanalogueSmallOfficeLegacyPBxATA-2FXS-SIP--MainOffice AsteriskServer This way also I would use ATA device as a Trunk without requiring an Asterisk server on every smalloffice and no need to buy many ATAs neither VoiP hardphones.Is this affordable or i'm missing already basic functions required for a production system? -- Best regardsMarco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server
Marco Mouta wrote: Hi all, Do you think it could be an affordable solution using a two fxs ATA device to connect an old legacy pbx (with few users) with a main asterisk server. phonesanalogueSmallOfficeLegacyPBxATA-2FXS-SIP--MainOffice AsteriskServer This way also I would use ATA device as a Trunk without requiring an Asterisk server on every smalloffice and no need to buy many ATAs neither VoiP hardphones. Is this affordable or i'm missing already basic functions required for a production system? One item you will need to research and tends to create problems for people doing this is line supervision. In other words, disconnect supervision, answer supervision, etc, are often times not provided by legacy pbx's, and therefore the ATA may not recognize hangups, etc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 echo canceller
I have had a bad experience with Asterisk and a Carrier's channel bank. The carrier brought in a PRI (data/voice integrated), the data and voice channels are split from the channel bank. I connected Asterisk to the channel bank via T1 cross cable with a Digium T205. On many calls users hear themselves on the phone during inbound or outbound calls. I have even tried MG2 echo canceller and no relief. I believe a hardware solution might be the way to go. Does anyone have a suggestion of a cheap used or refurbished echo canceller I could use? Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to manipulate a plus in a phone number
At 03:21 AM 9/5/2006, you wrote:
exten = 1,n,set(INNAT=${FIELDQTY(+,${ATELNO})})
None of them work.
This is what I do:
exten = s,n,gotoif($[${EXTEN:0:2} = +1]?fixcid:okcid)
exten = s,n(fixcid), set(xxx=${EXTEN:0:2})
exten = s,n(okcid), noop()
Ira
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Re: [asterisk-users] includes in realtime ??
I use rtcachefriends=yes and any changes I make in my database become effective immediately along with also getting the MWI functionality. Even though what you say makes sense. Go figure! Ben, yeah if it shows it's loaded then it's there for sure. Sorry I asked for it as in your module listing there wasn't any of these modules. I'm at the end of the rope on troubleshooting your issue. Maybe more detail is needed. Esp when you're saying that your sip.conf general section has just two entries. Where's the rest of it, not that a lot needs to necessarily be there if you're not doing anything too tricky. But I would go with removing the rtcache command from the sip.conf file and try and get realtime working in realtime, if that doesn't sound too whacked, just in case it's working off of some cached data, which is why your old codec selection seems to still work even after you change it. Have you looked in your asterisk log file (full) to see if its complaining about errors when you do a realtime load command? The only time my realtime load comes back empty is when it's got a permission problem of some sort on the DB side and one time it happened because of some delay that was introduced coz of some heavy logging or something, don't quite remember it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Find-Me/Follow-ME
Has anyone developed a web interface where users could setup their own find-me/follow-me services? Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 FWD Network: 56505 Website: http://www.upperclassman.net Billing Questions: billing @ upperclassman.net Rental Questions: rentals @ upperclassman.net Maintenance: help @ upperclassman.net This e-mail and any of its attachments may contain sensitive information, which is privileged, confidential, or subject to copyright belonging to Asset Management LLC, Universal Holdings LLC or Upperclassman LLC. This e-mail is intended solely for the use of the individual or entity to which it is addressed. If you are not the intended recipient of this e-mail, you are hereby notified that any dissemination, distribution, copying, or action taken in relation to the contents of and attachments to this e-mail is strictly prohibited and may be unlawful. If you have received this e-mail in error, please notify the sender immediately and permanently delete the original and any copy of or printout of this e-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RR Sent: Tuesday, September 05, 2006 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] includes in realtime ?? I use rtcachefriends=yes and any changes I make in my database become effective immediately along with