[asterisk-users] How to trace incoming AMI requests ?
Hello, Is there a way to trace into a log file, incoming AMI requests ? For instance, I've got several apps accross the LAN, sending AMI requests such as : Action: originate Channel: Local/7...@internal Exten: 00123456789 Priority: 1 ... Some of them might be sometimes producing some erroneous requests. I would like to check from server side, that each received request is formatted and compliant. Ideally, I would to read in a log file, a (reliable) copy of AMI originate requests. Any suggestion ? If this helps, I can use astmanproxy. I also saw a debug=on option inside /etc/asterisk/manager.conf but I couldn't find a change in produced output. And http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.confis not helpful on this specific option. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording from g729 to wav means transcoding ?
Hello list, when the conversation is using the G729-codec and the conversation is recorded with the Monitor()-application in wav-format, will there be transcoding (and thus a need for licenses ?) Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: fail2ban, spam and mail servers
Many of you are interested in and have used or recommended fail2ban for your linux boxes. I finally installed it on our FreeBSD server (no asterisk, hence the OT) with the help of a friend from the VoIP Users Conference and Asterisk community. After a lot of new learning about regex, I extended the actions and filters to look at our mail server, plagued by spammers - who isn't? Our server has a unique setup now. The customer found a spam filtering service that works VERY well as the MX for the domain. Their server then connects to ours to deliver. Obviously, the IPs of that service are entered as RELAY in the sendmail config. Here is my question: We are still getting a lot of direct spam. Being that only account holders and the spam filtering servers should be connecting, I started blocking various connections bith in /etc/mail/access and in pf. However, I soon saw that I'll need to block the en tire Internet IP space. Blocking by IP is a problem for a small number of nomad users whose IP may just be in China, Russia or Argentina at some point. I was thinking of closing port 25 and using an alternate port (587?) setup if the spam service is able to connect to an alternate port. That way, the users can also change their configs to 587 and most spammers will be trying 25 which is closed. Is this a tenable idea? What are your experiences and opinions? tia /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On 13 July 2010 09:52, Randy R randulo2...@gmail.com wrote: Many of you are interested in and have used or recommended fail2ban for your linux boxes. I finally installed it on our FreeBSD server (no asterisk, hence the OT) with the help of a friend from the VoIP Users Conference and Asterisk community. After a lot of new learning about regex, I extended the actions and filters to look at our mail server, plagued by spammers - who isn't? Our server has a unique setup now. The customer found a spam filtering service that works VERY well as the MX for the domain. Their server then connects to ours to deliver. Obviously, the IPs of that service are entered as RELAY in the sendmail config. Here is my question: We are still getting a lot of direct spam. Being that only account holders and the spam filtering servers should be connecting, I started blocking various connections bith in /etc/mail/access and in pf. However, I soon saw that I'll need to block the en tire Internet IP space. Blocking by IP is a problem for a small number of nomad users whose IP may just be in China, Russia or Argentina at some point. I was thinking of closing port 25 and using an alternate port (587?) setup if the spam service is able to connect to an alternate port. That way, the users can also change their configs to 587 and most spammers will be trying 25 which is closed. Is this a tenable idea? What are your experiences and opinions? tia /r Hi Randy, How many users are on this 'domain'? Google Apps Free is a great solution for upto 50 users with 7.6GB per user. Their spam filtering usually does the job for our customers. Regards, Brian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tue, Jul 13, 2010 at 11:04 AM, dotnetdub dotnet...@gmail.com wrote: Hi Randy, How many users are on this 'domain'? Google Apps Free is a great solution for upto 50 users with 7.6GB per user. Their spam filtering usually does the job for our customers. Hi Brian, Thanks for the reply. I'm familiar with Google's pro services, we have other customers on them. However, this doesn't address the question I asked, which is regarding a FreeBSD mail server we run. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
What I do, is only open port 25 to the list of ips of the spam filtering service -- I use an iptables script called rc.firewall which I found several years ago which works well and has a nice syntax for this and I get no direct spam, I get some which gets by the filters. Randy R randulo2...@gmail.com wrote: Many of you are interested in and have used or recommended fail2ban for your linux boxes. I finally installed it on our FreeBSD server (no asterisk, hence the OT) with the help of a friend from the VoIP Users Conference and Asterisk community. After a lot of new learning about regex, I extended the actions and filters to look at our mail server, plagued by spammers - who isn't? Our server has a unique setup now. The customer found a spam filtering service that works VERY well as the MX for the domain. Their server then connects to ours to deliver. Obviously, the IPs of that service are entered as RELAY in the sendmail config. Here is my question: We are still getting a lot of direct spam. Being that only account holders and the spam filtering servers should be connecting, I started blocking various connections bith in /etc/mail/access and in pf. However, I soon saw that I'll need to block the en tire Internet IP space. Blocking by IP is a problem for a small number of nomad users whose IP may just be in China, Russia or Argentina at some point. I was thinking of closing port 25 and using an alternate port (587?) setup if the spam service is able to connect to an alternate port. That way, the users can also change their configs to 587 and most spammers will be trying 25 which is closed. Is this a tenable idea? What are your experiences and opinions? tia /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tue, Jul 13, 2010 at 12:29 PM, cov...@ccs.covici.com wrote: What I do, is only open port 25 to the list of ips of the spam filtering service -- I use an iptables script called rc.firewall which I found several years ago which works well and has a nice syntax for this and I get no direct spam, I get some which gets by the filters. Hi John, I'd like to do that, but there are nomad users who might be anywhere in the world. True maybe I could ask them to use port 587 and then allow ONLY the service IPs access to port 25. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
What you can do -- I don't know about nomad, but can you make them use authentication? Randy R randulo2...@gmail.com wrote: On Tue, Jul 13, 2010 at 12:29 PM, cov...@ccs.covici.com wrote: What I do, is only open port 25 to the list of ips of the spam filtering service -- I use an iptables script called rc.firewall which I found several years ago which works well and has a nice syntax for this and I get no direct spam, I get some which gets by the filters. Hi John, I'd like to do that, but there are nomad users who might be anywhere in the world. True maybe I could ask them to use port 587 and then allow ONLY the service IPs access to port 25. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tuesday 13 Jul 2010, Randy R wrote: I was thinking of closing port 25 and using an alternate port (587?) setup if the spam service is able to connect to an alternate port. That way, the users can also change their configs to 587 and most spammers will be trying 25 which is closed. Can't you just insist on SMTP AUTH? Or (crude but still just about usable) require a POP3 connection before allowing an SMTP connection? -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tue, Jul 13, 2010 at 12:53 PM, cov...@ccs.covici.com wrote: What you can do -- I don't know about nomad, but can you make them use authentication? They do identify, but they have to connect first :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tue, Jul 13, 2010 at 12:58 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Tuesday 13 Jul 2010, Randy R wrote: I was thinking of closing port 25 and using an alternate port (587?) setup if the spam service is able to connect to an alternate port. That way, the users can also change their configs to 587 and most spammers will be trying 25 which is closed. Can't you just insist on SMTP AUTH? Or (crude but still just about usable) require a POP3 connection before allowing an SMTP connection? The problem is that mail to legitimate users is being sent here although here is NOT the MX. On the other hand, when the users on the road try to connect to use the server to send on port 25, it needs to be open. I'm pretty sure closing 25 would kill the spam. But the users would need to connect to a port for SMTP. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording from g729 to wav means transcoding ?
