Re: [asterisk-users] Asterisk Message Logs
[EMAIL PROTECTED] wrote: Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through an Asterisk server. I can't seem to get communication between the two phones. Does anyone have any experience using these open-source Jain-sip-applet-phones? Thanks, Denis Add this to logger.conf: full = notice,warning,error,debug,verbose and you should have most of the output stored in /var/log/asterisk/full Brian. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Message Logs
[EMAIL PROTECTED] wrote: Thanks for your reply. I have previously looked at the logger.conf file. I see that the various types of information can be logged in different ways. After setting the various information types with whatever I want logged, is it possible to save the actual logs to a file (ie: As the messages are bring printed, save them all to a file to be viewed later). What do you mean by actual logs? Console (CLI) output? Brian. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jared Smith Sent: Thursday, August 23, 2007 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Message Logs On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote: Is it possible to print the Asterisk message logs to a file, or is this already done? You want to look at the logger.conf configuration file, and see how your Asterisk system is set to log the various types of information (such as debug messages, verbose messages, DTMF messages, etc.) are logged. After changing logger.conf, you can type logger reload at the Asterisk CLI to make the changes take effect. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum retries exceeded on transmission
I've encountered a similar problem with Cisco equipment. The Cisco proxy was not replying to Asterisk with an ACK after * sent an OK. Since version 1.2.14, * was changed so that not receiving an ACK to an OK is considered a FATAL error. The specific change that causes this problem is in sip_answer() in chan_sip.c: res = transmit_response_with_sdp(p, 200 OK, p-initreq, 2); Changing the 2 to a 1 will probably fix it. Note that this is NOT a bug in * but improper implementations--either caused by latency, or a software bug (not sending an ACK). Perhaps it might be beneficial to have an option in sip.conf to change how * handles not receiving an ACK? I know... it's someone else's problem, but might help those of us stuck with buggy implementations in production environments. :) Brian. On 4/12/07, Joao Pereira [EMAIL PROTECTED] wrote: Hello Thanks a lot for your reply. Im now using asterisk-1.2.10 and the problem disappeared. Thanks regards Joao Pereira Edoardo Serra wrote: Same to me !! Calls from OpenSER to Asterisk It happens only with Asterisk versions = 1.2.14 I'm going to capture some traffic Tnx for help Regards Alex Balashov ha scritto: Joao, It sounds like the proxy is not acknowledging the Asterisk's processing of the INVITE, but I could be wrong. It would be helpful to supply a packet capture between OpenSER and Asterisk so we could see the setup flow. Thanks, -- Alex On Tue, 10 Apr 2007, Joao Pereira said something to this effect: Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --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 -- Alex Balashov [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 ___ --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 detect answering machine
On 13-Aug-04, at 12:46 PM, Christian Victor wrote: Hi! Does anyone of you have an idea how to detect an answering machine on a dialout call? I have thought of timing the amount of silence there is at the start of a call, as an answering machine will usually start speaking as soon as the call is picked up. There would probably be a brief silence when a human picks up a handset. This would probably require a bit of work though. Brian. I am working an a voicemail system wich calls the subscriber but I don't want to fill their answering machine. Maybe I could detect somehow if there is incoming voice when playing the message. usually real persons don't talk when they listen but answeringmachines do. ;-) Thanks in advance, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Database App
Hi,
I was wondering if there is an app available that would let me run
queries on MySQL or Postgress database through the extensions.conf file
in Asterisk.
My goal is to be able to run a simple query on a database that could
return a value in an ${var} variable, or even the number or rows that
were returned in the query in a ${var}. There are a couple of things
in Asterisk I would like to accomplish, without having to fork an AGI
application, that involved getting/checking data in a database.
If there isn't such an app then I will look into creating one, which
I'd be willing to share.
Brian.
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Re: [Asterisk-Users] Database App
My apologies, I was wondering about that.
Brian.
On 15-Jul-04, at 2:48 PM, Steven Critchfield wrote:
After having just seen this in -dev and responding to it, I have to
say... PLEASE do NOT cross post. Almost every person on -dev is also on
-user. Post to one place, and wait for your answer. If someone directs
you to the other, then fine. Respect our time and resources by not
doubling up on your request.