also getting the MWI functionality. Even though what you say makes sense. Go figure! Ben, yeah if it shows it's loaded then it's there for sure. Sorry I asked for it as in your module listing there wasn't any of these modules. I'm at the end of the rope on troubleshooting your issue. Maybe more detail is needed. Esp when you're saying that your sip.conf general section has just two entries. Where's the rest of it, not that a lot needs to necessarily be there if you're not doing anything too tricky. But I would go with removing the rtcache command from the sip.conf file and try and get realtime working in realtime, if that doesn't sound too whacked, just in case it's working off of some cached data, which is why your old codec selection seems to still work even after you change it. Have you looked in your asterisk log file (full) to see if its complaining about errors when you do a realtime load command? The only time my realtime load comes back empty is when it's got a permission problem of some sort on the DB side and one time it happened because of some delay that was introduced coz of some heavy logging or something, don't quite remember it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
I just ran an SVN update to attempt resolution of this issue and now there is a completely different issue...Very strange. 1. inbound call comes into phone A and is answered. 2. transfer button pressed 3. number of phone B is entered 4. phone B rings and is answered. audio between A and B is good. 5. transfer button on phone A is pressed to complete transfer and is completely ignored. only the conf button appears to function. Can anyone else try todays code with Polycom 1.6.7 and see if they get the same result? Tim McKee attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls but strange messages
Please post your misdn-init.conf as well as misdn.conf so i can try to help uOn 9/5/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi,I hava an Asterisk box with a monoBRI + install-misdn-mqueue 0.3.1-rc23package installed.I can make outbound calls but cannot receive any. I get no Asteriskmessages on the console except for these: P[ 1] GOT IGNORE SETUPP[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE]P[ 1] release_chan: Ch not found!Is there anybody who can help me, please?TIAGiorgio Incantalupo___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk vicidial question
Hello all!,
Ive install all package of Vicidial and astguiclient as I read
on a scratch install notes in a CentOs 4 with trixbox. But when I use some AGIS
added in a dialplan of the install documentation i get some sintax error on this
scripts like agi-VDAcloser_inboundCIDlookup.agi. From the console of asterisk I
saw this errors:
Launched AGI script /var/lib/asterisk/agi-bin/agi-VDADcloser_inboundCIDlookup.
String found where operator expected at
/var/lib/asterisk/agi-bin/agi-VDADcloser_inboundCIDlookup.agi line 283, near
if ($AGILOG) {$agi_string =
(Migth be a runaway multi-line string starting on line 276 )and so
many more errors.
What is wrong with this?...thanks in advance.
G.
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Re: [asterisk-users] Find-Me/Follow-ME
Hi Roger , Has anyone developed a web interface where users could setup their own find-me/follow-me services? Yes, this is available on the ScopServ Telephony GUI (Commercial). -- Joel Vandal, CTO ScopServ Inc. http://www.scopserv.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 echo canceller
On Sep 5, 2006, at 11:56 AM, Michael Araba wrote: I have had a bad experience with Asterisk and a Carrier's channel bank. The carrier brought in a PRI (data/voice integrated), the data and voice channels are split from the channel bank. I connected Asterisk to the channel bank via T1 cross cable with a Digium T205. Sorry to nitpick but a PRI is NOT a data/voice integrated T1. A PRI is a T1 with one channel (normally the last) used for call signalling. If the T1 from the carrier is split into two pieces (data voice) at your office and the hand off to Asterisk is a T1 then it is not a channel bank. A channel bank splits a T1 into 1-24 DS0 voice channels. If your interface with Asterisk is a T1 you should probably look at getting one of the newer Digium cards with the add- on hardware echo canceller. -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Find-Me/Follow-ME
You could implement this very easily yourself. Just write a small webpage that saves the user's find-me/follow-me extension to a text file somewhere (or a database of course) Then write a small agi, that checks for the file (or db value) and sets a variable to jump to that extension M Has anyone developed a web interface where users could setup their own find-me/follow-me services? Yes, this is available on the ScopServ Telephony GUI (Commercial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxing ..