On Tue, Jul 13, 2010 at 4:29 AM, Jonas Kellens jonas.kell...@telenet.be wrote: when the conversation is using the G729-codec and the conversation is recorded with the Monitor()-application in wav-format, will there be transcoding (and thus a need for licenses ?) I believe so, Yes. You can check your license use by using: *CLI g729 show licenses -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to open pseudo device
Hi List, I'm new to asterisk and currently running the newest of version. I'm encountering the error below when I dial my meetme conference #: WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo device I already tried googling this issue and found some procedure but still no luck on fixing it. My server does not have any digium hardware and I'm trying this via ztdummy. Please advise, Malvin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording from g729 to wav means transcoding ?
I have no licenses and I want to avoid transcoding all together. When the phone supports G729 and the SIP provider support G729, then the audio can just pass through... However, in some cases the audio is recorded. Any change that we can record in G729 format then ?? And how about voicemail, is this then also translated ?? Jonas. On 07/13/2010 01:15 PM, Paul Belanger wrote: On Tue, Jul 13, 2010 at 4:29 AM, Jonas Kellensjonas.kell...@telenet.be wrote: when the conversation is using the G729-codec and the conversation is recorded with the Monitor()-application in wav-format, will there be transcoding (and thus a need for licenses ?) I believe so, Yes. You can check your license use by using: *CLI g729 show licenses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tue, 13 Jul 2010, Randy R wrote: On Tue, Jul 13, 2010 at 12:58 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Tuesday 13 Jul 2010, Randy R wrote: I was thinking of closing port 25 and using an alternate port (587?) setup if the spam service is able to connect to an alternate port. That way, the users can also change their configs to 587 and most spammers will be trying 25 which is closed. Can't you just insist on SMTP AUTH? Or (crude but still just about usable) require a POP3 connection before allowing an SMTP connection? The problem is that mail to legitimate users is being sent here although here is NOT the MX. On the other hand, when the users on the road try to connect to use the server to send on port 25, it needs to be open. I'm pretty sure closing 25 would kill the spam. But the users would need to connect to a port for SMTP. Technically/pedantically, users ought to be connecting to port 587 to submit their email anyway, with port 25 being reserved for MTA to MTA communications, so block 25 for everyone but the MX relaying host and insist your users connect on port 587 to relay their outgoing email (with smtp-auth) I'd assume that most MTAs listen on 587 these days (as well as 25) - it's been in the standards for quite a number of years now. (Introduced in 1998 in RFC2476) And I don't know about where you are, but where I am (UK) some ISPs are now blocking outbound SMTP connections on port 25, or force-proxying them via their own email servers, making the use of port 587 almost mandatory - BTretail and Orange, and I think AOL do, but there's probably others. However it's only a matter of time before they catch up and as soon as the spammers start to use that port, the ISPs will block them too. Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Hylafax + Iiaxmodem - Outbound number.
Marta Silva wrote: Hi there, Thank you for your response. So I can use the ModemSetOriginCmd command to assign the outbound number on the iaxmodem, but how do I choose which modem to use for my specific sip client (GXW-4004), as I have 2 faxes connected to my GXW box? Why would you need to choose a specific modem? I believe everything can be set from the command line, when generating your fax. Granted, I haven't read the complete thread. If you have specific HylaFAX+ or iaxmodem questions, it'd probably be better to move this thread to either of those lists. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
Hi Gordon, On Tue, Jul 13, 2010 at 1:55 PM, Gordon Henderson gordon+aster...@drogon.net wrote: Technically/pedantically, users ought to be connecting to port 587 to submit their email anyway, with port 25 being reserved for MTA to MTA communications, so block 25 for everyone but the MX relaying host and insist your users connect on port 587 to relay their outgoing email (with smtp-auth) Yes. The only thing that is delicate here is the insist part, but they'll get over it. Users are customers. I'd assume that most MTAs listen on 587 these days (as well as 25) - it's been in the standards for quite a number of years now. (Introduced in 1998 in RFC2476) Yes, if you have that port open. (we do) And I don't know about where you are, but where I am (UK) some ISPs are now blocking outbound SMTP connections on port 25, or force-proxying them via their own email servers, making the use of port 587 almost mandatory - BTretail and Orange, and I think AOL do, but there's probably others. However it's only a matter of time before they catch up and as soon as the spammers start to use that port, the ISPs will block them too. Yes, more and more providers do this. So (even before I read your message) I decided to limit port 25 access to the restricted IP set we know about. This will be an interesting 48 hours or so while we see if the users are still using port 25 :-) /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MyFuel Express FO - Shortcomings
Re-sent copying UNON and Expand Technologies. Apologies for the omission. Rgds, Alphonse On Tue, Jul 13, 2010 at 3:27 PM, Alphonse Ogulla aogu...@gmail.com wrote: Dear Esther, The foregoing mail notes sent to Expand Technology refer, in view that you were not copied in the initial correspondence. We were hoping that Expand Technology would make a comeback today with a course of action to resolve certain shortcomings flagged in MyFuel Express but unfortunately this has not been the case. Kindly contact Expand Technology and make it clear that we need the five critical elements resolved in the next 7 days to enable us progress with the system upgrade as planned. An improved version of MyFuel Express should be released speedily without unnecessary mention to the ToR, more so because we are requesting minute revision to code in beta stage and not significant modification or new functionality in a final product. Please follow up on our behalf and revert with firmed up dates when we can get a new version of MyFuel Express without the listed drawbacks. Best Regards, Alphonse Ogulla -- Forwarded message -- From: Jacques de Gersigny j.degersi...