On Thu, 2004-07-15 at 13:21, Brian Jones wrote:
Hi,
I was wondering if there is an app available that would let me run
queries on MySQL or Postgress database through the extensions.conf
file
in Asterisk.
My goal is to be able to run a simple query on a database that could
return a value in an ${var} variable, or even the number or rows that
were returned in the query in a ${var}. There are a couple of things
in Asterisk I would like to accomplish, without having to fork an AGI
application, that involved getting/checking data in a database.
If there isn't such an app then I will look into creating one, which
I'd be willing to share.
Brian.
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--
Steven Critchfield [EMAIL PROTECTED]
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[Asterisk-Users] Predictive Dialers
Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Predictive Dialers
On 9-Jul-04, at 3:29 PM, mattf wrote: Golly, are you thinking about building a predictive dialer with Asterisk? Detecting voicemail and answering machines over 50% of the time is pretty darn tough, and the way that several commercial churn-dialing systems do it is to detect silence or other typical audio patterns (like clicks or quite static) by analyzeing tiny samples(recordings) of the first couple seconds of the phone call(this is one of the reasons we usually hear a couple-second-long quiet pause before a telemarketer comes on the line). This kind of signal processing is very processor intensive for a predictive dialer system and usually occurs through a dedicated DSP on the board(like on a Dialog board) like the kind that us Asterisk users don't have to work with. This makes TRUE predictive dialing on a large scale(more than 6 agents per server) very difficult on an Asterisk system(many say GOOD to this). Thanks for the info Matt. This was exactly what I wanted. :) I am looking at the possibility of developing some type of predictive dialing system, or customize a pre-existing solution. I might try doing some experiments designing an audio processor to detect silence at the beginning of a call. We'll see what I can come up with. Brian. There are a couple dialer projects going on right now with Asterisk that have released code: - Shady-dial (http://shadydial.sourceforge.net/) Lead by some nice Europeans, they have a beta of it up and running supposedly handling up to 10 agents per server, although I'm not sure of exactly what level of 'Predictive' the dialer is(whether it detects voicemail/answering machines and such). This dialer alters the code of Asterisk and is dependant upon PostgreSQL as a database backend. There is some documentation on installation and usage and it is released mostly under the GPL. This dialer does not restrict what kind of phones you can use with it(I'd love to hear more from them or people using this system on the specifics of their project) - VICIDIAL (http://astguiclient.sf.net/) Lead by my company, we are currently developing our 6th production release of a one-call-at-a-time dialer(this is NOT predictive). We have placed over one million calls through this system in the last 12 months mostly to the UK and Australia. It has complete installation instructions, full web-based administration as well as a cross-platform GUI client. It is mostly written in perl, runs on top of an unaltered Asterisk codebase, is dependant on a MySQL as it's backend database and is released under the GPL. If you use this you are limited to SIP clients and Zaptel channels. There are currently 6 companies that are using this system in production envoronments that I know about.(I'd love to hear from other companies using this system too.) MATT--- -Original Message- From: Brian Jones [mailto:[EMAIL PROTECTED] Sent: Friday, July 09, 2004 2:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Predictive Dialers Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/SIP solution?