Hello, What are some solutions folks are using for faxes (inbound)? I was considering the Stanafax option. Regards, --- LB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Find-Me/Follow-ME
I was hoping to find a add-on package for my current configuration. I don't need and completely new platform. I have looked at the two previous posted websites. Scopserv is a total package solution from what I gather and iotum is still in beta. Thanks for the quick replys! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal Sent: Tuesday, September 05, 2006 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Find-Me/Follow-ME Hi Roger , Has anyone developed a web interface where users could setup their own find-me/follow-me services? Yes, this is available on the ScopServ Telephony GUI (Commercial). -- Joel Vandal, CTO ScopServ Inc. http://www.scopserv.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 echo canceller
Michael Araba wrote: I believe a hardware solution might be the way to go. Does anyone have a suggestion of a cheap used or refurbished echo canceller I could use? http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Timothy R. McKee wrote: I just ran an SVN update to attempt resolution of this issue and now there is a completely different issue...Very strange. 1. inbound call comes into phone A and is answered. 2. transfer button pressed 3. number of phone B is entered 4. phone B rings and is answered. audio between A and B is good. 5. transfer button on phone A is pressed to complete transfer and is completely ignored. only the conf button appears to function. Can anyone else try todays code with Polycom 1.6.7 and see if they get the same result? Tim McKee I updated to Asterisk SVN-branch-1.2-r41989 and my transfer problems went away. I followed the steps you laid out and it worked fine for me. I'm using a 501 and a 601 running 1.6.7 with the call coming in from an IAX client. -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk vicidial question
You should probably ask this in the VICIDIAL forums:
http://www.eflo.net/VICIDIALforum
Your problem is a known bug in the 2.0.1b1 release that has been fixed
in SVN 2-X trunk.
MATT---
On 9/5/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello all!,
Ive install all package of Vicidial and astguiclient as I read
on a scratch install notes in a CentOs 4 with trixbox. But when I use some AGIS
added in a dialplan of the install documentation i get some sintax error on this
scripts like agi-VDAcloser_inboundCIDlookup.agi. From the console of asterisk I
saw this errors:
Launched AGI script /var/lib/asterisk/agi-bin/agi-VDADcloser_inboundCIDlookup.
String found where operator expected at
/var/lib/asterisk/agi-bin/agi-VDADcloser_inboundCIDlookup.agi line 283, near
if ($AGILOG) {$agi_string =
(Migth be a runaway multi-line string starting on line 276 )and so
many more errors.
What is wrong with this?...thanks in advance.
G.
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Re: [asterisk-users] Faxing ..
Lenny wrote: Hello, What are some solutions folks are using for faxes (inbound)? I was considering the Stanafax option. **Regards,** **---*** **LB*** I got an 500 Internal Server Error when I entered my email address. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different MOH between waiting calls and transfer calls
Could I use different music on hold between waiting calls in queue and calls that are waiting to be tranfered?Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX and rsa
Hi
I am tyring to connect two * boxes over IAX with rsa, but I am having a slight
problem. It just doesn't work. My configuration looks like this:
iax.conf on box 1
[asterisk2]
type=friend
context=main
auth=rsa
inkey=asterisk2.mydomain.com
outkey=asterisk1.mydomain.com
host=asterisk2.mydomain.com
extensions.conf looks like this:
exten = _XX.,1,Dial(IAX2/asterisk2/${EXTEN})
iax on box 2
[asterisk1]
type=friend
context=main
auth=rsa
inkey=asterisk1.mydomain.com
outkey=asterisk2.mydomain.com
host=asterisk1.mydomain.com
extensions.conf looks like this
exten = _XX.,1,Dial(IAX2/asterisk1/${EXTEN})
I generated the key with astgenkey -n asterisk1.mydoamin.com on box 1 and
astgenkey -n asterisk2.mydomain.com on box 2. I have also exchanged the .pub
files between the servers.
When I try to call, I can see on a console that the call is not authenticated.
I know I did something wrong (but what?). Is it possible to have rsa
authentication with type=friend? Any help would be appreciated.