@expand-technology.com Date: Mon, Jul 12, 2010 at 8:34 PM Subject: Re: MyFuel Express FO - Shortcomings To: Alphonse Ogulla aogu...@gmail.com Cc: Lovena Modelly l.mode...@expand-technology.com, James Gathoga j.gath...@expand-technology.com, Simon Beamish simon.beam...@unon.org, Sanjita Sehmi sanjita.se...@unon.org, Sheila Cardovillis sheila.cardovil...@unon.org Hi Alphonse, Simon, I'm in a Business trip and I will get back on monday next. Rgds, JDG On 8 July 2010 16:05, Alphonse Ogulla aogu...@gmail.com wrote: Dear Jacques, We tried getting you on phone in the office at noon (Kenya time) but unfortunately you had stepped out for lunch. We however managed to get hold of Lovena and briefly deliberated the critical items in the ensuing email. In principle, we agreed to address these drawbacks in the following manner: 1) Expand Technologies to resolve items 1a (card printing) and 1c (bank card cheque payment currency) without further deliberations.. 2) Refer to the final signed TOR for items 1b (card transfer) and 1d (FO direct topup). I shall get the final TOR from Easther Wanjoga of Kenya Shell Ltd. 3) Lastly, Expand Technologies to check if the chip card has sufficient space to store the expiry date in order to implement item 1e (card validity). I'm also made to understand that you called Simon Beamish and discussed further the items listed above. Kindly look into these issues keenly and revert with a proposal on headway latest by Monday 12th July AM. Please remember to copy Shell in your rejoinder. Looking forward to hearing from you soon. Best Regards, Alphonse Ogulla Tel: +254 20 7621510 Mobile: +254 723 465172 On Mon, Jul 5, 2010 at 12:15 PM, Alphonse Ogulla aogu...@gmail.comwrote: Dear Jacques et alia, We have identified certain shortcomings in MyFuel Express Front Office (FO) software that we need rectified as soon as possible and in-time for the go live scheduled for next month. The critical elements should be given uttermost priority as it is impossible to commence printing and issuing of cards with the listed drawbacks still in place. 1) CRITICAL ELEMENTS a) Card Printing and Personalisation Increase padding on the left margin so that the card holder name, description and vehicle registration do not print on the UN logo. b) Card Transfer The required functionality should be transfer of card value on the e-purse and not transfer of the card-holder particulars to another card as is currently the case. c) Local Epurse Remote Top-up (Bank card cheque payment) Currently only the cash top-up function has the option of selecting the paying currency. A similar option is required for the bank card and cheque top-up since many clients run US$ transactions on their credit/debit bank cards. Similarly, US$ account holders have cheque books for US$ transactions only. d) Front Office Direct Top-up The FO lacks direct top-up capability whereas there is a card reader/writer directly connected to the FO. A work-around has been implemented by connecting a hand held POS to the Ethernet network to download and effect the actual top-up on the card. This two step procedure is time consuming and shall drastically slow down the top-up process at the station. e) Client Management - Card Validity Provide entry for expiry date i.e. dd/mm/ instead of number of years since client contracts expire on specific dates and not at the end of the year. 2) IMPROVEMENTS a) User Management - User Rights Group the list of functions into various roles to ease creation of a new user. We acknowledge that new users shall not be created frequently but currently one has to chose from over 200 different functions/rights when defining a user for the first time. 3)
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tue, 13 Jul 2010, Randy R wrote: Hi Gordon, On Tue, Jul 13, 2010 at 1:55 PM, Gordon Henderson gordon+aster...@drogon.net wrote: Technically/pedantically, users ought to be connecting to port 587 to submit their email anyway, with port 25 being reserved for MTA to MTA communications, so block 25 for everyone but the MX relaying host and insist your users connect on port 587 to relay their outgoing email (with smtp-auth) Yes. The only thing that is delicate here is the insist part, but they'll get over it. Users are customers. Indeed - and with another 'hat' on, I run an ISP business, providing email and web hosting facilities to clients - and facing exactly the same issues. It's been (being) a struggle to get people to change their settings, but we're slowly getting there. Because I have multiple servers, I can run both in parallel and are giving groups of customers cut-off dates for final migration based on the servers their using to relay outbound email... I'd assume that most MTAs listen on 587 these days (as well as 25) - it's been in the standards for quite a number of years now. (Introduced in 1998 in RFC2476) Yes, if you have that port open. (we do) And I don't know about where you are, but where I am (UK) some ISPs are now blocking outbound SMTP connections on port 25, or force-proxying them via their own email servers, making the use of port 587 almost mandatory - BTretail and Orange, and I think AOL do, but there's probably others. However it's only a matter of time before they catch up and as soon as the spammers start to use that port, the ISPs will block them too. Yes, more and more providers do this. So (even before I read your message) I decided to limit port 25 access to the restricted IP set we know about. This will be an interesting 48 hours or so while we see if the users are still using port 25 :-) Good luck! Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MyFuel Express FO - Shortcomings
Did you mean to send this to a mailing list?.. S On 13 Jul 2010, at 13:33, Alphonse Ogulla wrote: Re-sent copying UNON and Expand Technologies. Apologies for the omission. Rgds, Alphonse On Tue, Jul 13, 2010 at 3:27 PM, Alphonse Ogulla aogu...@gmail.com wrote: Dear Esther, The foregoing mail notes sent to Expand Technology refer, in view that you were not copied in the initial correspondence. We were hoping that Expand Technology would make a comeback today with a course of action to resolve certain shortcomings flagged in MyFuel Express but unfortunately this has not been the case. Kindly contact Expand Technology and make it clear that we need the five critical elements resolved in the next 7 days to enable us progress with the system upgrade as planned. An improved version of MyFuel Express should be released speedily without unnecessary mention to the ToR, more so because we are requesting minute revision to code in beta stage and not significant modification or new functionality in a final product. Please follow up on our behalf and revert with firmed up dates when we can get a new version of MyFuel Express without the listed drawbacks. Best Regards, Alphonse Ogulla -- Forwarded message -- From: Jacques de Gersigny j.degersi...