I've heard of this: http://sarp.sourceforge.net I have no idea if it will be of any help, but it's an interesting programme for handleing SIP over NAT. Brian. - Original Message - From: Stig Hess To: [EMAIL PROTECTED] Sent: Sunday, September 28, 2003 2:17 PM Subject: [Asterisk-Users] NAT/SIP solution? Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig
Re: [Asterisk-Users] Cisco Gateways
Same here... Works great once you get the little bugs worked out. Brian. - Original Message - From: Michiel Betel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 16, 2003 10:09 AM Subject: RE: [Asterisk-Users] Cisco Gateways I'm using cico's with SIP... And it works great :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Gomez Sent: dinsdag 16 september 2003 15:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco Gateways Hi all, Just wondering if * can work with Cisco Gateways such as Cisco 2600/3600 routers or a VG200? -- Edward J. Gomez Director of Network Services ProxyMed, Inc 2555 Davie Road, Suite 110 Fort Lauderdale, Florida 33317 (954) 473-1001 x315 (954) 473-1656 FAX http://www.proxymed.com/ Confidential, unpublished property of ProxyMed, Inc. (c) copyright as of the date of this email. ProxyMed, Inc. CONFIDENTIALITY NOTICE: This e-mail message, including any attachments and files transmitted with it, are confidential and are intended solely for the use of the individual or entity to whom they are addressed. It may contain information that is privileged, confidential and exempt from disclosure under applicable laws. Moreover, this communication may contain the original sender's personal views and opinions, which do not necessarily reflect those of ProxyMed, Inc. . If the reader of this message is not the intended recipient, or the employee or agent responsible for delivering the message to the intended recipient, or if you have received this communication in error, please notify us immediately by return e-mail and delete the original message and any copies of it from your system. If you are not the intended recipient, be advised that you have received this e-mail in error, and that any unauthorized review, use, disclosure, distribution, forwarding, printing, or copying of this e-mail is strictly prohibited without our prior, written permission. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Security vulnerability report
But this is the problem... What happens if I'm using the non-cvs version and there is a security vulnerability? Will a patch be released? btw... Does anyone know if the stable version of Asterisk is affected by this vulnerability? Brian. - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 3:32 PM Subject: Re: [Asterisk-Users] Asterisk Security vulnerability report Read the security vulnerability. It referenced CVS as of a certain date. If you aren't keeping up with CVS changes, why are you running CVS at all? One would hope people are not using the latest CVS checkup as their production system. Most sane people do a bit better quality control and testing then that on a misson critical system. So fielded systems are likely to be a bit in back of CVS. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Security vulnerability report
I heard about this a while ago too. How come I didn't hear anything about it from asterisk-announce? (at least I don't recall receiving any emails about it. Also, is there any plans in the future to create a stable and development branches of code? Upgrading to the lastest CVS version may be difficult for some who have complex installations. It would be easier just to recieve a patch for the stable version. Brian. - Original Message - From: Lubomir Christov [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, September 09, 2003 3:54 PM Subject: [Asterisk-Users] Asterisk Security vulnerability report Hello, today I found this security report regarding Asterisk SIP Security. http://www.securiteam.com/securitynews/5LP0720B5G.html Maybe It could help somebody who isn't using a newer than 15th of August cvs version. Best regards Lubo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dtmf detection from AS5350 over SIP
Hi, Just wondering if anybody has encountered a similar problem as I have with recieving calls on Asterisk from a CISCO AS5350 (over SIP). I have dtmf relay configured on the AS, however, when someone calls in from the PSTN sometimes their digits are inputted twice, which messes up the extensions. If there is a better way to terminate calls from a AS without using SIP, that would fix this problem, then I'd be interested in that too. Have any ideas? If it would help, I could provide you with some of my config files. Brian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAX over IAX
You should be able to packetize your fax calls without any problems. We have hundreds of fax and modem lines on MGCP IADs that connect to PRI gateways without problem, although I haven't tried with IAX. Be sure to use G.711 codec and disable echo cancellation, and if your network is robust with low latency and jitter, you should be fine. Brian Brian F. Jones [EMAIL PROTECTED] 256.705.5012 888.357.0500 x 5012 -Original Message- From: Brian J. Schrock [mailto:[EMAIL PROTECTED] Sent: Thursday, April 03, 2003 9:21 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FAX over IAX From what I have heard packetizing fax does not work well, does not matter if it is IAX or SIP. I think that was straight from digium tech support. On Wednesday, April 2, 2003, at 09:53 AM, John Harragin wrote: Hi, We are looking at consolidating our lines with PRI. This will allow the elimination of many fax lines. Some of them will be replaced with this type of config ... PRI * IAX * Channel-Bank FAX We will have daggressor suppressor enabled. Is anyone doing this and should I expect smooth operation? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