Cheers
Andrutto
--
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Re: [asterisk-users] Different MOH between waiting calls and transfer
equis software wrote: Could I use different music on hold between waiting calls in queue and calls that are waiting to be tranfered? Yes, with the SetMusicOnHold command. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetMusicOnHold Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] blf aastra 9133i working but can't pickup calls
I have not experimented with it lately but I think that is how it is supposed to work. The speedial buttons can be programmed to do BLF+speedial to a given extension. If your getting an incoming call from one of the speeddial extension as indicated by the BLF status you do not pick it up by pressing the speeddial button. You simply lift the handset. -Original Message- From: Jean-Louis curty [mailto:[EMAIL PROTECTED] Sent: Monday, September 04, 2006 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] blf aastra 9133i working but can't pickup calls Hi, I'm trying to get the blf / pickup working properly on the aastra 9133i, I read the wiki voip-info.org for the setup, setup is working fine for the snom, it works also for the aastra ( the light is flashing when a call comes in on another phone ) but I can't pickup the call ... when I press the prog key corresponding the extension I want to pickup, it just dial the extensions like a new call instead of the picking up any idea ? jean-louis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meet-me recording formats
Does anyone know what the options are for the meet-me recording formats?I can't seem to find any documentation on the ${MEETME_RECORDINGFORMAT} variable and what it can be. Michael Lively[EMAIL PROTECTED]229-316-0011 ___
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Re: [asterisk-users] Asterisk Linksys PAP2 ATA
Glad to help. Happy dialling. On September 2, 2006 23:05, Nick Ellson wrote: Hi Tim, The dial plan trick worked great. Added |40[01]x| to my plan and 4000-4019 connect instantly from the PAP2 :) Added it to my X-Lite as well, and worked there too. Thanks! -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpa0WjPdmz6f.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] config include issues
Here is my extensions_custom.conf. The WakeUp context will not work. If I change the context name to say, CRAP, it works like a charm. Can anyone explain this? [from-internal-custom] exten = 1234,1,Playback(demo-congrats) ; extensions can dial 1234 exten = 1234,2,Hangup() exten = h,1,Hangup() include = NewsClips include = WakeUp [NewsClips] exten = 511,1,Answer exten = 511,2,Wait(1) exten = 511,3,AGI(test.php) exten = 511,4,Hangup [WakeUp] exten = 611,1,Answer exten = 611,2,Playback(demo-congrats) exten = 611,3,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk t.38 fax failed
Is T.38 fax work through Asterisk? I have the config below in my sip.conf, but the fax doesn't work and give me the CLI lines below. My current version is 1.2.10. Please help.[Inboundtopbx]type=friendcontext=pbxhost=10.18.188.84insecure=portdtmfmode=rfc2833canreinvite=nodisallow=allallow=g729allow=ulawt38pt_udptl=yest38pt_rtp=not38pt_tcp=no[OutboundfromPBX]type=peerhost=10.18.161.222 canreinvite=nodtmfmode=rfc2833disallow=allallow=g729qualify=yest38pt_udptl=yest38pt_rtp=not38pt_tcp=no-- SIP read from 10.18.188.84:50096: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0Via: SIP/2.0/UDP 10.18.188.84:5060From: sip:[EMAIL PROTECTED];tag=19D429E8-2084To: sip:[EMAIL PROTECTED];tag=as3c87a22eDate: Tue, 05 Sep 2006 19:42:28 GMTCall-ID: [EMAIL PROTECTED]Max-Forwards: 6Content-Length: 0CSeq: 101 ACK--- (9 headers 0 lines)---Sep 5 15:30:31 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 receivedSep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 receivedSep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 receivedSep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 receivedSep 5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown SDP media type in offer: image 16406 udptl t38 Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Catch an event
Hello, I would like for some reasons, catch the ring event since Asterisk, in real-time. Is this information record in a database? How can I read it, immediatly? I either think to catch the information by a little shell script as: asterisk -r |tail -1|grep ring|awk ... and redirect the internal number to an application fir process? Do you see a best way to do that?? The reason of this exercise is simply that i develop a softphone with an iax dll, but we use the french language Windev. All functions are OK, except sounds functions, as ring or dialling tone... So, i try to get some solutions... Best regards, OLS ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Articulation Palm client and Asterisk
Hello, Has anybody configured Asterisk and the Articulation palm client to work ? I can make calls but I cannot make it register to receive calls. It does not register to the box. There are so few parameters that I think Asterisk sip.conf must be changed somewhat. I do not pass any parameters here because my box works perfectly with polycom, grandstream and linksys/sipura, and I know what to touch. The articulation software has only SERVER,DOMAIN, DISPLAY NAME, USER, PASSWORD, codecs are configured correctly (it only supports G711u and GSM), and I configured SERVER=DOMAIN (ip address) since it does not try to register until DOMAIN has something in i, Regards, Jorge Alayon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] config include issues