@expand-technology.com Date: Mon, Jul 12, 2010 at 8:34 PM Subject: Re: MyFuel Express FO - Shortcomings To: Alphonse Ogulla aogu...@gmail.com Cc: Lovena Modelly l.mode...@expand-technology.com, James Gathoga j.gath...@expand-technology.com, Simon Beamish simon.beam...@unon.org, Sanjita Sehmi sanjita.se...@unon.org, Sheila Cardovillis sheila.cardovil...@unon.org Hi Alphonse, Simon, I'm in a Business trip and I will get back on monday next. Rgds, JDG On 8 July 2010 16:05, Alphonse Ogulla aogu...@gmail.com wrote: Dear Jacques, We tried getting you on phone in the office at noon (Kenya time) but unfortunately you had stepped out for lunch. We however managed to get hold of Lovena and briefly deliberated the critical items in the ensuing email. In principle, we agreed to address these drawbacks in the following manner: 1) Expand Technologies to resolve items 1a (card printing) and 1c (bank card cheque payment currency) without further deliberations.. 2) Refer to the final signed TOR for items 1b (card transfer) and 1d (FO direct topup). I shall get the final TOR from Easther Wanjoga of Kenya Shell Ltd. 3) Lastly, Expand Technologies to check if the chip card has sufficient space to store the expiry date in order to implement item 1e (card validity). I'm also made to understand that you called Simon Beamish and discussed further the items listed above. Kindly look into these issues keenly and revert with a proposal on headway latest by Monday 12th July AM. Please remember to copy Shell in your rejoinder. Looking forward to hearing from you soon. Best Regards, Alphonse Ogulla Tel: +254 20 7621510 Mobile: +254 723 465172 On Mon, Jul 5, 2010 at 12:15 PM, Alphonse Ogulla aogu...@gmail.com wrote: Dear Jacques et alia, We have identified certain shortcomings in MyFuel Express Front Office (FO) software that we need rectified as soon as possible and in-time for the go live scheduled for next month. The critical elements should be given uttermost priority as it is impossible to commence printing and issuing of cards with the listed drawbacks still in place. 1) CRITICAL ELEMENTS a) Card Printing and Personalisation Increase padding on the left margin so that the card holder name, description and vehicle registration do not print on the UN logo. b) Card Transfer The required functionality should be transfer of card value on the e-purse and not transfer of the card-holder particulars to another card as is currently the case. c) Local Epurse Remote Top-up (Bank card cheque payment) Currently only the cash top-up function has the option of selecting the paying currency. A similar option is required for the bank card and cheque top-up since many clients run US$ transactions on their credit/debit bank cards. Similarly, US$ account holders have cheque books for US$ transactions only. d) Front Office Direct Top-up The FO lacks direct top-up capability whereas there is a card reader/writer directly connected to the FO. A work-around has been implemented by connecting a hand held POS to the Ethernet network to download and effect the actual top-up on the card. This two step procedure is time consuming and shall drastically slow down the top-up process at the station. e) Client Management - Card Validity Provide entry for expiry date i.e. dd/mm/ instead of number of years since client contracts expire on specific dates and not at the end of the year. 2) IMPROVEMENTS a) User Management - User Rights Group the list of functions into various roles to ease creation of a new user. We acknowledge
Re: [asterisk-users] Unable to open pseudo device
On Tuesday 13 July 2010 06:35:45 Malvin Rito wrote: Hi List, I'm new to asterisk and currently running the newest of version. I'm encountering the error below when I dial my meetme conference #: WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo device I already tried googling this issue and found some procedure but still no luck on fixing it. My server does not have any digium hardware and I'm trying this via ztdummy. It's likely an issue of permissions. Check the permissions of /dev/dahdi/pseudo versus the user your Asterisk daemon runs at. If necessary, change your permission script in /etc/udev.d/ to match the ownership of the pseudo device to the user running Asterisk. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MyFuel Express FO - Shortcomings **PLEASE DELETE THREAD**
My sincere apologies for inadvertently sending this mail note to this list which happens to be the first entry in my address book. If you are a list administrator, kindly delete this thread from the list. My apologies once again and please do not reply. Rgds, Alphonse On Tue, Jul 13, 2010 at 4:04 PM, Steve Howes steve-li...@geekinter.netwrote: Did you mean to send this to a mailing list?.. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tue, Jul 13, 2010 at 2:45 PM, Gordon Henderson gordon+aster...@drogon.net wrote: Good luck! A few have written me off list (thanks) so I thought I'd close out my own thoughts on this. It's been about two hours and it does look like things are working great. I removed the huge number of CONNECT...REJECT statements in sendmail (not needed since the port isn't there any more). I put the authorized IP list in a pf table and all that is working just fine. Yes, many people over the past few years have complained they weren't able to send mail and were told to change the port to 587, which we opened last year. Our situation is unusual as the people out on the road vary a lot, and some of them use a VPN to access the customer EXCHANGE directly. That can connect directly to our box, so it causes no problems for them. So unless we hear from stragglers on port 25, this is looking very good. I don't know how many IP can be put in a pf table but it was obvious that there would be tens of thousands in a very small number of days. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MyFuel Express FO - Shortcomings **PLEASE DELETE THREAD**
On 07/13/2010 08:27 AM, Alphonse Ogulla wrote: My sincere apologies for inadvertently sending this mail note to this list which happens to be the first entry in my address book. If you are a list administrator, kindly delete this thread from the list. My apologies once again and please do not reply. Threads cannot be deleted from the list; once messages are posted, they appear in the archives (of which there are many) and are delivered to thousands of subscribers. Sorry. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to install speex codec for Asterisk that is downloaded from Digium Yum Repository?