Curt Shaffer wrote: Here is my extensions_custom.conf. The WakeUp context will not work. If I change the context name to say, CRAP, it works like a charm. Can anyone explain this? What does show dialplan from-internal-custom display? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] config include issues
[ Context 'from-internal-custom' created by 'pbx_config' ] '1234' = 1. Playback(demo-congrats) [pbx_config] 2. Hangup() [pbx_config] 'h' =1. Hangup() [pbx_config] Include ='NewsClips' [pbx_config] Include ='WakeUp' [pbx_config] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, September 05, 2006 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] config include issues Curt Shaffer wrote: Here is my extensions_custom.conf. The WakeUp context will not work. If I change the context name to say, CRAP, it works like a charm. Can anyone explain this? What does show dialplan from-internal-custom display? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and REFER authentication
Hello, The service I am using requires authentication. In sip.conf, setting: [authentication] auth=name:[EMAIL PROTECTED] Gets the authentication working for the INVITES but when I try a transfer, I can see the REFER but then asterisk quickly says BYE. The provider sends back a 401 UNAUTHORIZED but asterisk never resends the REFER with the required authentication info. Is there a way that I can get Asterisk to authenticate on REFERs? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk t.38 fax failed
No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/file_download.php?file_id=9335type=bug For me it only worked well with patch for version 1.2.4 of Asterisk. Regards, Ricardo. Kokfoo Soo wrote: Is T.38 fax work through Asterisk? I have the config below in my sip.conf, but the fax doesn't work and give me the CLI lines below. My current version is 1.2.10. Please help. [Inboundtopbx] type=friend context=pbx host=10.18.188.84 insecure=port dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 allow=ulaw t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no [OutboundfromPBX] type=peer host=10.18.161.222 canreinvite=no dtmfmode=rfc2833 disallow=all allow=g729 qualify=yes t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no -- SIP read from 10.18.188.84:50096: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.18.188.84:5060 From: sip:[EMAIL PROTECTED];tag=19D429E8-2084 To: sip:[EMAIL PROTECTED];tag=as3c87a22e Date: Tue, 05 Sep 2006 19:42:28 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK --- (9 headers 0 lines)--- Sep 5 15:30:31 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:32 NOTICE[25233]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received Sep 5 15:30:34 WARNING[6839]: chan_sip.c:3475 process_sdp: Unknown SDP media type in offer: image 16406 udptl t38 Yahoo! Messenger with Voice. Make PC-to-Phone Calls http://us.rd.yahoo.com/mail_us/taglines/postman1/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com to the US (and 30+ countries) for 2¢/min or less. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linking Asterisk with PBX through E1
Hello,
I linked an Asterisk server to a Brazilian PBX (Leucotron) through an E1
connection, using MFC/R2, that's common down here. The connection works
properly. I'm able both to dial and receive calls through that link,
among their extensions.
The problem is that the PBX configuration is very tough. Just a few
options in the GUI software and I cannot play with it in lower level.
That PBX has two E1 interfaces. One of them is connected to the PSTN and
the other to the Asterisk server. Both connections are working ok.
I need to make calls from the Asterisk server to the PSTN, i.e., coming
from an E1 and going through the other one. Here is my pain. That PBX
assumes that an E1 connection is always PSTN, so an E1 link doesn't
need to talk to each other. Zero flexibility.
The manufacturer support gave me a solution. Coming from Asterisk, I
can dial a special code, then I get a simulated dial tone, and then I
dial (through DTMF) the number I want. That's odd, but it works.
In my case, that code is . Since E1 is digital-signaled, the best to
do would be dialing just like I do between two Asterisks:
exten = _,1,Dial(Unicall/g1/${EXTEN})
But it doesn't work. The PBX just ignores the numbers after and
gives me a dial tone.
Another way would be dialing and then sending the number to dial
through DTMF tones, with something like this:
exten = _,1,Dial(Unicall/g1/|20|D(w${EXTEN}))
That would work, BUT a little detail broke my legs. The Dial
application only sends the DTMF tones after receiving the channel
answered signal from the E1 channel, and that PBX only sends that
signal when the remote party has answered the call, what's useful for
accounting purposes. So, when I dial something using the above dial
plan, Asterisk dials and I hear the dial tone. If I dial something
in my phone (DTMF), the PBX hears that and makes the call. When the
remote party answers the call, the Dial application releases the DTMF
tones.
Possible solutions:
1) Finding a way that Asterisk sends the DTMF tones immediately after
opening the channel, without waiting the answer signal.
2) Making the PBX works the way it should do, receiving all the numbers
in the digital channel and making the call without simulating any dial
tone.