Hi Everyone, I have done yum install speex libspeex-devel speex-devel and it was succesful on CentOS. I then tried yum install asterisk16 asterisk16-addons asterisk16-configs but core show translation doesn't show speex loaded. Is there a way to or an option that I can append to the asterisk install to make sure it compiles with speex in mind? Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan I am getting this error without webmeetme at all, after upgrading to svn-275706 from an earlier version 262801. Its a certain argument of meetme which I have not trafcked down yet which is causing this. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STRFTIME function declared in globals context
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
If I evaluate these variables in the dialplan later, using
exten = ,n,Verbose(${TIMESTAMP} - ${YEAR} - ${MONTH} - ${DAY})
My output is as follows:
-- Executing [7...@phones:3] Verbose(SIP/2625-d5f0, Tue Jul 13
11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue
Jul 13 11:08:42 2010) in new stack
Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42
2010 - Tue Jul 13 11:08:42 2010
However, the following line:
exten = ,n,Verbose(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} -
${STRFTIME(${EPOCH},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
${STRFTIME(${EPOCH},,%d)})
evaluates with what I expect:
-- Executing [7...@phones:4] Verbose(SIP/2625-d5f0,
20100713-110853 - 2010 - 07 - 13) in new stack
20100713-110853 - 2010 - 07 - 13
Is what I'm trying to do possible? It seems like it's at least recognizing
that I'm trying to grab a date, but it's not taking the date format
parameters that I want.
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
_
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
cov...@ccs.covici.com wrote: Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan I am getting this error without webmeetme at all, after upgrading to svn-275706 from an earlier version 262801. Its a certain argument of meetme which I have not trafcked down yet which is causing this. OK, if the argument to meetme is conference number,TcMsrm it does not crash, but if it is conference number, cMs then it dies -- asterisk dies. Is this enough for someone to figure out? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Tuesday, July 13, 2010 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] STRFTIME function declared in globals context
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
If I evaluate these variables in the dialplan later, using
exten = ,n,Verbose(${TIMESTAMP} - ${YEAR} - ${MONTH} - ${DAY})
My output is as follows:
-- Executing [7...@phones:3] Verbose(SIP/2625-d5f0, Tue Jul 13
11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue
Jul 13 11:08:42 2010) in new stack
Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42
2010 - Tue Jul 13 11:08:42 2010
However, the following line:
exten = ,n,Verbose(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} -
${STRFTIME(${EPOCH},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
${STRFTIME(${EPOCH},,%d)})
evaluates with what I expect:
-- Executing [7...@phones:4] Verbose(SIP/2625-d5f0,
20100713-110853 - 2010 - 07 - 13) in new stack
20100713-110853 - 2010 - 07 - 13
Is what I'm trying to do possible? It seems like it's at least recognizing
that I'm trying to grab a date, but it's not taking the date format
parameters that I want.
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
You don't state which version you are on (These things change from 1.2 to
1.4 to 1.6/8), but that being said, you would probably more likely to
succeed doing Set(GLOBAL) in an isolated context instead of using the
[global] context for this bit of voodoo. Looking forward to a better answer
(folks like to correct my shots across the bow).
--
_
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Re: [asterisk-users] STRFTIME function declared in globals context
On Tue, Jul 13, 2010 at 11:47 AM, Danny Nicholas da...@debsinc.com wrote: You don’t state which version you are on (These things change from 1.2 to 1.4 to 1.6/8), but that being said, you would probably more likely to succeed doing Set(GLOBAL) in an isolated context instead of using the [global] context for this bit of voodoo. Looking forward to a better answer (folks like to correct my shots across the bow). Sorry about that. I'm on version 1.4.33.1. I'm wanting to use these variables in multiple contexts throughout the dialplan, and also as arguments to some AGI scripts, etc. I'd rather they be set in the [globals] context if at all possible, that way I'm not limited to where I can use them. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
On Tue, Jul 13, 2010 at 11:30:44AM -0500, Warren Selby wrote:
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
If I evaluate these variables in the dialplan later, using
exten = ,n,Verbose(${TIMESTAMP} - ${YEAR} - ${MONTH} - ${DAY})
My output is as follows:
-- Executing [7...@phones:3] Verbose(SIP/2625-d5f0, Tue Jul 13
11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue
Jul 13 11:08:42 2010) in new stack
Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42 2010 - Tue Jul 13 11:08:42
2010 - Tue Jul 13 11:08:42 2010
However, the following line:
exten = ,n,Verbose(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} -
${STRFTIME(${EPOCH},,%Y)} - ${STRFTIME(${EPOCH},,%m)} -
${STRFTIME(${EPOCH},,%d)})
evaluates with what I expect:
-- Executing [7...@phones:4] Verbose(SIP/2625-d5f0,
20100713-110853 - 2010 - 07 - 13) in new stack
20100713-110853 - 2010 - 07 - 13
Is what I'm trying to do possible? It seems like it's at least recognizing
that I'm trying to grab a date, but it's not taking the date format
parameters that I want.
Try adding preload = func_strings.so to modules.conf
--
Barry
--
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Re: [asterisk-users] Delay between answer and pickup ?
2010/7/11 Julian Lyndon-Smith aster...@dotr.com Anyone got a clue ? (he asks in desperation!) Julian On 9 July 2010 17:48, Julian Lyndon-Smith aster...@dotr.com wrote: We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing hello ? Hello ? and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of asterisk don't show this discrepancy. We are using a local channel to dial a landline through a sip provider. Could it be possible that the firewall is waiting for outbound RTP media, before letting inbound RTP come in ? When the call is answered, the agent's phone is then dialled. the logs go something like this [Jul 9 13:29:26] VERBOSE[23396] logger.c: [Jul 9 13:29:26] -- SIP/provider-0001ed6e is making progress passing it to Local/somenum...@dialleroutbound-4c93,2 [Jul 9 13:29:44] VERBOSE[23396] logger.c: [Jul 9 13:29:44] -- SIP/provider-0001ed6e answered Local/01577864...@dialleroutbound-4c93,2 What would rtp debug show ? .. [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Executing [*00...@diallerconnected:2] Dial(Local/somenum...@dialleroutbound-4c93,1, SIP/*0086*|5|iA(autoanswer)) in new stack [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- SIP/*0086*-0001ed73 answered Local/somenum...@dialleroutbound-4c93,1 .. as you can see, the call is answered at 13:29:44 and the agent gets called (auto-answer phones) at 13:29:46, yes if you listen to the call recording, there is a 6 second gap between the person saying hello and the agent being connected. Is it possible that the call was answered 5 seconds *before* I get notification of the answer ? i.e. is the provider taking too long notifying me of the answer ? Julian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
On Tue, Jul 13, 2010 at 01:07:34PM -0400, Barry Miller wrote: Try adding preload = func_strings.so to modules.conf Ah, sorry. I just saw your earlier response that said you're on 1.4 - I was remembering that after I migrated from 1.4 - 1.6, I had to preload func_db.so so that I could use the DB function in [globals]. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk un-registering from provider
All: Starting switching over my phone lines. I got phone line 1 switched. Everyone working. I switched the second phone line, and it worked about an hour, then I started getting errors from the cli saying the server could not register with the providing. I restarted the system, and it worked ok for about 30 minutes, and then started giving he same errors. The error is [Jul 13 11:21:14] NOTICE[27331]: chan_sip.c:10169 sip_reg_timeout:-- Registration for '4342201...@ia.ntelos.net' timed out, trying again (Attempt #19) doing dnsmgr_lookup for 'ia.ntelos.net' It keeps doing this until I restart asterisk. No, the password hasn't changed - the system works fine for anywhere from 5 minutes to 30 minutes, but again, I suspect it depends on the call load. I suspect it has something to do with the call load, but there really wasn't that many calls in progress - maybe 5 at the time of the failure. I have since switched the last phone back to the old system (it handles the volume of our client calls). Anyone else experience this? Where should I start looking - at my server, or at the provider? Thanks, Eddie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
On Tuesday 13 July 2010 11:30:44 Warren Selby wrote:
I'm trying to declare a few date-related global variables to ease my
dialplan. When I declare the following in the [globals] context of
extensions.conf, I get unexpected results:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
MONTH = ${STRFTIME(${EPOCH},,%m)}
DAY = ${STRFTIME(${EPOCH},,%d)}
TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}
When you load the dialplan, do you see the global variables getting set?
That would at least tell you whether the problem lies at the point where the
values are loaded into memory, or later, at evaluation time.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] STRFTIME function declared in globals context
On Tue, Jul 13, 2010 at 1:15 PM, Tilghman Lesher tles...@digium.com wrote: When you load the dialplan, do you see the global variables getting set? That would at least tell you whether the problem lies at the point where the values are loaded into memory, or later, at evaluation time. == Setting global variable 'YEAR' to 'Tue Jul 13 13:36:55 2010' == Setting global variable 'MONTH' to 'Tue Jul 13 13:36:55 2010' == Setting global variable 'DAY' to 'Tue Jul 13 13:36:55 2010' == Setting global variable 'TIMESTAMP' to 'Tue Jul 13 13:36:55 2010' So apparently they're loaded into memory when the dialplan is reloaded? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Tuesday, July 13, 2010 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] STRFTIME function declared in globals context
On Tue, Jul 13, 2010 at 1:15 PM, Tilghman Lesher tles...@digium.com wrote:
When you load the dialplan, do you see the global variables getting set?
That would at least tell you whether the problem lies at the point where the
values are loaded into memory, or later, at evaluation time.
== Setting global variable 'YEAR' to 'Tue Jul 13 13:36:55 2010'
== Setting global variable 'MONTH' to 'Tue Jul 13 13:36:55 2010'
== Setting global variable 'DAY' to 'Tue Jul 13 13:36:55 2010'
== Setting global variable 'TIMESTAMP' to 'Tue Jul 13 13:36:55 2010'
So apparently they're loaded into memory when the dialplan is reloaded?
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
Since you never know when you'll need this, I slapped the code into my
1.4.30.
Here is the corrected code that works
YEAR = ${STRFTIME(${EPOCH}||%Y)}
MONTH = ${STRFTIME(${EPOCH}||%m)}
DAY = ${STRFTIME(${EPOCH}||%d)}
TIMESTAMP = ${STRFTIME(${EPOCH}||%Y%m%d-%H%M%S)}
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Re: [asterisk-users] STRFTIME function declared in globals context
On Tue, Jul 13, 2010 at 1:49 PM, Danny Nicholas da...@debsinc.com wrote:
--
Since you never know when you’ll need this, I slapped the code into my
1.4.30.
Here is the “corrected” code that works
YEAR = ${STRFTIME(${EPOCH}||%Y)}
MONTH = ${STRFTIME(${EPOCH}||%m)}
DAY = ${STRFTIME(${EPOCH}||%d)}
TIMESTAMP = ${STRFTIME(${EPOCH}||%Y%m%d-%H%M%S)}
Wow, I'm surprised that worked, but it did! Thanks very much!
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] OT: fail2ban, spam and mail servers
On Tue, 2010-07-13 at 06:53 -0400, cov...@ccs.covici.com wrote: What you can do -- I don't know about nomad, but can you make them use authentication? Randy R randulo2...@gmail.com wrote: On Tue, Jul 13, 2010 at 12:29 PM, cov...@ccs.covici.com wrote: What I do, is only open port 25 to the list of ips of the spam filtering service -- I use an iptables script called rc.firewall which I found several years ago which works well and has a nice syntax for this and I get no direct spam, I get some which gets by the filters. Hi John, I'd like to do that, but there are nomad users who might be anywhere in the world. True maybe I could ask them to use port 587 and then allow ONLY the service IPs access to port 25. /r Just wondering, Most spammers or cases of joe-jobs originate from an other URL then they claim to come from. Can this not be dealt with using certificates? Something like for nomads, only accepting signed messages... hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
On Tue, 13 Jul 2010, Warren Selby wrote:
YEAR = ${STRFTIME(${EPOCH},,%Y)}
On Tue, 13 Jul 2010, Danny Nicholas wrote:
YEAR = ${STRFTIME(${EPOCH}||%Y)}
Good catch. Looks like a bug to me.
Not that anybody cares, but the 2 statements exhibit the same bug in 1.2.
Just out of curiosity, why is the time the dialplan is reloaded of
interest?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!
so nobody seems to like dealing with fax!!
2010/7/12 khalid touati khalidtou...@gmail.com
Hi Guys,
i am using the latest version of asterisk 1.4 (1.4.33.1), dahdi (2.3.0.1)
and FFA (Applications: 1.4_1.2.0, Digium FAX Driver: 1.4_1.2.0). the issue
i'm having is that i'm able to receive faxes from a website (that offer this
service) but not able to receive from a regular fax machine (that is working
perfect).