I'm not hopeful that the manufacturer will be able to change the way the
PBX works, so I better keep looking for the first solution.
Any help is pretty welcome.
TIA
--
Marlon Dutra
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[asterisk-users] Adding custom fields (more than one) to CDR DB
Hi all, I just found out how to set the column userfield, in the CDRDBto whatever I desired. Can I add multiple custom columns to the DB and fill them from the dialplan, or is it limited to one column? I am using Asterisk 1.2.4 and MYSQL for the CDR DB. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is this a warning or not...MYSQL Fetch
Hi all, I got the following "warning" in the console (using 1.2.4): Sep 5 16:24:02 WARNING[4375]: app_addon_sql_mysql.c:318 aMYSQL_fetch: ast_MYSQL_fetch: numFields=3 Im not sure why I am being "warned" that there are 3 fields returned by my query (It's what's supposed to happen, the query is SELECT COLA,COLB,COLC FROM TABLEA). 1) Does this apparently wrong warning hide something more dangerous? 2) If not, can I turn this off? I like being able to monitor my console once in a blue moon, and warnings popping all over the place when there is no real issues keep me from doing so efficiently. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding custom fields (more than one) to CDR DB
Mike wrote: Hi all, I just found out how to set the column userfield, in the CDR DB to whatever I desired. Can I add multiple custom columns to the DB and fill them from the dialplan, or is it limited to one column? I am using Asterisk 1.2.4 and MYSQL for the CDR DB. As far as I know, it is just userfield. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why executed Hangup doesn't exit DialPlan?look my dialplan...
Your problem is caused by using exten = _. DON'T DO THAT!
When Hangup() is being run then Asterisk will jump to exten = h Since
_. will match h it will go there.
Marco Mouta wrote:
Hi all,
I think i'm missing something very very basic! I want my calls with DID
48XX
(From pstn E1 TE110P) to be answered then playback a file and hangup.
Part of my extensions.conf where from-pstn is the context for all calls
from
pstn line is:
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did-custom
include = from-pstn-timecheck ; this has to be included otherwise
it overrides ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
[ext-did-custom]
exten = _48XX,1,Answer
exten = _48XX,n,SetVar(FROM_DID=${EXTEN})
exten = _48XX,n,Playback(vm-goodbye)
exten = _48XX,n,Hangup
[from-pstn-timecheck]
exten = _.,1,Goto(s,1) ; catch-all matching for calls that have
DID
info (if a DID route hasn't matched them)
exten = s,1,GotoIf($[${IN_OVERRIDE} =
forcereghours]?from-pstn-reghours,s,1:)
exten = s,2,GotoIf($[${IN_OVERRIDE} =
forceafthours]?from-pstn-afthours,s,1:)
exten = s,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?from-pstn-reghours,s,1:)
exten = s,4,Goto(from-pstn-afthours,s,1)
Problem, look my Asterisk CLI :
-- Accepting call from '2132' to '4888' on channel 0/1, span 1
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, FROM_DID=4888) in new stack
-- Executing Playback(Zap/1-1, vm-goodbye) in new stack
-- Playing 'vm-goodbye' (language 'pt')
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (from-pstn, 4888, 4) exited non-zero on 'Zap/1-1'
-- Executing Goto(Zap/1-1, s|1) in new stack
-- Goto (from-pstn,s,1)
-- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack
-- Goto (from-pstn-reghours,s,1)
-- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in
new stack
-- Goto (from-pstn-reghours,s,2)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing PlayTones(Zap/1-1, ring) in new stack
-- Executing NVFaxDetect(Zap/1-1, 8) in new stack
-- Channel 0/1, span 1 got hangup request
After the hangup the call seems to keep executing Dialplan why?? Does
this is related with autofallback option in globals???
Why Hangup didn't exit dialplan?
Hope some one can help me.
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Re: [asterisk-users] config include issues
Curt Shaffer wrote: Here is my extensions_custom.conf. The WakeUp context will not work. If I change the context name to say, CRAP, it works like a charm. Can anyone explain this? And the output from the console when you dial 611 and it doesn't work? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to notify an ACD agent before he/she picks up
Hi, I need to send a message to an agent when the ACD starts to ring on he/she. I have and application already built that sends such a message (just like a cti), just don't know how to get from asterisk which agent was selected prior to ringing him (or during ringing), so that I can get information about the call and send it over. any one done this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