[fax-rx]
exten = receive,1,NoOp( FAX RECEIVE ) exten =
receive,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ]) exten =
receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten =
receive,n,Set(FAXFILE=${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM})}.tif)
exten =
receive,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},GMT-5,%F_%T_${CALLERIDNUM})})
exten = receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
exten = receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
exten = receive,n,NoOp( SETTING FAXOPT ) exten =
receive,n,Set(FAXOPT(ecm)=yes) exten = receive,n,Set(FAXOPT(headerinfo)=MY
FAXBACK RX) exten = receive,n,Set(FAXOPT(localstationid)=15184893772)
exten = receive,n,Set(FAXOPT(maxrate)=14400)
exten = receive,n,Set(FAXOPT(minrate)=2400)
exten = receive,n,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)}) exten =
receive,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)}) exten =
receive,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)}) exten =
receive,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)}) exten =
receive,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)}) exten =
receive,n,NoOp( RECEIVING FAX : ${FAXFILE} ) exten =
receive,n,ReceiveFAX(/var/spool/asterisk/fax/${FAXFILE})
exten = receive,n,System('/usr/local/bin/fax2mail -p -f ${FAXFILENOEXT}
--cid-number ${CALLERID(num)} --cid-name ${CALLERID(name)} --dest-name
Sir/Madam')
a previous debugging showed:
*- for a fax from myfax.com that was received successfully:*
pbx1*CLI
Channel 'DAHDI/1-1' fax session '53', [ 034.021683 ], channel
sent 59 frames (1180 ms) of energy.
pbx1*CLI
-- Channel 'DAHDI/1-1' fax session '53', [ 040.489601 ],
STAT_EVT_HW_CLOSE st: WT_HW_CLSrt: WCLSNCLS
pbx1*CLI
-- Channel 'DAHDI/1-1' fax session '53', [ 040.489798 ],
STAT_SES_COMPLETE
pbx1*CLI
-- Channel 'DAHDI/1-1' fax session '53' is complete, result: 'SUCCESS'
(FAX_SUCCESS), error: 'NO_ERROR', pages: 2, resolution: '204x196', transfer
rate: '14400', remoteSID: 'FAX'
pbx1*CLI
-- Executing [rece...@fax-rx:21] System(DAHDI/1-1,
/usr/local/bin/fax2mail -p -f
/var/spool/asterisk/fax/2010-05-18_03:59:42_ --cid-number --cid-name
--dest-name Sir/Madam) in new stack
pbx1*CLI
== Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN'
pbx1*CLI
-- Hungup 'DAHDI/1-1'
-* for a fax from regular machine that failed:*
*
*
pbx1*CLI
Channel 'DAHDI/1-1' fax session '54', [ 032.782251 ], channel
sent 3 frames (60 ms) of energy.
pbx1*CLI
-- Channel 0/1, span 1 got hangup request, cause 16
pbx1*CLI
[May 17 19:02:41] NOTICE[1316]: res_fax.c:993 generic_fax_exec: Channel
'DAHDI/1-1' did not return a frame; probably hung up.
pbx1*CLI
-- Channel 'DAHDI/1-1' fax session '54', [ 038.131701 ],
STAT_EVT_HW_CLOSE st: WT_HW_CLSrt: WCLSNCLS
pbx1*CLI
-- Channel 'DAHDI/1-1' fax session '54', [ 038.131879 ],
STAT_SES_COMPLETE
pbx1*CLI
-- Channel 'DAHDI/1-1' fax session '54' is complete, result: 'SUCCESS'
(FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: '204x196', transfer
rate: '14400', remoteSID: '518 489 3772'
pbx1*CLI
== Spawn extension (fax-rx, receive, 20) exited non-zero on 'DAHDI/1-1'
pbx1*CLI
-- Hungup 'DAHDI/1-1'
I would really appreciate any help! thanks!
--
Abdullah
--
Abdullah
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Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!
- khalid touati khalidtou...@gmail.com wrote: so nobody seems to like dealing with fax!! 'Fax for Asterisk' is a commercial application sold by Digium. This is not their official support channel. Since you paid for the product, why not contact them directly about your problem? And yes, to answer your question, not many people like dealing with fax. Some of us however are so lucky that we get to deal with fax every day. :-) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME function declared in globals context
On Tue, Jul 13, 2010 at 2:36 PM, Steve Edwards asterisk@sedwards.comwrote: Good catch. Looks like a bug to me. I'll create an issue on the tracker later today. Just out of curiosity, why is the time the dialplan is reloaded of interest? http://lists.digium.com/mailman/listinfo/asterisk-users I hadn't thought about it, but I suppose if it's evaluating at reload and not runtime, this would indeed be the case. I'm hoping to get the various date-related values at runtime, not the reload time. Hmmm...it looks like I'll have to re-evaluate what I'm trying to do... -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!
Hi! 'Fax for Asterisk' is a commercial application sold by Digium. This is not their official support channel. Since you paid for the product, why not contact them directly about your problem? Maybe because having to deal with Digium support is an ... uncomfortable experience that I've made twice. It really feels like a support avoidance system, and you are an unwelcome guest that should please leave as soon as possible. I really hope that Digium takes steps to a) make their behaviour less bureaucratic when establishing a support call (try to do that on behalf of a clue-less customer of yours and you know what I mean). Also my feeling is that b) the level of competence (or familiarity with the product in question when it comes to software prodcuts) could use some improvement. If you take a look at the Digium web forum for Skype for Asterisk, for example, you will see that any kind of half-official answer or helpful reaction from Digium's side is by now non-existent - most probably due to internal policies. This just doesn't feel right, and many other companies haven proven that I can be done differently. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!
On Tue, Jul 13, 2010 at 4:43 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! 'Fax for Asterisk' is a commercial application sold by Digium. This is not their official support channel. Since you paid for the product, why not contact them directly about your problem? Maybe because having to deal with Digium support is an ... uncomfortable experience that I've made twice. It really feels like a support avoidance system, and you are an unwelcome guest that should please leave as soon as possible. I really hope that Digium takes steps to a) make their behaviour less bureaucratic when establishing a support call (try to do that on behalf of a clue-less customer of yours and you know what I mean). Also my feeling is that b) the level of competence (or familiarity with the product in question when it comes to software prodcuts) could use some improvement. If you take a look at the Digium web forum for Skype for Asterisk, for example, you will see that any kind of half-official answer or helpful reaction from Digium's side is by now non-existent - most probably due to internal policies. This just doesn't feel right, and many other companies haven proven that I can be done differently. Philipp I couldn't have said it better myself. http://tinyurl.com/3x4yt9k -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording from g729 to wav means transcoding ?
On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens jonas.kell...@telenet.be wrote: I have no licenses and I want to avoid transcoding all together. For terminating a call into Asterisk, you need g729 licenses. It is that simple. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?
I agree with horns you'll usually get better coverage. I have done this in the past with 5 speakers for a 30k sq ft warehouse very good coverage. Using bogen horns. This was for a 300ft by 100ft warehouse. Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft side I installed a horn every 60ft alternating facing one north and the other south, which ended up 3 facing one way and 2 the other. You can get double horn speakers which will face 2 sides. I wouldn't mount them on the wall specifically not so low as fork lifts and what not will damage them. On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce bruceb...@gmail.com wrote: Well, these are horn speakers with 30 Watt which will receive 10 Watt only from Amplifer. I am not connecting them to ceiling so maybe 10 feet off the ground. I guess my coverage would be better??? Based on your calculations for for 40k sqfeet that would be 33 speakers. I think that's way too much of an overkill. thanks, Bruce On Mon, Jul 12, 2010 at 1:05 AM, C F shma...@gmail.com wrote: In my experience using height for radius works, for example if you have a 20 ft high ceiling then the coverage for one speaker would be 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft has never killed anyone, but this really depends on the power of the speaker, I usually deal with 70v speakers tapped at 16 or 8 watts depending on how many speakers I put on one amplifier and the output wattage of that amplifier. On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. I appreciate your advice. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] power outage
It has nothing to do with the D-channel, however you will never know if the B-channels work if the D-channel is down. D-channel is what allows the B-channels to work, and is the first place to troubleshoot. If something is screwed up with the power the symptom you'll get is a non working PRI, the way to check it is by means of seeing if the D-channel synced up or not. On Mon, Jul 12, 2010 at 2:17 AM, Justin Case nogoodnameswereavaila...@gmail.com wrote: What would the power have to do with the D Channel ? Isn't which channel used a logical setting (as opposed to physical). I am not saying your wrong I am just trying to understand why it happens. On Mon, Jul 12, 2010 at 7:56 AM, C F shma...@gmail.com wrote: I have found that sometimes shutting down the machine waiting a full minute while the power cable is unplugged then restarting can fix such problems if it's power related. On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis ge...@pagestation.com wrote: I have a TE205P that has been working fine for 2 years. power outage yesterday took out my everything for over an hour. Everything has come back up except the PRI. My provider has checked it to the box and says everything looks good on their end. I get this message: [Jul 9 12:40:32] WARNING[13709] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! ztcfg -vvv Zaptel Version: 1.4.12.1 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 7 channels to configure. and show status gives me condition RED of course. How do I find out whats wrong here? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to open pseudo device
Thanks for the reply. There is no folder dahdi under /dev folder. I cannot also find /udev.d on /etc folder. Under /dev folder I only see /dev/zap/pseudo. Regards, Malvin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Tuesday, July 13, 2010 9:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to open pseudo device On Tuesday 13 July 2010 06:35:45 Malvin Rito wrote: Hi List, I'm new to asterisk and currently running the newest of version. I'm encountering the error below when I dial my meetme conference #: WARNING[13220]: app_meetme.c:1097 build_conf: Unable to open pseudo device I already tried googling this issue and found some procedure but still no luck on fixing it. My server does not have any digium hardware and I'm trying this via ztdummy. It's likely an issue of permissions. Check the permissions of /dev/dahdi/pseudo versus the user your Asterisk daemon runs at. If necessary, change your permission script in /etc/udev.d/ to match the ownership of the pseudo device to the user running Asterisk. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to pass through supported 100rel
hello I want to know how to pass through 100rel header. and I hope that asterisk PRACK to UAS.(RFC3262 behavior) _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk, capable of receiving from website but not from fax machine !!
'Fax for Asterisk' is a commercial application sold by Digium. This is not their official support channel. Since you paid for the product, why not contact them directly about your problem? i did get this version for free after buying a (actually several) digium telephony card, but i realized that they're not supporting the free version after talking and emailing them, actually i was calling Digium support for all the past year and i can say that (for me) it was good:4/5 satisfaction, but this time with fax, i didn't get much help, i was redirected to the community and that why i posted. by the way is there a reliable alternative? is for 1.6 rfax is doing good (if anyone worked with it)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't compile DAHDI - wrong kernel source
I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux When I try to compile DAHDI it fails with: make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. Is there a way to trick DAHDI to use the installed kernel? Thanks for the help! -- -Thermal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?
Thanks for the input guys. For other refrence, a CyberData Voip Amplifier which supplies 10 Watt to each of the two bogen 30 Watt speakers did the job for a 35, square feet warehouse with environmental noise level of slightly higher than standard but not those of industrial. Only two speakers and done deal. Though I know that three speaker would have been the perfect solution but 4 would cover every single little corner and be an overkill. -Bruce On Tue, Jul 13, 2010 at 8:47 PM, C F shma...@gmail.com wrote: I agree with horns you'll usually get better coverage. I have done this in the past with 5 speakers for a 30k sq ft warehouse very good coverage. Using bogen horns. This was for a 300ft by 100ft warehouse. Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft side I installed a horn every 60ft alternating facing one north and the other south, which ended up 3 facing one way and 2 the other. You can get double horn speakers which will face 2 sides. I wouldn't mount them on the wall specifically not so low as fork lifts and what not will damage them. On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce bruceb...@gmail.com wrote: Well, these are horn speakers with 30 Watt which will receive 10 Watt only from Amplifer. I am not connecting them to ceiling so maybe 10 feet off the ground. I guess my coverage would be better??? Based on your calculations for for 40k sqfeet that would be 33 speakers. I think that's way too much of an overkill. thanks, Bruce On Mon, Jul 12, 2010 at 1:05 AM, C F shma...@gmail.com wrote: In my experience using height for radius works, for example if you have a 20 ft high ceiling then the coverage for one speaker would be 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft has never killed anyone, but this really depends on the power of the speaker, I usually deal with 70v speakers tapped at 16 or 8 watts depending on how many speakers I put on one amplifier and the output wattage of that amplifier. On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. I appreciate your advice. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Silence RTP
hello I found silence RTP packet from Asterisk in early dialog. I want to know reason and how to solve. RTP packet 80 00 40 22 00 0c 74 58 06 98 eb 44 ff ff ff ff ..@..tX...D 0010 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0020 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0030 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0040 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0050 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0060 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0070 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0080 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0090 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 00a0 ff ff ff ff ff ff ff ff ff ff ff ff _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
